Information Technology Network and Internet: Innovative Single Window Book from Base to Research

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DEDICA TED to DEDICATED My late parents parents,, Satish Chandra Bhunia and Santa Prabha Bhunia Shyamsundar pur, Sabong Shyamsundar pur, East Midnapore, West Bengal, India

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Preface

I am by character quite unsystematic. That has cost me many things. Yet I do not like to change. Because I find our mother nature is unsystematic. It is only we who have gone out of nature’s way to become systematic to meet our own objectives, the natural consequences of which are environmental crisis, lack of imagination and increased disharmony, conflict and wars. Water flows from hill to plain following haphazard paths. We make dams to make some systemic control, which is the cause of floods to my mind. We destroy un systemic natural forest to build systematic towns and cities, the consequence of which is environmental problem. There need to have science and technology for development and growth as we want but in nature’s way. This will give a better society. I sincerely believe students should be given guidance in natural way, and he has himself to become mastery of systematic integration of knowledge so earned and acquired subsequently. Lots of scopes need to be left out for students to imagine and think. I do not like to teach a kid, as is done in all cases first +, then –, but I prefer to leave—to the kid after teaching him/her +, so that the himself/herself may discover—for which some tips may be given. This is exactly what I try in high-level teaching in institutes/universities. I was wondering to prepare a manuscript following this philosophy of teaching that creates imagination in the students. Accordingly the present book is not that way a so called systematic book, but a book for my way of teaching. The chapters in the book have been prepared in a way to make them independent to each other as far as possible. This will make readers to read chapter as chosen, being independent on other chapters. Besides, I wonder the present practice of writing books that on a same subject there will be separately one for B Tech level, one for M Tech level and another one for research level. This practice is not appropriate for the students of developing countries. For our students, a concise book for all levels is appropriate, and is surely possible to prepare. My attempt does not belie this motivation in this book too. However I do believe my different strokes in writing this book will be best judged by readers. The criticism is welcome. In order to complete this book, I could not give adequate time to my family members as they expected. Their sacrifice is greatly acknowledged. As I write this preface, the remembrance of Late Paresh Maity, science teacher of my higher secondary school suddenly floats in my mind alongwith those of Pabitra Sir and Satish Sir. This may be called telepathy. But I must sincerely acknowledge Paresh Sir’s way of teaching physics to me that is suddenly one of the pillars that has made me what I am today. Indian School of Mines C T Bhunia Deemed University

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Contents

Chapter 1: • • • • • • • • •

1

Introduction ....................................................................................................................... 1 Recent Progress of Computer Technologies .................................................................... 2 Current and Future Communication Technologies ...................................................... 22 Local Loop Transport Technology .................................................................................. 67 Multimedia Communication and Conferencing Standards .......................................... 76 UTN Personal Communication ...................................................................................... 81 From 2G to 3G ................................................................................................................. 82 e-business etc. .................................................................................................................. 84 Knowledge Age and Management .................................................................................. 86

Chapter 2: • • • • • • • • • • • • • • • • • • • •

Information Technology in 21st Century

Network and Internet Technology

99

History Behind Computer Network ............................................................................... 99 Objectives of Networking .............................................................................................. 115 Functions and Scopes of Networks .............................................................................. 115 Different Networks ....................................................................................................... 120 Introduction of Internet ................................................................................................ 137 Protocol .......................................................................................................................... 139 Switching Techniques ................................................................................................... 140 Layered Protocol ............................................................................................................ 142 OSI Protocol ................................................................................................................... 143 HDLC/SDLC Frame of Data Link Layer ..................................................................... 148 Error Control ................................................................................................................. 149 Control Field .................................................................................................................. 191 Other Protocols .............................................................................................................. 193 TCP/IP Protocols ........................................................................................................... 206 IP Header Descriptions ................................................................................................. 207 Unregistered or Private Address Space ....................................................................... 217 Subnet Mask .................................................................................................................. 217 Classless Addressing and Routing ............................................................................... 221 Option Fields of IP ........................................................................................................ 224 Description of TCP Headers ......................................................................................... 228

(ix)

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(x) • • • • • • • • • • • • • • • • • • • •

UDP Headers ................................................................................................................. 233 Address Resolution Protocol ......................................................................................... 241 Proxy ARP ...................................................................................................................... 245 Reverse Address Resolution Protocol ........................................................................... 246 IPv4 to IPv6 ................................................................................................................... 248 How Did IPv6 Come Up ................................................................................................ 252 IPv6 in Details ............................................................................................................... 253 Details of Use of Extension Headers ............................................................................ 256 ICMP .............................................................................................................................. 258 BOOTP ........................................................................................................................... 261 DHCP ............................................................................................................................. 262 IPv4 and IPv6 Address Compatibility ......................................................................... 262 Dual Stack/Tunneling ................................................................................................... 271 Objective Type Questions ............................................................................................. 280 Domain Name Service .................................................................................................. 287 Voice Over Internet or Internet Telephony ................................................................. 289 Technical Problems of Voice Packet Transmission Over Internet ............................. 290 IPv6 for Real Time Services ......................................................................................... 292 Physical Layer Interface ............................................................................................... 293 Transmission Media Option ......................................................................................... 299

Chapter 3: Advanced Error Control Techniques in Network • • • • • • • • • • • • • • • • • • •

306

Introduction ................................................................................................................... 306 Basic BEC Techniques .................................................................................................. 309 Different Modified Techniques ..................................................................................... 312 Sastry’s Scheme and Morris Modification ................................................................... 313 Other Modifications ...................................................................................................... 313 Two Level Coding .......................................................................................................... 314 Parity Selection in Two Level Coding .......................................................................... 316 Packet Combining Scheme ........................................................................................... 317 Modified Packet Combining Scheme ............................................................................ 318 ARQs for Variable Error Rate Channels ..................................................................... 321 YAO Technique .............................................................................................................. 321 Chakraborty’s Technique .............................................................................................. 322 New Schemes ................................................................................................................. 323 ARQ Schemes Under Practical Situations .................................................................. 326 GBN and SRQ Under Different Schemes .................................................................... 327 Issues of Sending Different Signal Waveforms for Repeated Retransmitted Copies .... 328 Application of Multilevel Coding Scheme in Variable Error Rate Channel .............. 329 Majority Technique ....................................................................................................... 330 Analysis of the Majority Scheme for SW ARQ ............................................................ 330

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(xi) Chapter 4: • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • • •

Data/Network Security Techniques and Approaches

355

Data or Information Security Introduction ................................................................. 355 Cryptography ................................................................................................................. 358 Conventional Encryption .............................................................................................. 359 Classical Cipher ............................................................................................................. 359 Substitution Codes ........................................................................................................ 359 Transposition Codes ...................................................................................................... 360 Cryptanalysis of classical ciphers ................................................................................ 360 General Attacks ............................................................................................................. 362 Secret and Private Key Cryptography ......................................................................... 362 Stream Cipher ............................................................................................................... 363 Block Cipher .................................................................................................................. 363 DES ................................................................................................................................ 364 Modes of Operation of DES .......................................................................................... 370 Automatic Variable Key ................................................................................................ 373 Proof of DES .................................................................................................................. 375 Merits and Demerits of DES ........................................................................................ 375 Quantification of Performance ..................................................................................... 377 Triple DES ..................................................................................................................... 379 International Data Encryption Algorithm .................................................................. 380 Advanced Encryption Standard ................................................................................... 382 Comparisons of Secret Key Cryptosystems ................................................................. 385 Modes of Operation of AES ........................................................................................... 386 Limitations of AES ........................................................................................................ 387 Limitations of Secret or Private Key Cryptography ................................................... 388 Key Transport Protocol ................................................................................................. 389 Needham—Schroeder Protocol ..................................................................................... 389 Key Agreement Protocol ............................................................................................... 390 Diffie-Hellman Protocol ................................................................................................ 390 Station to Station Protocol ............................................................................................ 390 Merkles’s Puzzle Technique of Key Agreement .......................................................... 391 Quantum Security ......................................................................................................... 392 Public Key Cryptography .............................................................................................. 395 RSA Algorithm .............................................................................................................. 395 How Secured is RSA ...................................................................................................... 398 Limitations of RSA Algorithm ...................................................................................... 399 Trapdoor Knapsack Problem ........................................................................................ 401 McEliece’s Public Key ................................................................................................... 402 Comparison of RSA and TRAP DOOR Public Key Cryptosystems ............................ 402 Public Key Cryptographic Mechanisms ....................................................................... 402 Digital Signature ........................................................................................................... 404 Digital Signature under RSA algorithm ...................................................................... 404 Check Functions for Authenticity, Integrity and Norepudiation of the Message Content ........................................................................................................... 404

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(xii) • • • • • •

Illustrate the Non-repudiation by Digital Signature of RSA .................................. ...411 Strength of Mechanism ................................................................................................. 412 PGP (Pretty Good Privacy) ........................................................................................... 412 Modern Crypto Systems................................................................................................ 416 Integrated Solution for Error and Security ................................................................. 416 Internet Security ........................................................................................................... 417

Chapter 5: • • • • • • • • • • • • •

Reviewing Information, IT and Looking into Future IT

427

Information and Knowledge ......................................................................................... 427 Proof of Tom Stonier’s Theorem ................................................................................... 431 Tom Stonier’s Theorem With Shannon’s Theorem ..................................................... 432 Proposing Laws of Information .................................................................................... 433 Mass Energy Equivalencey ........................................................................................... 434 Present Imbalance in IT Era, Digital Divide ............................................................... 434 DD Between the Developed and the Developing ........................................................ 436 DD Trend in Future ...................................................................................................... 441 DD Betwen India and China ........................................................................................ 444 DD Within a Country .................................................................................................... 445 DD in Language Zone ................................................................................................... 447 Looking Differently ....................................................................................................... 447 Looking into Future IT ................................................................................................. 452

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1 1.

Information Technology in 21st Century

INTRODUCTION

The basic motivations behind all scientific and technological inventions and discoveries are two: (1) man’s inherent desire to live with the principle of least action and (2) man’s inherent desire to be a master like nature for which they quest for to know what are there in nature’s actions and designs. All the discoveries from the fires to computers conform to the going with the principle of least actions. Man’s aim of becoming the creator or master of all has lead to design or redesign himself or herself which has been manifested in the recent development of cones in laboratory, in continuing research on high speed computing, autonomic computing, quantum computing and in possible designing of intelligent or brainy computer in near future. In the field of communication engineering, its trends of development duly conform to these two basic motivations of discoveries and inventions. To achieve all sorts of communication with least action, the developmental phase of communication has proceeded as: connecting geographically separated but location-fixed machines (conventional wired telephones/fax) to connecting geographically separated but movable machines (chord less /mobile phones) to connecting people rather than machines (communication that supports both man and machine mobility which is personal communication). This is how the total wireless communication is the lust of tomorrow’s communication. In order to achieve the nature like communication, the communication we do in our day-to-day life, PTN/UTN (Personal Telecommunication Number)/ (Universal Telecommunication Number) has evolved out. In the existing communication the connection number changes from location to location and from service to service. We are having separate telephone numbers while at Calcutta than that from while at Delhi. This is not the case in the natural communication. A person is called by his name where he is in Calcutta or in Delhi. A person is called or addressed by his unique name whether it is voice communication or letter communication. Basic motivations behind scientific and technological development have moved the communication research and development on the footings of TOTAL WIRELESS COMMUNICATION and PTN/UTN—in the combined form of Personal Communication Network/Service(PCN/PCS). There are several other parameters including techno-economic and socio-economic aspects that have caused the total wireless communication becoming pillar of tomorrow’s communication; and to name a few are the lower maintenance cost of wireless, easier up gradation and reconfiguration of wireless networks, easier installation of wireless network over difficult regions like over hills and seas, and avoiding threats of theft of costly copper wire used in wired communication. Only existing disadvantage of wireless

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communication is the higher initial deployment cost of wireless networks over wired networks and high error rate probability of the wireless links. But over the time and once the maturity of the wireless technology and its systems is attained, these disadvantage will undoubtedly be the past issues. High-speed communication and integrated services are other two important directions of communication technology. High bit rate carriers like SONET and integrated transport technology ATM are future power of communication technology. In the same conformity of principle of least action and man’s earnest desire to be a master of nature, the knowledge age is believed to follow the current information age. The technical capability and the technology are readily available to transform data into knowledge and that is how there emerge challenges of expanding vision to turning from data to knowledge. Actually knowledge age is the next natural consequence of networked age. In the knowledge age, knowledge workers, knowledge factories, knowledge organizations and knowledge economy will be the rule of law. The main wealth of the knowledge age will be knowledge rather than any physical wealth. The subject knowledge management (KM) is therefore will be key issue in the 21st century. This chapter reviews the growth of computer and communication technologies along with knowledge management that are all trying to merge with human axis (Fig. 1)[1], critically analyze the problems thereon, attempts for possible solution and predicts what is there after knowledge age.

2. RECENT PROGRESS OF COMPUTER TECHNOLOGIES Since the inception of electronic classical computer in the year 1948 by the brand name ENIAC, computer has undergone four generations. Present age is of the fifth generation. Hectic research is going on to make “brainy computers”[2,3]. Worldwide research on optical, chemical and quantum technology is being reported [4-6]. Classical computer was the brainchild of Von Neumann. Classical computer is also known as serial computer. Problems of classical computer were two folds: • How to use its power for general purpose small computing jobs thereby having a cost effective solution and raising the system productivity. • How to raise its power, performance and capacity to tackle extensive, complex numerical jobs (for example design of supersonic aircraft, modeling of global weather etc.) where if a serial computer is used, it may take even a year to many years to solve the problem. The solution to the first problem came in the year 1960, with the introduction of timesharing multi-user concept. This was based on the philosophy of utilization of slowness of human as compared to computer, so that if one user is thinking, the computer can be used by other users (resources sharing by time slice). This provided a means of distribution of the cost of computation over many users. Other early solution to the first problem is “batch system” which remained dominant where large amount of data was processed with minimum human interaction (one operator). But as it was not of interactive type, it lost itself to time-sharing system. One of the answers to the second problem gave the birth of parallel computers, which is the ultimate aim of the fifth generation computing system. A few parallel computers are in operation in world. Parallel computing is to speed up operation. With this in mind the concept

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3

of optical computer was developed. In optical computers it is the light that will carry the signals; and in universe it is the light that has the ultimate speed. Accordingly, non-linear optics emerged as the new frontier of science and technology. The other important deviation from classical computer, that emerged due to technological growth and demand, was the design of “brainy computers”. The chemical computer is a bold step in formulating the “brainy computer”. Optical and chemical computers are now merged under a new field of electronics known as molecular electronics. There are several empirical laws that correlate, govern and predict the technological progress and growth in the last few decades [7-9]. These are: 1. Joy’s law, which states that the computing power, expressed in MIPS (Millions of Instructions Per Second), doubles every 2 years, 2. Ruge’s law estimates that the communication capacity necessary for each MIPS is 0.3-1 Mbps (Million of Bits Per Second), 3. Metcalfe’s law which states that if there are ‘n’ computers in a network, the power of the computers in a network like Internet is multiplied by ‘n’ square times. The law has been applied on Table (1) that lists the growth of Internet users over several years; assuming year 1988 as the reference year, and assuming that in that year the power of a computer was one unit (used for normalization). In that case the power of a computer over different years would be as shown in the table. Assume that each user on average uses only a computer for world access through Internet. Applying the Metcalfe’s law to a lowest extent that the power of individual computer in the Internet is multiplied by square of the number of users in the Internet, the power of computer would be as shown in the last column of the Table (1). From a figure of 0.25 × 1012 in 1988 to 2433600 × 1012 in 2000, a 9734400 (≅107) times increase over a gap of only 12 years! What a future is ahead of ! Super information power or infinite information power! Due to this power, the flexible transport technology, ATM and very high rate carriers like SONET/SDH (Table 2), the requirement of any services at any time at anywhere with a single device and with a single communication number may be possible even through modest Internet, which was basically designed to carry data only. Table 1: Trend in Internet/Computer power Year

Internet Users in 106

1988

0.5

1

0.25

1989

1.3

1.5

2.535

1990

2.4

2

Computer power normalized to year 1988 on standalone condition

Computer power on networking in 1012

11.52

1991

4.4

3

58.08

1992

8.7

4

302.76

1993

14.8

6

1314.24

1994

26.1

8

5449.68

1995

49.2

12

29047.68

2000

195

64

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2433600

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Computer Technology Progress

Brainy Computer Human Axis Year

Personal Communication

Communication Technology Progress

Fig. 1: Trends of Computer and Communication Technology.

Table 2: Bit Rates of Digital Hierarchy North American Type

European Type

Bit rates

Type

E1

2.048 Mbps or 2 Mbps

DS0

T1 or DS1 1.544 Mbps

E2

8.448 Mbps or 8 Mbps (≅ 4 × 2 Mbps)

J1

1.544 Mbps

T2 or DS2 6.312 Mbps (or 4 × 1.5 Mbps)

E3

34.368 Mbps or 34Mbps (≅ 4 × 8 Mbps)

J2

6.312 Mbps (≅ 4 × 1.5 Mbps)

T3 or DS3 44.736 Mbps (or 7 × 6 Mbps), sometimes referred to as 45

E4

139.264 Mbps or 140 Mbps (≅ 4 × 34 Mbps)

J3

32.064 Mbps (≅ 5 × 6 Mbps)

T4 or DS4 (1) 139.264 Mbps (or 3 × 45 Mbps) (2) 278.176 Mbps (or 6 × 45 Mbps)

E5

564.992 Mbps or 565 J4 Mbps (≅ 4 × 140 Mbps)

DS0

Bit rates

Other (used predominantly by Japan)

64 Kbps

Bit rates 64 Kbps

97.728 Mbps (≅ 3 × 32 Mbps)

4. Moore’s laws state that (a) the number of components on an IC would double every year (this is the original Moore’s law predicted in 1965 for the then next ten years), (b) the doubling of circuit complexity on an IC every 18 months (this is known as revised Moore’s law), (c) the processing power of computer will double every year and a half (Moore’s second law which closely resembles to Joy’s law). 5. Law of “Price and Power” that states that over the years the computing, processing, storage and speed up power of computers will continue to increase whereas the price of computers will continue to fall. 6. For a new law of communication, readers may refer to Appendix-A. In table (3), a list of computer generations with power in terms of information processing, storage and speed up factor is given. It is seen that first three laws fit well into the list. In

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pace with increased processing power in terms of volume and speed, and the wide and flexible use of computers, the communication transport technology and transmission media have been developed. Table 3: Computer power over years Generation of Intel processors Processor

Number of Transistors in the chip

Word length in bits

Internal bus size in bits

External bus size in bits

8080

8

8

8

8088

16

16

8

8086

16

16

16

80286

134,000

16

16

16

i386

275,000

32

32

32

i486

1,600,000

32

32

32

32

64

32

P24T Pentium

3,300,000

32

64

64

Celeron

4,000,000

64

64

64

Pentium Pro

5,500,000

64

64

64

Pentium with MMX (multimedia) Technology

4,500,000

64

32

64

Pentium II

7,500,000

64

64

64

In the chip level integration till date, Moore’s laws say the last word. From SSI to ULSI, the trend set (Table 4) by Moore’s law is followed. But beyond ULSI, what is there? The extrapolation of the trend predicts that the future will be the age of molecular dimension inherited by the already established subject of molecular electronics that is based on organic materials rather than inorganic semiconductor. Beyond ULSI, the further integration on a chip will face serious problem from physical constrain like the quantum effect. This may lead to the death of Moore’s law. But another interesting dimension may be added to the cause of the death of Moore’s law. This is based on the law of “Price and Power”. It is said that: “ The price per transistor will bottom out sometime between 2003 and 2005. From that point on, there will be no economic point in making transistors smaller. So Moore’s law ends in” a few years. “In fact, economies may constrain Moore’s law before physics does.” Table 4: Generation of IC integration Generation

Number of components

Small Scale Integration (SSI)

2–64

Medium Scale Integration (MSI)

64–2000

Large Scale Integration (LSI)

2000–64,000

Very Large Scale Integration (VLSI)

64,000–2,000,000

Ultra Large Scale Integration (ULSI)

2,000,000–100,000,000

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2.1 Newer Technologies To go beyond the conventional laws mentioned above, the computer technology has taken a few new direction: (1) recently reported Intel’s Terahertz Transistor, (2) molecular electronics, (3) autonomic computers and (4) quantum computers. 2.1.1 Terahertz Transistor The Intel’s terahertz transistor is reported to be a new method of making transistors with a new class of material to overcome the problem of heat dissipation and quantum effect. This transistor will save power and provide miniature chips. The transistors will stay cooler and be smaller in size but with faster operating speed. This is made possible, as the new method will be based on innovative design that will eliminate leakage. It is believed that by 2007, this transistor will be available in the market. 2.1.2 Molecular Electronics The subject of molecular electronics has emerged as an important area of research and application during 1980’s [10]. The definition of molecular electronics is not unique and simple. Even within a country scientists differ. A leading scientist of the field [11] “molecular electronics can be divided into two main themes: these are molecular materials for electronics (MME) and molecular scale electronics (MSE). The topic of molecular materials for electronics deals with the use of macroscopic properties of organic materials in devices, and includes current and near-term application. In the near-term it seems likely that conductive polymers will offer the prospect of novel electronic devices and that organic materials with pronounced non-linear optical properties will find application in upto electronics. A simplistic extrapolations of the reduction of time leads eventually to the molecular scale i.e. molecular scale electronics.” Prof. Bloor further observed [12] “many regard the quest for molecular scale devices as true molecular electronics. However it can be argued that the distinction between MME and MSE is somewhat arbitrary and that both need to be considered as constituent parts of molecular electronics if the topic is to grow and prosper.” Ashwell, Sage and Trundle [13] defined that “its definition has broadened from electronics at molecular level to include molecular materials with potential electronics and photonic applications.” Peterson defined that “in the most general sense, molecular electronics covers the use of molecular (and hence essentially organic) materials to perform signal processing or transformation function.” However, the famous Link programme of Britain defines molecular electronics as [14]” systematic exploitation of molecular, including macro molecular, materials in electronics and related area such a photo-electronics.” The molecular electronics is therefore to explore the potential application of organic materials and non-linear optics in the field of electronics. It is a highly interdisciplinary field and prospects lies on the successful interaction and co-operation of scientists of different fields like biology, chemistry, computing, physics and electronics. 2.1.2.1 History of molecular electronics In history the concept of molecular electronics dates back to the last century. The familiar example is the use of organic materials in displays. The use of liquid crystals display found in watches, calculators and TV sets in historically patented over fifty years ago [3]. As Prof. Bloor pointed out [15] that “molecular exhibit great variety in their structure and properties from simple diatomic species through to very large synthetic and bio-macro-molecules. It is not surprising therefore that molecules can be found that process unique combination of properties which find application in fields of electronics and opto-electronics.” This idea stimulated work on MME since 1950s. The reduction of size of active electronic device compound problems in

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regard to quantum effects.. At this juncture, molecular electronics, the application of molecular materials in electronics, started exploiting some of the new advanced technologies that may be beyond the scope of the silicon chop. Prof. Bloor explained [16-17] that “the continuing development of silicon micro-electronic devices of smaller size and grater complexity has brought more compact and powerful instrumentation and computing facilities into the laboratory and office. Though silicon technology holds a dominant position the continuing reduction in dimensions of an individual device creates problems both at the fundamental and systems level. On one hand quantum effects must ultimately come into play dissipation and the design of testable architectures are already with us. These pressures lead inevitably to a search for alternatives to current technology that can offer prospects for the realization of devices with even higher densities of active components. MSE is one avenue which is being explored with these targets in mind.” The research and the interest in molecular electronics were mainly initiated by the late Forest Carter who conducted a series of international conferences on molecular electronics [18-20] in 1980s. Prof. Bloor wrote that [21] “organic solids have attracted the interest of materials scientists and solid-state physicists since the 1950s both as alternative semiconductor and because of their optical properties. Strong research groups grew up in the USA, Russia, Germany and France at this time.” Although the progress of molecular electronics has not always been smooth, yet the prospects for the future are good. In this article, we shall review the present position and future aspects of molecular electronics. 2.1.2..2 Molecular Materials for Electronics (MME/M2E) The study of MME is to see the use of molecular materials in key and active roles in electronic and opto-electronic devices and systems. It is based on understanding and use of macroscopic properties of the bulk molecular materials i.e. of the organic materials. The main categories of MME are[22] • Organic semiconductors and metals • Liquid crystalline materials • Piezo/pyro-electric materials • Photo/Electro-chromic materials • Non-linear optical materials/photonics. Organic Semiconductors and Applications Organic semiconductors and metals have been much less studied than their inorganic counterpart. Under MME, a good study is gradually emerging. The major applications of organic semiconductor are in (1) electronic active devices and (2) xerography. Therefore before going to organic semiconductors, the process in amorphous materials is required to be studied. What are amorphous materials? In crystal, atoms or molecules are arranged in a regular structure with periodicity. But in amorphous materials there is no ordered structure. The developments of electronic devices in last few decades were tremendous because the electrical conductivity of crystalline semiconductors such as silicon can be controlled over much order of magnitudes by doping. But [23] “there are a number of areas where the expenses of preparing. These crystals and where the limited size to which they can be grown (at present about 25 cm in diameter) have prevented any very large-area applications. For example,

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crystalline silicon solar cells are widely used in space vehicles for converting sunlight into electrical power, but the economics of their production is such that their use here on earth is relatively limited. Silicon can be prepared very cheaply in large areas by vacuum evaporation or by sputting, but the materials is then amorphous rather than crystalline ……… sine (the) work on doping amphorous silicon (a-Si) was published, there has been a considerable research into and development of this materials, leading to a member of commercial products.” Table 1 [46] shows a progress list. MME makes a study with electronic processes as distinct from ionic processes, in organic crystals. What are organic crystals? By organic we usually mean a compound containing carbon. Almost 90% of 2 millions compounds known to us are organic. But for MME, there is choice and limitation that need a careful study. Till today organic materials have not presented to be a real competitor to the silicon/ inorganic material in terms of active electronics devices. However, during last five years the progress in the synthesis of high purity semiconductor polymers and oligomers is note worthy. Experiments showed that conductive polymers could be employed as either metallic or the semi conducting component of metal-semiconductor junction devices [14]. “semi conducting polymers can be used to produced Schottky diodes [6]. Where the polymer has temperature dependent properties have been observed, with rectifying behavior at room temperature changing to ohmic behavior above 100°C [15]. Burroughes et al. first reported an active polymer transistor in 1988 [16,17]. The important characteristic of this device were: (1) no chemical doping or side reactions and (2) the characteristic of the polymers device was insensitive to disorder. But the major disadvantage of the device was that its maximum operating frequency was limited. This is because the carrier mobility in the amorphous polyacetylene layer is very low. The mobility’s of electrons in semi conducting polymers, amorphous silicon and crystalline silicon are of the order of 10–4, 1 and 103 cm2 /Vs respectively. One can see the large gap between properties of polymers and silicon. However a dramatic lead was done by Frincis Garnier and co-workers [18-19]. They reported a totally organic transistor. This transistor is known as thin film transistor (TFT) or organic FET. This transistor is a metal insulator semiconductor structure comprising an oxidized silicon substrate and a semi conducting polymer layer. It has grater flexibility and can even function when it is bent (disorder is acceptable). The operating speed is still poor. The problem of low carrier mobility of insulating polymer is under active research. The diodes made of semiconductor with rectification’s ratios in excess of 103 have been reported in [23], and light emitting diodes, made in organic semiconductor with external quantum efficiencies in excess of 1% photons per electrons are reported in [16-22]; and organic photovoltaic cells are reported in [19-22]. However, within a short period, a rapid progress has been observed on use of semi conductive polymers and oligomers in electronic devices. If this progress is maintained, in near future it could be competitive to silicon. The field of optical computation starts with the search of a bi-stable optical switch based on non-linear optical properties of materials. Non-linearity can be used for device basically by two techniques: frequency conversion and reflective index modulation. The frequency conversion technique, which is due to second order non-linearity, may be used to second harmonic generation frequency mixing and parametric amplification etc. Refractive index modulation

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particularly Kerr effect which is due to third order non-linearity may be used for optical bistable switches and parallel processing. Till date a few optical gates and all optical bi-stable switches have been reported, but the field is still confined in the laboratories. Yet optical computation is a promising field. Optical computing and processing of information are the important application of photonic. The gain of photonics switching speed (of order of femto second 10-12 ) is many order of magnitudes over that of electronic switching. Optical processing is free from interference from electrical or magnetic sources. “Based on the prospect of three dimensional interconnectivity between sources and receptors of light concepts of optical neural networks that mimic the fuzzy algorithms by which learning takes place in the brain have been proposed and experimentation has begun. Integrated optical circuits, which are counterparts of electrical circuits photons, can provide for various logic, memory, and multiplexing operations. Utilizing non-linear optical effects, analogs of transistors or optical bistable devices with which light controls light have also been demonstrated” [23]. So far nlo materials are concerned, all materials in forms of gases, liquids or solids, exhibit nlo phenomena. However broadly we can defined two classes of nlo materials : (1) molecular materials or organic materials which “consist of chemically bonded molecular units that interacts in the bulk through weak van der waals interactions” and (2) bulk materials and traditional inorganic materials. Today rapid progress and research in organic nlo materials proved to be attractive. The nlo devices utilize two different techniques: frequency conversion and refractive index modulation. Based on letter effects, the developments of frequency converter and light modulator have been reported in [23]. However organic materials are seen to be quite attractive for electro-optic light modulation as “their low -frequency dielectric constant is quit low leading to a small RC time constant, thus permitting a higher bandwidth for light modulation compared to that achievable using inorganic materials.” The application of second order non-linearity needs that the crystal must not be centrosymmetric structure. In centrosymmetric structure the non-linearities, which are vectorial, cancel each other to give zero microscopic effect. This is a stumbling block in the progress of application of second order non-linearity. To solve the problem two approaches are being examined: 1. Use of LB films “with either alternating layers of a polar molecule or molecules which inherently from polar multi-layers, 2. Inclusion of “non-linear optically active molecules in polymer films which are poled with an applied electrical field.” In a single way, a materials with a bulk where, its molecules are non-centrosymmtric nature may be defined as anisotropically oriented over volumes measure in cm3 . “These conditions are best achieved by growing a crystal. The Langmuir-Blodgett (LB) technique is a comparable high tech organic fabrication method, appropriate when the implementation of the function requires a high degree of molecular anisotropy in an extremely thin layer of uniform thickness. For OICs, particularly for single processing, L-B technique offers the possibility to orient molecules with in a thin layer of highly precise thickness. It has thus become an attraction. However films are not the final answers. There are many drawbacks with films namely mechanical softness, limited high temperature range and extremely slow rate of deposition etc. But rapid research is going on L-B film technology and its application in molecular electronics materials both for ME and MSE .

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2.1.2.3 Molecular Scale Electronic (MSE) The quest for an ever decreasing size but more complex electronic components with high speed ability gave the birth of MSE. The concept that molecules may be designed to operate as a selfcontained device was put forwarded by Carter, and he only proposed some molecular analogous of conventional electronic switches, gates and connections [9]. Accordingly Aviram and Ratner first advanced a molecular P-N junction idea. MSE is a simple interpolation IC scaling. Scaling is an attractive technology. Scaling of FET and MOS transistors is more rigorous and well defined than that of bipolar transistor. But there are problems in scaling of silicon technology. In scaling on the one hand propagation delay should be minimum and packing density should be high; on the other hand these should not be at the expenses of the power dissipated. With these scaling rules in minds, scaling technology of silicon is to reach a limit. Another thing is that scaling can due to the quantum nature of physics. At this junction molecular scale scaling technology. Dr. Barker reported in [9] that “change, spin, conformation, color, reactivity and lockand-key recognition are just a few examples of molecular properties, which might be useful for representing and transforming logical information. To be useful, molecular scale logic will have to function close to the information theoretical limit of one bit on one carrier. Experimental practicalities suggest that it will be too easiest to construct regular molecular arrays, preferable by chemical and physical self-organization. This suggests that the natural logic architectures should be cellular automata: regular arrays of locally connected finite state machines where the state of each molecule might be represented by color or by conformation. Schemes such as spectral hole burning already exist for storing and retrieving information in molecular arrays using light. The general problem of interfacing to a molecular system remains problematic. Molecular structures may be the first to take practical advantages of novel logic concepts such as emergent computation and ‘floating architecture’ in which computation is viewed as a selforganizing process in a fluid-like medium.” MSE spans several disciplines and requires a co-ordination of scientists of different group if the subject is to grow and prosper based on cross fertilization of ideas of different subjects. But problem is how can the properties of individual molecules and/or small aggregates be studied? Fortunately day-by-day we are evolving new techniques and methods to tackle this problem. At present we are having technologies like STM (scanning tunneling microscope) AFM(atomic force microscope) and NFOM (near field optical microscope) etc. In addition, submicron lithography, L-B films and adsorption/reaction in 2D/3D are also there. L-B technique is particularly important because it provides one of the few ways of marketing separate electrical connection to two ends of a molecule. A very good illustration of molecular electronics logic and architecture can be seen in [10]. 2.1.2.4 Bio/Chemical Computer A new radical information processing system is being thought of where organic cells or bacteria are to act as the basic element. Living organisms are made of organic compounds. As such thinking function can be easily realized in such system. As scaling will be at biological level, very high-density circuit can be at biological level, very high density circuit can be achieved. Our average brain comprises 1011 neurons ranging in size from 0.2mm linear dimension to

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about 100 mm, each with an average connectivity of 104 giving a crude bit-count of 1011 to 1015. An equivalent artificial brain may therefore be of such dense circuit. Enzymes and proteins are being studied. We should not forget that an example of a natural molecular device. Is the bacterial photo-reaction center. Recent research to produce analogous have been successful through the synthesis of single and complex molecules, which release charge on photo-excitation. This subject of molecular electronics has moved from conjuncture to experimental study and scientific development. With the rapid growth of research and development of few liquid crystals, polymers, L-B films and NLO materials; molecular electronics is now with us. With advances in Physics, Chemistry, Materials Science, Biology and Engineering as our understanding of molecular materials both at microscopic and microscopic level with grow; the field of molecular electronics will prosper. The better understanding of natural system and processes and living organisms, will enhances the capability and potentiality of molecular electronics particularly in terms of its application in radical new computational machines and engineering. Much more work remains to be done. It needs scientific, intellectual and technological challenges on one hand; and Government and Industrial supports on the other hand. The progress of all these will determine actually whether molecular electronics if so, when. But research in molecular electronics and device technology it, will emerge as exciting and frontier fields of science and technology in the current century. The molecular electronics is a revolutionary idea. To attain maximum miniaturization, it is proposed that instead of using transistor’s states, namely ON and OFF to implement 1s and 0s, the characteristics of electrons may be used for the same. For example, the positive and the negative spin be respectively used to implement 1s and 0s. The idea is new. It will take lots of time to mature and to develop the technology. This will be the last resort of miniaturization. The molecular electronics is believed to be based on new organic material technology that may lead to bio or chemical computer. A new radical information processing system is being thought of where organic cells or bacteria are to act as the basic element. Living organisms are made of organic compounds. As such thinking function can be easily realized in such system. As scaling will be at biological level, very high density circuit can be achieved. Our average brain comprises 1011 neurons ranging in size from 0.2mm linear dimension to about 100 mm, each with an average connectivity of 104 giving a crude bit-count of 1011 to 1015. An equivalent artificial brain may therefore be of such dense circuit. Enzymes and proteins are being studied. We should not forget that an example of a natural molecular device is the bacterial photo-reaction center. Recent research to produce analogous have been successful through the synthesis of single and complex molecules, which release charge on photo-excitation. However while the above new technologies aim to attain miniaturization going in line and/or beyond Moore’s law, the autonomous computing technology aims at the economic aspect of technology. 2.1.2.5 Autonomic Computing Consider the computing paradigms of the Internet. Fig. 2 and Fig. 3 show the exponential growth of Internet users and Information Technology. It is therefore understood the need of huge technologists to keep on running Internet without much disruption of services. A statistics says: “At current rates of expansion, there will not be enough skilled IT people to keep the world’s computing systems running. Even in uncertain economic times, demand for skilled IT workers is expected to increase by over 100 percent in the next six years.”

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Growth Rate of Internet Users

Internet users in thousand

250000 200000 India

150000

USA 100000

UK

50000 0 1997

2002

2004

Year

Fig. 2: Growth of Internet Users. IT as % share of GDP in India : Source NASSCOM 3.5 3.15

3

2.87

2.66 2.5 2 1.87 1.5 1

1.45 1.22

0.5 0 1997-98

1998-99

1999-00

2000-01

2001-02

2002-03

Fig. 3: IT growth related to economy.

Under such a scenario, it is not unbelievable to believe that there might be an exponential relationship between the growing complexity and power of the computing systems and the technical manpower required to manage and administer them. A new paradigm to relieve humans of the burden of managing, administering, and maintaining the computer systems, and thereby passing these back to computers is to design “Computers that help themselves”, now known as Autonomic Computers. Consider how we, the humans do act when we face problems. When we are physically attacked, we protect ourselves. This solution uses a biological metaphor. Just as the autonomic nervous system of our bodies monitors, regulates, controls, repairs and responds to hazardous conditions without any conscious effort on our part, so the autonomic computer systems. The autonomous computers are to self control, self monitor, self regulate, self-repair and respond to problematic conditions, again without any conscious effort of humans.

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The autonomous computing technology therefore is a major deviation from the conventional rules like Moore’s law. The aim is not to attain more complex, more integrated, more powerful computers but self healing computers that will be economic in terms of maintenance and operation. The key characteristics of an autonomic computer systems system are: • They should be able to fix failures, and able to configure and reconfigure themselves under varying, undefined and unpredictable conditions so that they prevent system freezes and crashes • The systems should known themselves fully and comprise components with proper identity • The systems should work always in optimize conditions and adopt itself accordingly to varying conditions • The systems should be self healing, self correcting and capable of recovering from common, routine and extraordinary, known and unknown events that might cause some of its parts to malfunction or crash • The systems should be self protective against unwanted intrusion • The systems should be expert to know its environment and the surrounding activity, and act accordingly in order to easy recovery from crashes and interoperations • The systems should adhere to open standards to ensure interoperability among myriad devices • The system should better prevent themselves from failures at first place • The systems should optimize resource in anticipation while keeping its operation hidden to users. The self-managed computers will have four major components (Fig. 4): • Self optimized—components and devices of the system will automatically and continually check their performance and seek to improve the same • Self configurable—components and systems will automatically configure and reconfigure to required adjustments seamlessly • Self healing—system will automatically detects and repairs localized problems • Self protected—System automatically protects itself from intentional attacks

Self Healing Self Optimized Self Configurable Self Protected

SELF MANAGED /AUTONOMOUS COMPUTER

Fig. 4: Autonomous Computer.

2.1.2.6 Quantum Computing The conventional computing is based on the concepts of bits. The bits in the classical computation may have two possible states 0 and 1. The fundamental concept of the quantum computing is

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the quantum bits, referred to as qubit. Two possible states of qubit are |0> and |1>. Like binary bits of the classical computing, all possible superposition of qubits are possible. Therefore, a two qubit system has four computational states, namely |00>, |01>, |10> and |11>. With Moore’s Law being saturated, it is expected that quantum computers will be one of the future solutions for high speed and high power computing. A few theoretical work has been reported but practical implementation is yet to reach. However an important milestone in application of quantum computers has been achieved due to pioneer work of Bennett et al in quantum cryptography in the area of data security. BOX 1 Quantum Computing: a bit review QUANTUM GATES The information processing in the quantum computing has a component of qubit manipulation. The qubit manipulation is performed by unitary operations. A quantum logic gate is a device that performs a particular unitary operation on the selected qubits at a given time. There are infinite numbers of single-qubit quantum gates unlike only two (identity and the logical NOT) in classical information. The quantum NOT gate performs |0> to |1> and vice versa analogous to classical NOT Two-qubits quantum gates performs many possible unitary operation, an interesting subset of which is |0> <0| ⊗I + |1> <1|⊗ U where I single-qubit identity operation and U is some other single-qubit gate. Such gates are called controlled gates as action of I or U on the second qubit is controlled by whether second qubit is in state |0> or |1>. This gives to define controlled NOT, CNOT gate as: |00> |00> |01> |01> |10> |11> |11> |10> this shows that: (a) second qubit undergoes NOT if and only if the first qubit is in state |1>; (Fig. 1) (b) the effect of CNOT on states |x> |y> may be written as : x ® x, y → x⊕y for the reason of which this gate is also called. XOR gate Fig. (1). X

Xo = X

Y

Yo

Y

Notes: (a) x-wire means NOT but controlled by o-wire. (b) Each horizontal line represents a single qubit evolving in time from left to right A symbol on a line represents a single qubit gate. (c) A vertical line connects two or more qubits. Symbols on two qubits connected by a vertical line represent a two-qubits gate on those two qubits. CNOT GATE: The output, Yo at x-wire is controlled by the input, X of the o-wire. When input to o-wire is |1> , the output Yo of the x-wire is NOT of its input state, Y. XOR GATE: Whatever the first qubit, the output second at the x-wire is always XOR of the two input qubits. Fig. 1: CNOT/ XOR gate.

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Other logical operations do require additional qubits. The most popular three qubits gate is Controlled- Controlled NOT gate/CCN or C2NOT gate (Fig. 2). This gate is also known as Toffoli gate that demonstrated that the classical version is universal for classical reversible computation. A gate is reversible when for a given output; one can reconstruct the input(s). The output of the gate on o-wire can be described as: (a) if third qubit is in state |0>, then output is AND of two other qubits. The effect on the input states |x> |y> |0> is x → x, y → y and output → x.y. (b) the effect on the input state |x> |y> |1> is that output is XOR of x and z, (c) the effect on |1> |1> |z> is that output is not of z. X

Xo = X

Y

Yo = Y

Z

Zo = Z

Note:

Sum generation

(X.Y)

to mean toggle control. Others as in Fig. 1. Fig. 2: Controlled Controlled NOT/ CCN gate.

It has been argued that any logic circuit can be made of only CN and CCN gates only. For example, Fig. 3 illustrates a half adder circuit.

X

Xo = X

Y

Yo = X

Z

Zo = (X.Y) Carry generation

Fig. 3: Half adder using CN and CCN gates.

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Table 2 Superposition

In general this means that two things can overlap with each other with interfering with each other. In quantum mechanics two electrons can overlap with each other making a combined waveform that is a set of amplitude probabilities.

Principal ideas of quantum physics

Energy is in discrete units. Photons are each a discrete bundle of energy. A photon of characteristics frequency,n carries a quanta of energy equals to h.n where h is the Planck’s constant. The particles in quantum physics behave both a particles and waves. The state vector of particles obeys Schrodinger wave equation.

Uncertainty principle

It is impossible to measure both the position and the momentum of the particles at the same time. More accurately one is measures, less precisely the other is known.

Entanglement

With entanglement the systems are correlated in a way that does not involve force and the restriction of the speed of light is not applicable .

QUANTUM TELEPORTATION Teleportation is by which an object or person while physically remains present in one place, is made to appear as a perfect replica somewhere else. The classical or conventional approach of teleportation is illustrated in Fig. 4. Fax machine is an example of teleportation machine. Till recently the quantum teleportation was assumed impossible as it would violate the uncertainty principle of quantum mechanics. The uncertainty principle prohibits any scanning or measuring process to extract all the information in an atom or such object. As the more accurately an object is scanned, the more accurately the object is disturbed that may ultimately lead to complete change of the original state of the object even before the whole of information is extracted to make a perfect replica of the original one. But quantum mechanics has an aspect known as entanglement. If outside force is applied on two atoms, the aspect of entanglement occurs whereby the second atom can take the properties of the first atom. Thus if left alone, an atom will spin in all directions; but the instant it is disturbed it chooses one spin, or one value; and at the same time, the second entangled atom will choose an opposite spin or value. This allows learning the value of qubits without actually looking at them, which could collapse them back into 1’s or 0’s. Sending Station Original object, A physically present at location, P Receiving Station A replica of the original Object A is Generated / Received at a location, Q away from P

A is Scanned or Processed

Send Data

Original, A remains intact at the sending location

Fig. 4: Classical Teleportation/FAX.

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The property of the EPR (Einstein Podolsky Rosen) or “entanglement’ has made the quantum teleportation possible hurdling the principle of uncertainty. Fig. (5) illustrates the quantum teleportation. In the process, part of the information of the original object is scanned out. The un scanned part of the information is passed viz EPR effect into anther object C. the object C was never in contact with the original object A. the intermediary object or the delivery vehicle, B conveyed the un scanned part of information from A to C. It is now possible to apply treatment on C to make it as A before A, was disrupted by the scanning process. So a real transportation is achieved in C rather than replica. Sending Station Original object, A physically present at location, P Receiving Station A replica of the original Object A is Generated / Received at a location, Q away from P

A is Scanned or Processed

Apply treatment

Send Data B

Original, A becomes completely disrupted

C

Entangled pair, B and C

The intermediary object, B

Fig. 5: Quantum teleportation.

QUANTUM CRYPTOGRAPHY The disadvantage of key distribution in secret key cryptography can be removed with the aid of quantum technology. If key distribution problem is solved, the use of Vernam technique will be best technique of security. In order to solve distribution problem, use of quantum channel for sending information about key is being explored. In quantum mechanics one cannot measure something without causing noise to other related parameter. For example Hysenberg’s uncertainty principle state that ∆x.∆m.= constant. Thus if ∆x. is changed, ∆m is bound to change. An ideal quantum channel supports transportation of the single photon. Thus a single photon can represent a bit 0 (zero) or 1 (one). The phase or state of polarization of photon may be used for identifying the 0 or 1. For example. Photons with 0° and 90° of polarization may therefore be treated as bit 0; and photons with 45° and 135° (also known as – 45°) of polarization may be assumed as bit 1. Data security through quantum channel is under active research in the UK and USA. Some positive breakthroughs have been made by Charles Bennet of IBM Research at Yorktown Heights, New York, and by Gilles Brassard at the University of Montreal. If, in the example discussed earlier, Alice wants to send Bob the secret key as required in the Vernam cipher, she can send the key, say of N bits, through quantum channels. Bob will be instructed by Alice to detect the photons (bits) from the quantum channel starting from a given time. There may be some transmission loss, and Bob may be able to detect some fraction of photons or bits. Bob will have to inform Alice over a telephone as to which photon he has seen. For this, they may share both a common and variable key. For instance, if Alice sends

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11110000 as the key, and Bob replies that he has seen the first, seventh and eighth photons (starting from the leftmost bit), then their common key shall be 100. Alice can send data haphazardly using different polarized photons. Alice can do so (Fig. 6) either on rectilinear basis: When a horizontal polarized photon represents a 0 and a vertical polarization represents a 1 Or on diagonal basis: When a – 450 polarized photon represents a 0 and a + 450 polarized photon represents a 1.

=1

=0

Fig. 6: Use of polarization for representing 1s and 0s typically.

Alice haphazardly uses both to send qubits (Fig. 7). Bob will haphazardly try to filter out the qubits. For the purpose of qubits detection Bob will use a polarization beam splitter. The polarization beam splitter is a device that allows the photons of orthogonal polarization to pass through but shunts the photon of other polarization. The quantum nature dictates that: (a) the same basis beam splitter will pass the received same basis polarized photons, but (b) the rectilinear beam splitter will pass the received diagonally polarized photons either as vertical or horizontal polarization with equal probability and the diagonal beam splitter will pass the received rectilinear polarized photons either as vertical or horizontal polarized photons with equal probability. This will provide the different combinations of Alice’s sent photons and Bob’s detected photons. Therefore when both Alice and Bob use the splitter on same basis they with correctly communicate qubits, but when they use on different basis, the chance of matching between sent and received qubits is 50%. Bob now tells Alice (over conventional method, say telephone, as there is no need to keep secret these) how he used the beam splitter to detect received qubits. Assume Bob’s choice was as rectilinear, rectilinear, diagonal, rectilinear, diagonal (Fig. 7). Bob does not announce the results of detection. Alice replies publicly (means over conventional method as there is no need to keep this secret) Bob, which times her choices of base match with Bob’s choices. Then they use the qubits of those instant when they use same base (in those instant they correctly communicate the bits), and ignores the bits of other instants. The matching bits (Fig. 7) generate the secret key for the session.

(Rectilinear) (Diagonal) (Rectilinear) (Rectilinear) (Diagonal)

(a) Alice sends qubits to Bob randomly (we have taken only 5 qubits for illustration)

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(Rectilinear) (Rectilinear) (Diagonal) (Rectilinear) (Diagonal)

(b) Bob measures the received photons using random polarization basis Same Base

Different Base

Different Base

Same Base Same Base

(Uncertain) (Uncertain) (Correctly (Correctly (Correctly detected by detected by detected by Bob) Bob) Bob)

© Alice and Bob Communicate and identify locations whether they correctly used the polarization base. COMPARE (a) with (b). BUT THEY KEEP SECRET THE POLARIZATION OF SENT OR RCEIVED PHOTONS. 1

Ignored

Ignored

1

1

(d) Correct bits are taken for key. Bits of other positions are ignored. So the key in this example is 111. Fig. 7: Key exchange between Alice and Bob.

Should any eavesdropper attempt to intercepted photon transmission; there shall be garbage with the key accepted by Alice and Bob. This is because the quantum theory ensures that, without changing the phase of the photon, an intercepted photon cannot be retransmitted. Therefore, a change in the polarity of the photon will let Alice and Bob immediately known of an interception. The scheme of sending information at the one-photon-per bit level as proposed by IBM research and research of university of Montreal reported that “to send the key, the transmitter (Alice) tells the receiver (Bob) that the plans to send n bits (photons) starting at a given time. Alice than sends the bits by randomly switching the phase in the transmitter between 00 to 1800; this switches the output in the receiver between “0” and “1”. Although transmission and detection losses mean that Bob will only see a small classical communication channel (the telephone, for example) to tell Alice which photons he has seen—but not which detector he has seen than in. This allows Alice and Bob to share the same random number. For example, Alice uses ten photons to send the random number 1001011101; Bob replies that he only received the second, fifth and last photon; therefore they have shared the random number 001. However, it is conceivable that an eavesdropper could intercept the signal, copy Alice’s message, and send it on to Bob without either Alice or Bob realizing. One way to overcome this, and ensure absolute security, is for both the transmitter and receiver to use non-orthogonal measurement bases. In other words, Alice sends parts of the message by switching the transmitter phase between 900 and 2700, say, and other part by switching between 00 and 1800. When the Bob and Alice are using the same base, the system works as before. However, if Alice is using 00/1800 and Bob is using 900/2700 (or vice versa), the message is meaningless

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- a photon that Alice sends as a “0” has a 50% chance of being received as a “1” and vice versa. Therefore when Bob tells Alice which photons he has received, he now also says which base he was using and Alice must tell him if that is a valid photon (i.e. one which was sent and received when they were both using the same base). Paul Townsend of British Telecom, working with the Malvern group, recently demonstrated self-interference of short light pulses, containing on average 0.1 photons, down 10 km of standard communications fiber using the technique.” There is anther technique to minimize the hacking by Eve. The technique is known as privacy amplification protocol. In the protocol, Alice randomly chooses pair of bits from the key they have got over quantum channel. Then she performs XOR on the pairs. She then tells publicly to Bob on which bits the XOR operation was made but not the results. Bob then performs the XOR operation on the bits that Alice informed him. Alice and Bob then replace the pair with XOR results to design the new key. The is illustrated as below: (a) Alice and Bob have secret key 111 as in Fig. 7. (b) Alice chooses first and second bit as pair and she informs these to Bob publicly. She gets XOR result 1 ⊕ 1 = 0 and keeps it secret. (c) Bob performs XOR on the informed bits and get the result 1 ⊕ 1 = 0. (d) Alice and Bob both replace the pair by XOR result. So their new key = 01. (e) Note that even if Eve definitely knows one bit of the chosen pair, until & unless she gets the result of XOR (which Alice and Bob never communicates) she can not replace the pair for hacking the key. Quantum computer is very promising. It has numerous advantages over classical computers, namely in terms of speed (parallelism inherent in quantum computer), power consumption (nearly at the half of classical computer due to superposition), and tackling of computational problems here to impossible with conventional computers. The quantum computer will be based on quantum logic gates based on quantum circuit, and the technology for these is even prior to the infancy stage. On the other two problems of the quantum computers have been identified. It is estimated that the quantum error correction will generate more power than the chips can dissipate; the technology of quantum computer may not be so easy to develop. The problem of decoherence intervals that measure how long a qubit can maintain synchronized waveform to represent either 1 and 0 simultaneously. The decoherence time is estimated on average to be less than 1 microsecond. The challenge remains how to increase this interval time. Yet there is no stop, and shall not be a stop in development of quantum computer. We will be wrong to think that the quantum computers will replace classical computers. The quantum physics has not replaced the classical physics. They co exist each within their own parameter.

2.2 Quantum Security The disadvantage of key distribution can be removed with the aid of quantum technology. If key distribution problem is solved, the use of Vernum technique will be best technique of security. In order to solve distribution problem, use of quantum channel for sending information about key is being explored. In quantum mechanics one cannot measure something without causing noise to other related parameter. For example Hysenberg’s uncertainty principle state that Dx.Dm.= constant. Thus if Dx. is changed, Dm is bound to change. An ideal quantum channel supports transportation of the single photon. Thus a single photon can represent a bit 0 (zero) or 1 (one). The phase or state of polarization of photon may be used for identifying

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the 0 or 1. For example. Photons with 0° and 90° of polarization may therefore be treated as bit 0; and photons with 450 and 1350 of polarization may be assumed as bit 1. Data security through quantum channel is under active research in the UK and USA. Some positive breakthroughs have been made by Charles Bennet of IBM Research at Yorktown Heights, New York, and by Gilles Brassard at the University of Montreal. If, in the example discussed earlier, Alice wants to send Bob the secret key as required in the Vernam cipher, she can send the key, say of N bits, through quantum channels. Bob will be instructed by Alice to detect the photons (bits) from the quantum channel starting from a given time. There may be some transmission loss, and Bob may be able to detect some fraction of photons or bits. Bob will have to inform Alice over a telephone as to which photon he has seen. For this, they may share both a common and variable key. For instance, if Alice sends 11110000 as the key, and Bob replies that he has seen the first, seventh and eighth photons (starting from the leftmost bit), then their common key shall be 100. Eavesdropping can be tackled by sending photons with different phases. For example, the bit 0 may be represented by a photon having a phase of 0° or 180°, and the bit 1 can be denoted by a photon with a 90° or 270° phase. When Bob uses, he will be able to detect the bits correctly. Alice can send data haphazardly using different polarized photons. Bob will haphazardly try to filter out the bits. After the operation, Bob will inform Alice over the telephone of the timings and the state of filter used by him. Alice can then inform him at what instances they have used the same state of filters. Based on this exchange of information. Bob and Alice will get to know their keys. Should any eavesdropper attempt to intercepted photon transmission; there shall be garbage with the key accepted by Alice and Bob. This is because the quantum theory ensures that, without changing the phase of the photon, an intercepted photon cannot be retransmitted. Therefore, a change in the polarity of the photon will let Alice and Bob immediately known of an interception. The scheme of sending information at the one-photonper bit level as proposed by IBM research and research of university of Montreal reported that “to send the key, the transmitter (Alice) tells the receiver (Bob) that the plans to send n bits (photons) starting at a given time. Alice than sends the bits by randomly switching the phase in the transmitter between 00 to 1800; this switches the output in the receiver between “0” and “1”. Although transmission and detection losses mean that Bob will only see a small classical communication channel (the telephone, for example) to tell Alice which photons he has seen— but not which detector he has seen than in. This allows Alice and Bob to share the same random number. For example, Alice uses ten photons to send the random number 1001011101; Bob replies that he only received the second, fifth and last photon; therefore they have shared the random number 001. However, it is conceivable that an conceivable that an eavesdropper could intercept the signal, copy Alice’s message, and send it on to Bob without either Alice or Bob realizing. One way to overcome this, and ensure absolute security, is for both the transmitter and receiver to use non-orthogonal measurement bases. In other words, Alice sends parts of the message by switching the transmitter phase between 90° and 270°, say, and other part by switching between 0° and 180°. When the Bob and Alice are using the same base, the system works as before. However, if Alice is using 00/1800 and Bob is using 90°/2700 (or vice versa), the message is meaningless—a photon that Alice sends as a “0” has a 50% chance of being received as a “1” and vice versa. Therefore when Bob tells Alice which photons he has received, he now also says which base he was using and Alice must tell him if that is a valid photon (i.e. one which was sent and received when they were both using the same base). Paul Townsend of British Telecom, working

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with the Malvern group, recently demonstrated self-interference of short light pulses, containing on average 0.1 photons, down 10 km of standard communications fiber using the technique.” But remember Moore’s laws here to stay for at least anther decade! BOX 2 ILLION-TRANSISTOR IC—Hope or Hype Since the inception of digital electronic in the brand name of ENIAC in 1948, the computer has gone through a number of generations, and it is now in the fifth generation. The so vast and rapid changes of five generations of the computer technology just over a period of 50 years results in one hand the reduction of size & cost of computers and on the other hand the tremendous increase in the processing power & capacity of computers. The credit for these is due to IC (Integrated Circuit) technology. Out of many others the famous empirical laws known as Moore’s Laws, basically govern the pattern of growth of computers and that of IC technology. Mr Gordon Moore, Head of Research & Development of Fairchild coined these laws around 1965. Moore’s laws state that (a) the number of components on an IC would double every year (this is the original Moore’s law), (b) the doubling of circuit complexity on an IC every 18 months (this is known as revised Moore’s law), (c) the processing power of computer will double every year and a half (Moore’s second law). Presently ICs are made of around 250 million transistors. If Moore’s law continues to hold good, it is predicted that by 2010 ICs will be made of billion transistors. The threats to the survival of Moore’s laws are heat dissipation and quantum effect that is a physical limit to IC integration. Several predictions were therefore earlier made for imminent death of Moore’s laws. Contrary to these predictions, Moore’s laws are surviving and hold true for IC integration. Recent two research reports have further showed confidence of survival of Moore’s laws for al least another few years. A survey conducted jointly by IEEE (Institute of Electrical and Electronics Engineers) and the Response Center Inc of USA (a market research firm) over the fellows of IEEE showed that 17%, 52% and 31% respondents respectively predicts the Moore’s laws continuation for more than 10 years, 5-10 years and less than 5 years. The average predicted life term for the laws is then about 6 years. Moore’s laws existence if then guaranteed up to 2009, by the time of which following the laws the billion transistors IC will be a reality. The expectation of realizing billion transistors IC by 2010 has been further brightened by the current research of Intel expanding Moore’s laws. Mr Pat Gelsinger ‘s vision of expanding Moore’s laws includes Intel’s 90-nanometer fabrication process. Although a several alternative technologies, namely quantum computing, bio computing, molecular electronics and chemical computing are under investigation as possible replacement digital computing, the year 2010 may achieve the landmark of billion transistor IC, an another leap forward in IC technology— really a high hope and not a hype.

3. CURRENT AND FUTURE COMMUNICATION TECHNOLOGIES 3.1 Personal Communication Personal Communication is poised to bring a revolution in communication. Personal Communication shall be wireless, service independent and like natural communication. It shall support all sorts of mobility. Active research is going on in this field all over the world. As

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of today, personal communication is seen as a sum total of existing wireless communications like cellular communication, paging, mobile satellite services, VSAT (Very Small Aperture Terminal), wireless LAN, Wireless Internet etc., although ultimately personal communication shall be a UTN (Universal telecommunication Number) service. Personal communication is believed to be a total wireless communication. It is aimed to provide global coverage and to serve any sort of information like voice, data, messaging etc., to anywhere, at any time[24]. At any location or anywhere could imply home, office or in-transit or any other place. Personal communication has two different attractions. First it is total wireless and thereby it supports both man mobility and machine mobility[25]. Personal (or man) mobility and terminal (machine) mobility have distinct and separate characteristics[26]. For personal mobility a person need not to carry a terminal and needs to have a personal communication number. Personal communication number is typically a UTN. (UTN is discussed later). For terminal mobility, a person needs to carry a terminal and needs to be within its radio coverage. With personal mobility, all sorts of communication can be made through the personal number. A caller getting connection of a callee through callee’s personal number may opt for a particular terminal like telephone or fax, for the session. In terminal mobility different types of communication need different numbers and different call sessions. For example for mobile fax, we need to have separate numbers, and for mobile telephone we need to have another separate number. For personal communication, any device like conventional home phone, cellular phone, key phone, fax and pager can be used. Service wise therefore, personal communication is much more flexible, portable, accessible, and reachable compared to wired communication. The philosophy behind personal communication is unique. Personal communication is for connecting people rather than machines. Personal communication is believed to provide a single Universal Telephone Number (UTN) or Universal Personal Telecommunication (UPT) number to a subscriber for all sorts of communication at any time, anywhere. Today, we cannot reach many people most of the time, at most places even though they have a number of telecommunication devices like telephone, fax, telex, e-mail, etc. With single UTN, a subscriber can communicate all over the world. In to-day’s communication scenario, a person has many different numbers for communication at different locations (one’s telephone number in Kolkata is different from that in Delhi) and for different uses (fax number is different from home telephone number). With the two of the above stated characteristics, personal communication will seem like almost natural communication. In our day-to-day natural communications (acoustic communication), we use wireless mode and single name for addressing caller (or callee). A person’s name does not change whether he/she is in Kolkata or in New York. Personal communication is postulated as universal communication. Technology development, standardization, system development, performance analysis and spectrum allocation etc., for personal communication is actively underway. Experts and scientists view personal communication from different angles. One group views personal communication as a distinct and separate total mobile communication solution. Others see personal communication as a migration of the existing conventional wireless communication with enhanced features. The latter view is quite balanced one. Therefore, as of today personal communication can be viewed as a combination of various existing wireless services and new proposed services like UTN[27-28]. Personal communication includes the platforms of each of the existing services like cellular, wireless PBX, Centrex, cordless both home and public, CT2 and wireless LAN etc. Partial application of existing services by personal communication includes paging, SLMR (Special Land Mobile Radio), PSTN (Public Switched Telephone Network), VSAT, and Common Channel Signaling System-7 and ISDN (Integrated Services Digital Network)[29-30]. Personal communication shall fully include future services

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like next generation cellular or TGMS (Third Generation Cellular System) and UTN services. As it is expected that personal communication shall operate globally using the concept of UTN, the required switching and processing systems for personal communication shall be huge and complex. Intelligent capabilities of switching and nodes are a must. On the basis of this, we can define personal communication as an intelligence-based and natural-like communication. Wireless transmission can take place using different frequency bands. An overview of different frequency bands is given in table 5. The frequency allocation to some of the wireless communication is given at table 6. Table 5: Different Frequency Bands and their applications Frequency Band

Wavelength

Name of the Band

Usual Transmission Line Covering the band

Application

<30 KHz

>10 km

Very Low Frequency (VLF)

Twisted Pair

30-300 KHz

10-1km

Low Frequency (LF)

Twisted Pair/Coaxial Long Radio waves/ Used in Cable submarines because these waves can penetrate waters and follow earth’s surface

300 KHz-3 MHz

1 km-100 m

Medium Frequency (MF)

Twisted Pair/Coaxial Radio Waves/AM between Cable Long 520 KHz to 1605.5 KHz

3-30MHz

100-10 m

High Frequency (HF)

Twisted Pair/Coaxial Short Radio Waves/ AM Cable/Radio waves with 5.9 MHz to 26.1 MHz

30-300MHz

10-1 m

Very High Frequency (VHF)

Twisted Pair/Coaxial FM between 87.5-108 cable/Radio waves MHz/TV between 174-230 MHz

300MHz -3 GHz

1 m-100 cm

Ultra High Coaxial cable/Radio Frequency (UHF) waves/Micro waves

TV between 470-790 MHz

3-30 GHz

100 cm-1 mm

Super High Frequency (SHF)

Micro waves

Analog mobile phone (450465 MHz)/ Digital GSM (890-960 MHz)/DECT at 1880-1900 MHz/Fixed Satellites Service in C-band (4/6 GHz), Ku band (11/14 GHz) and Ka band (19/29 GHz)/Digital TV is planned at 470-862 MHz.

>30GHz

Less than 10 micrometer

Extra High Frequency (SHF)

Optical fiber/Infrared links

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Table 6: Frequency Bands in some of the important wireless applications US Mobile Phones

AMPS, TDMA, CDMA 824-849 MHz 869-894 MHz GSM, TDMA, CDMA 1850-1910 MHz 1930-1990 MHz

Europe

Japan

GSM 890-915 MHz 935-960 MHz 1710-1785 MHz 1805-1880 MHz

PDC 810-826 MHz 940-956 MHz 1429-1465 MHz 1477-1513 MHz

Cordless Telephones

PACS 1850-1910 MHz 1930-1990 MHz 1910-1930 MHz

CT + 885-887 MHz 930-932 MHz CT2 864-868 MHz DECT 1880-1900 MHz

PHS 1895-1918 MHz JCT 254-380 MHz

Wireless LAN

IEEE 802.11 2400-2483 MHz

IEEE 802.11 2400-2483 MHz HIPERLAN 1 5176-5270 MHz

IEEE 802.11 2471-2497 MHz

3.2 Cellular Communication The world of wireless communication actually began in the USA around 1930s when the American police started using radiotelephones for communicating with the field offices. Public radio applications like PLMR (Public Land Mobile Radio) and SLMR (Special Land Mobile Radio) gradually developed. In early 1980s four more wireless services were introduced. These are AMPS (Advanced Mobile Phone Systems) developed in Bell laboratory in 1980, airphone services, cordless services and telepoint. AMPS is the earlier example of cellular communication. AMPS and for that purpose, cellular communication, migrated from wide area radio communication system. Wide area radio transmission technology is the earliest form of mobile communication. The objective of wide area radio transmission technology of the early days is to cover as large area as possible, with a single base station. The single base-station of the wide area cell is equipped with high tower antennae. The transmitter of the base station is very high powered. The configuration of the system once designed is fixed. On the other hand, in cellular concept, smaller cells, each with a low powered base station are used. In cellular concept, the objective is to increase the customers (specifically subscriber density per MHz of allocated spectrum) rather than the coverage area. This objective is met by the concept of cell-splitting and frequency re-use. These have been illustrated with examples in reference. Cell-splitting and frequency re-usable plan are changeable; and hence configuration is flexible and changeable. In cellular system many cells are used. Each cells cover relatively small coverage radii of the order of 0.5 km to 10 km, compared to 50 km to 100 km of the early day mobile communication system. Small cells of cellular communication are formed in splitting large cells of previous mobile systems. For frequency re-use, the cells are clustered into a group with say, k number of cells per cluster. Allocated band of cellular may be divided into k, and each of the divided bands may be allocated to a cell for communication of a mobile base pair. Frequency re-use may be defined as use of some carrier frequency to cover different cells separated by a distance

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so that co-channel interference does not cause problems. Carrier re-use follows well-defined rules described in standard literature.

3.3 First Generation Cellular The first generation cellular is analog. AMPS is the standard analog cellular used in USA, Canada and Australia etc. Other first generation cellular standards are TACS (Total Access Communication System) use in UK, Austria, Spain and Italy; C-450 of Germany and RTMS (Radio Telephone Mobile System) to Italy etc. All the first generation cellular systems use frequency modulation for speech and frequency shift keying technique for signaling. Band sharing among users is done by frequency division multiple access (FDMA) technique. In USA a total of 50 MHz is allocated in bands of 824-849 MHz and 869-894 MHz for analog cellular communication. In AMPS system, each channel is 30 kHz wide. Hence 832 channels are provided in AMPS. Frequency modulation with 8 kHz deviation is used in speech, and frequency shift keying with 10 kbps is used for signaling. In AMPS, cluster size is either 12 with omni directional antennas or 7 with directional antennas per cell. In Japan, a total of 56 MHz is allocated for analog cellular communications in band of 860-885/915-940 MHz and 843-846/898-901 MHz. NTT (Nippon Telephone and Telegraph) employed a system in 1979 using bands of 925-940 MHz and 870-885 MHz respectively for uplink and downlink. 25 kHz channel spacing was used and 600 duplex channels were provided. The signaling rate was 300 bps. This system was upgraded in 1988 with reduced channel spacing of 12.5 kHz and increased signaling rate of 2400 bps. Frequency interleaving technique was used. Number of channels increased to 2400. For farther and more information, reference can be seen.

3.4 Second Generation Cellular Second generation cellular systems were evolved with digitization, digital technology and digital signal processing. With digital techniques in hand and application, it was seen that TDMA (Time Division Multiple Access) and CDMA (Code Division Multiple) could be other viable and potential alternatives to FDMA. Digital techniques offer a number of advantages over analog techniques, namely flexibility (digital systems can support mixed and/or integrated communication and wide range of services), reliability (digital systems are less noise/error prone; can support security easily), cost effective (one transceiver can be used in base station to serve a number of users in digital systems whereas in FDMA, this number increases with number of users) and reduced complexity etc. Digital cellular is known as second generation cellular. Digital cellular technology and techniques are well standardized. GSM (Global System for Mobile Communication), ADC (American Digital Cellular), IS-54 (developed by Electronic Industries Association–TIA of America and JDC (Japanese Digital Cellular) are examples of second generation cellular standards. They are respectively used in Europe (and some parts of Asia including India), USA and Japan. GSM was actually standardized in 1982 as ‘Group Special Mobile’ by CEPT (Conference European Post & Telecommunication). In GSM, 50 MHz band is allocated for cellular communication in the bands of 890-915 (mobile transit) and 935960 MHz (base transmit). Each radio channel is allocated 200 kHz. Thus there can be maximum 25 MHz/200 kHz = 125 carriers. As a convention, only 124 carriers are used. First 200 kHz in uplink and last 200 kHz in downlink are not used. Minimum and maximum number of carriers per cell can be respectively 1 and 15. TDMA is used with 8 slots per radio channel. Each mobile transmits periodically in its slot and receives in the corresponding slot. Each slot is of 0.577 msec duration. Each frame duration is 0.577 × 8 = 4.615 msec. GSM supports full rate operation

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at 22.8 kbps with 8 slots per frame as well as half rate operation at 11.4 kbps with 16slots per frame. For voice communication speech coders compatible with both the rates are available. For data communication various asynchronous and synchronous services at different rates of 9600, 4800 and 2400 bps are specified for both full and half rate service operation. These data services interface to audio modems (like V.22 bis or V.32) and ISDN (Integrated Services Digital Network). GSM can also support connectionless packet switched network X.25, Internet and group 3 FAX (Fly Away Xerox). GSM has recently extended to include ‘group calls’ and ‘push to talk’ services. Extension bands of GSM which are yet to be explored are 880-890 MHz for uplink communication and 925-935 MHz for downlink communication.

3.5 DCS 1800 DCS 1800 is an extension of GSM. In DCS 1800 standard uplink and downlink bands are respectively 1710-1795 MHz and 1805-1880 MHz. It is working at around 1800 MHz which is higher than that of GSM. Higher frequencies always have more penetration power. Therefore, compared to GSM, DCS 1800 system is better in terms of interference and fading. DCS 1800, besides third generation cellular is preferable to in personal communication.

3.6 CDMA Cellular Code Division Multiple Access (CDMA) cellular is another example of second generation cellular. It is a good competitor of TDMA cellular. In TDMA cellular, different users’ signals use the same frequency band, but are distinguished by different codes. The codes are spread codes. This is done basically by two techniques: frequency hopping and direct sequence. In frequency hopping, the transmitter jumps from one narrow band frequency to another according a sequence mutually known to transmitter and receiver. Thus several data bits may be sent at different frequencies. In direct sequence each bit (‘yes’ or ‘no’) of data is represented by a sequence of bits to be transmitted in the same time. The length of sequence is known as chip ratio. For example, one user may code ‘yes’ and ‘no’ states respectively as ‘0000’ and ‘1111’; whereas another user may do so by codes ‘0101’ and ‘1010’ respectively (chip ratio is four). With such spreaded code, signal is disturbed over a wide band. As signal is spread over wide band, effectively signal looks like a noise and becomes indistinguishable from noise. The CDMA system is a secure communication; and this makes it advantageous over TDMA cellular. Another plus point of CDMA is capacity. The capacity of CDMA is more than that of TDMA cellular. Other aspects of CDMA cellular are parallel with TDMA cellular and for that purpose GSM. In [14] it was shown that narrow band propagation, path loss (path loss of say, TDMA) should be applied to wide band path loss (path loss of CDMA). IS-95 is EIA/TIA standard of CDMA cellular system used in AMERICA. The basic user channel rate is 9.6 kbps. It is spread by a factor of 128 and channel chip rate becomes 1.2288 Mchips/sec.

3.7 Wide Area Connectivity In cellular, the wide area coverage or the world wide coverage is done through a basic connectivity scheme. In this scheme a group of basic stations are connected to a MSC (Master Switching Center). MSC is connected to other public and national or inter national networks. Through base stations the mobiles access the network over radio links. Base stations provide overall management and controls switching between radio channels and TDMA time slots in order to connect the mobile to MSC. Through MSC a mobile can connect to other mobiles of other cells as well as connect to subscribers of all the public national or international networks

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connected to MSC. A mobile of one MSC can connect to any other mobile of other MSC s via MSC-MSC switching.

3.8 Continuous Operation Originally a mobile belongs to a base station and is assigned a number for communication. This original number is its number for communication kept stored in HLR (Home Location Register) of MSC. When the mobile is within the coverage area of his original base location permanent number is made use of for communication. A mobile can cross its base region and enter into other base regions (foreign) while talking or communicating. In such situations to maintain continuous operation it is required that foreign base stations should take control of visiting mobile. That is, for continuous operation the control of mobile shall pass from original base station to visiting foreign base station. The pass over technique is called ‘hand off’ operation. Hand off is decided upon comparing the signal strengths received by mobile from the original base station and the foreign base station. As the mobile proceeds to cross the area of original base, the received signal strength from original base gradually diminishes; while the received signal strength from the foreign base station gradually increases. The cross over instant of the signal may be taken as the time of hand off operation. However to avoid falls due to noise some hysteresis is often used for cross over decision. On hand off operation a visiting mobile is assigned a temporary number for communication; and the said information is kept stored in VLR (visiting Location Register) of MSC for farther and future management and control. The hand off operation and technique are equally applicable when a mobile roams from one foreign location to other foreign location i.e. when a mobile crosses over boundaries to boundaries.

3.9 Cordless Telephone First generation cordless is analog. In USA, analog communication cordless is allocated 46.647.0 MHz (base transmit) and 49.6-50 MHz (handset transmit). Ten frequency pairs are used in these bands. Frequency modulation is used for voice. In Europe first standard that was used for cordless telephone is known as CT0. Eight channel pairs are used in this standard near 1.7 MHz (base transmit) and 47.5 MHz (handset transmit). The CEPT developed a standard for analog cordless, which is known as CT1. The bands used in CT1 standards are 914-915 MHz (base transmit) and 959-960 MHz (handset transmit). Forty 25 kHz duplex channel pairs are used in these bands. CT1+ standard was later developed with bands 885-887 and 930-932 MHz with provision of 80 channel pairs. It may be noted that CT1+ bands are chosen to avoid overlapping with GSM bands. In Japan, for analog cordless telephones using FM, 89 duplex channels are provided near 254 MHz (handset transmit) and 380 MHz (base transmit). Digital cordless is known as second generation cordless. Digital cordless is like digital cellular to some extent. Cordless is usually for walking indoors and outdoors. Naturally, cell size, antenna height, mobile speed, handset design complexity and handset transmitter power, in cordless are less compared to those of cellular. CT2 is the first standard of digital cordless in Europe. CT2 is allotted bands 864-868 MHz; and it can support 40 FDMA channels with 100 kHz spacing. In CT2 voice is digitized with 32 kbps ADPCM (Adaptive Differential Pulse Code Modulation) encoder. CT2 can also support data up to 2.4 kbps through speech codec, upto 4.8 kbps with increased error rates and higher data rates using 32 kbps voice channel. Telepoint concept is a migration of CT2 technique. Telepoint is wireless pay phone service. Another standard of digital cordless is DECT(Digital European Cordless Telecommunication). It uses TDMA with 12 slots per carrier for each upward and downward

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communication. With TDMA we have earlier seen in case of cellular communication that multiple users can simultaneously communicate with a single transceiver. The same is true for DECT also. It uses 32 kbps ADPCM technique for voice digitization. In addition DECT can support telepoint, wireless PBX and RLL (Radio Local Loop). In Japan, HS (Personal Handy phone System) is the main standard for digital cordless. PHS uses TDMA. Each channel has a width of 300 kHz. 77 channels are permitted in the band of 1895-1981.1 MHz. 37 carriers within band1895-1960.1 are allocated for home and office cordless; and 40 carriers within band of 1906.1-1918.1 MHz are allocated to public cordless. Digital cordless in USA was developed by Bellcore (Bell Communication Research) with a title WACS (Wireless Access Communication System). Actually PACS (Personal Access Communication Service) is now in use. It is a combination of WACS and PHS. In North America, ISM (Industrial, Scientific and Medical) bands like 902-928 MHz, 2400-2483.5 MHz and 57255850 MHz are in use for digital cordless.

3.10 Wireless Data Trend is towards wireless. Wireless communication offers a number of advantages including high performance-cost ratio. Cellular communication and cordless communication are basically for voice communication although they can be used for data communication for voice communication messaging. Wireless data networks are basically designed for packet mode communication. Cordless is for synchronous services (uses circuit switched techniques), whereas wireless data communication is asynchronous in nature (uses packet switched techniques). But we shall see in IEEE 802.11 standard, that wireless LAN has been proposed to provide both the asynchronous and synchronous services. BOX 3 Know thy Elegant Ethernet

1. INTRODUCTION One of the hottest topics of IT is the Local Area Network (LAN). LAN bears an indispensable role of service to information community. LAN provides basically a shared data access of an organisation, which has several systems, and nodes distributed geographically, logically and physically. The three main physical attributes–limited geographic scope (in the range of 0.1–10 KM [1], low delay or very high data rate (over 1 MBPS [02], and user’s ownership, make LANs substantially different from conventional computer networks. Moreover, while Wide Area Network (WAN) and Metropolitan Area Network (MAN) allow user in network to access the shared databases, LANs go a step ahead and allow users to have shared access to many common hardware and software resources [3] such as storage I/O peripherals and communication devices. For example, a costly high resolution laser printer is usually shared by users in a LAN, and all users in a LAN use an inexpensive single transmission medium in a multidrop environment as well as they use whenever required a single bridge or gateway to communicate with other homogeneous or heterogeneous network respectively. LAN is hence a resource sharing data communication network that is usually used to connect computers, printers’ terminal controllers (servers), terminals (keyboard NDU), plotters, mass storage units (hard disk) and any other piece of equipment (exam. Word-processing machine) that has some form of computer connectivity. LAN is to solve for “MY” problem [4] of “80/20 rule” [5] of communication in a cost-effective scope in an office, factory, university and such relevant environment.

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However PABX (Private Automatic Branch Exchange) differs from LAN in that unlike LAN, PABX user a separate pair of wires (transmission medium) to connect each device (or extension), low bandwidth (limited to that of telephone line) and rugged hardware switching for interconnection. The communication in LANs is peer to peer and not via intermediaries as with WANs and MANs. MAN’s coverage is from a few miles to 100 miles and WAN’s coverage is from 100’s miles to 1000’s miles [6]. These entire three networks follow layered architectural standard protocol like 7 like 7-layer ISO-OSI protocol or SNA protocol etc. for interconnection strategies [6]. LANs continue to be driving force to implement future’s white hope of digital wall socket [7], which will act like today’s electricity socket and telephone socket. The digital wall socket is to be used in handing explicitly low or high data rate devices like copying machines, word processing machines, facsimile displays. VDU, keyboard, microcomputers/PC, large computers etc. This may ultimately lead to 100 percent paper less “Office-of-the-future” and 100 percent automated “factory-of-the-future” with “diskless” managers, administrators and engineers etc. One of the most successful LANs is Ethernet. Ethernet was the most popular LAN in 1987. As per Forrester Research Inc, [5], in U.S.A Ethernet covers 33 percent of LAN market with IBM token ring lagging behind at 22 percent. Dataquest estimated that Ethernet had covered 52 percent of installed LANs U.S.A is Ethernet hottest now? Whatever may be the answer to this question, it is a fact that Ethernet is still today very popular and will continue to be so at least for some time to come. This paper will make a thorough review of Ethernet.

2. Ethernet Historically, Ethernet was developed by the Xerox Corporation on an experimental basis [8] around 1972. Based on this experimental experience, the second-generation system was soon developed by the Xerox Corporation in late 1970’s [9]. Around 1080-81, under a joint effort of DEC (Digital Equipment Corporation), Intel and Xerox, an update version of Ethernet specifications (table I) [8] was designed. This historically leads to development of IEEE (Institute of Electrical and Electronics Engineering Inc) 802 standards (table II) [4,6] of LAN in reference to 7-layer OSI-ISO (Open System Interconnection of International Standards Organisation) the LIC (Logical Link Control) is covered by IEEE 802.3 standard at MAC actually specify the accessing mechanism, physical level covers the electromechanical connectivity at network medium, LIC and MAC of LAN jointly form the data link of OSI-ISO protocol standard. Nowa-day Ethernet is available from many vendors [10]. Such Ethernet is as per IEEE 802.3 standard. These are actually “Ethernet-like” [11] networks. However, all LANs covering IEEE 802.3 standard are not Ethernet. But all Ethernets cover IEEE 802.3 standard. Table 1: Specification of Ethernet Parameters

Experiment Ethernet

Industrial Commercial Ethernet

1. Data rate

2.94 MBPS

10 MBPS

2. Maximum end-to-end length coverage using repeaters/bridge

1 KM

2.5 KM

3. Maximum segment length

1 KM

500 M

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4. Data encoding technique

Manchester

Manchester

5. Co-axial cable impedance

75

50

6. Co-axial cable signal level

0 to +3 volt

0 to – 2 volts

7. Transceiver cable connectors’ size

25 and 15 pin D series

Only 15 pin D Series

8. Preamble

1 byte of a pattern of 10101010

1 byte of a pattern of 10101010

9. Size of CRC (Cycle Redundancy Check)

2 byte

4 bytes

1 byte

6 bytes

10. Size of address field

31

Table 2: IEEE 802 Standard Standard for MAC & Physical layer

Access Technique and Topology

Transmission medium with allowed data

Basic application area

802.3

CSMA/CD with BUS topology

Broad band: Co-axial cable with 1 MBPS/ 5 MBPS/10 MBPS/ 20 MBPS Base Band : Co-axial Cable with 1 MBPS

Office Automation (OA)

802.4

Token passing with BUS topology

Broad : Co-axial cable with 1.5444 MBPS/ 5 MBPS/ 10 MBPS 20 MBPS. Base band: Co-axial cable With 1 MBPS/ 5 MBPS 10 MBPS.

Manufacturing Automation (MA)

802.5

Token passing with RING topology

Base band: Shielded twisted wire pair with 1.4 MBPS. Co-axial cable with 4 MBPS/ 20 MBPS 40 MBPS.

Process real time application

802.6 802.7 802.8 802.9

Yet to be finalized -do-do-do-

Yet to be Finalized -do-do-do-

MAN Broadband LAN LAN with fiber optical LAN in ON (Integral service digital network)

802.2 standard is for LIC of LAN. 802.10 is for network security.

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2.1 Features of Ethernet Why Ethernet is so popular? This is due to some of its important features. The most appealing features of the Ethernet are its protocol simplicity, and the relative low-cost and elegant implementation of LAN system which meets the following desirable characteristics [6,7] of a local networking facility. • High flexibility i.e easily adaptability when devices are system to be added or removed. This is due to the bus topology and the cable tapping facility of Ethernet • The transmission medium and access control is easily extensible with minimum service disruption. • High reliability, which assures the continuation of the operation of the network in failure of one or more active element (node) like PC, terminal or workstation etc. This is due to the passive feature of Ethernet cable. Moreover, there is no centralized control but distributed control in Ethernet. • The traffic will be bursty in nature. In office and engineering environment, nature of data is infrequently bursty [10] and ironically Ethernet, was specially made for office automation, although not in general.

2.2 Components and Operation of Ethernet The Ethernet is itself a hardware system. Ethernet can connect typically a maximum number of nodes of 100 per segment [5] and 1024 per total Ethernet [10]. An Ethernet LAN must have Ethernet cable, transceiver, and interface unit, control unit, the user system (Fig 1) and terminals. Two types [12] of ‘co-axial cable popularly known as “thick Ethernet” and “thin Ethernet” is used, mainly as backbone Ethernet. On this back bond cable, the communicating systems and peripherals are attached (tapped). Taps may be intrusive where the cable is cut for tapping or may be non-intrusive where the cable is cut drilled and a tap added without hampering the operation of the network. The most common Ethernet, the baseband Ethernet is tapped nonintrusively, whereas the broadband Ethernets used intrusive tapping using T-junctions scheme. Baseband Ethernet is an implementation where the entire bandwidth of backbone cable is used only for Ethernet communications. Singles in cable are not modulated signal. Thick Ethernet cable resembles a marking every 02.4 meters usually by black ring around cable to show where the taps go. However, thick wire co-axial cable has maximum length limitation of 500 meters and thin wire co-axial has limitation in the range from 189 meters to 1 km depending upon the vendors of transceivers and controllers. Ethernet may also run on twisted pair under certain restriction and on fiber. The length of twisted pair may range from 20 meters to 100 meters. Ethernets on fiber optic medium have length restriction in the range of 30 meters to 5 km. In some cases, thin wire Ethernet may be required to be connected to a thick wire Ethernet. Thin wire cable may be connected to thick wire though a barrel connector. In such case, the restriction on segment length will follow the formula [5]. (3.28* thin wire length) + thick wire length
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straight Manchester coding ensures simple synchronization and a dc value. At any instant cable can be in any one of the three states: transmitting a 1 bit (high followed by low), transmitting a 0 bit (low followed by high) or idle state (0 volts). The high and low level are represented by respectively + 0.85 volts and – 0.85 volts. However, Ethernet using differential Manchester coding is also there [6]. Such 10 MBPS baseband Ethernet actually uses a signaling rate of 20 MHz due to the adoption of differential Manchester encoding. This encoding actually uses ¹ bit times to transfer 1 bit of information and a clock single. By this time, you may probably be wondering of why Ethernet is called Ethernet. It was once thought that “Ether” a hypothetical passive universal element is there to bound together the entire universe and its all parts. And as you see that this LANs transmission medium is a passive Ethernet that is bounding the “smart” devices in a net. This is why the name “Ethernet” was adopted. The Ethernet is a broadcast LAN. All nodes can listen each and every message transmitted on the net. Transceiver is another important component of any LAN. It is carped securely onto the Ethernet cable so that its tap makes contact with inner core. Transceiver is available in many different shapes, sizes and price-ranges, but they all provide users’ devices to communicate with the cable. They also contain electronics circuit that handles carrier detection and collision detection too. A transceiver is so named because it allows simultaneous transmission and reception. A transceiver is fairly a dumb system. It transmits data, receives data and detects collision and notifies the same if occurred to the controller. Transceiver cable (maximum length is 50 meter) contains usually five numbers of individual shielded twisted pairs. Two of these pairs are used for data in and data out. Two more are similarly used for control signals in and out. The fifth pair is not always used, and it is used to allow the node to power the transceiver. Some transceivers allow upto eight nearby computers/workstation/users’ terminals to be attached to them to reduce the number of transceiver needed. For example DEC has developed special box (DELNI–Digital Ethernet Local Network Interconnect) that allows upto eight systems to connect to the box, and a single Ethernet transceiver taps the eight systems onto the main cable. DELNI has the ability to work star alone and emulate an eight node Ethernet cable. When the systems are no more than 50 meters away from DELNI or there are no more than eight co-located system that require being on an Ethernet, DELNI is cost-effective than eight transceivers and cable. The disadvantage is that DELNI is self-powered. So failure of DELNI will fail eight nodes to access network. The interfacing unit detects data and accepts the data if it is mean for this address. It also creates and checks the CRE for error correction and recovery. The controller unit (is a firm wire or sift wire device) transmits data frames to and receives data frames from transceiver via interfacing unit. It also buffers the data and retransmits it when collision occurs, and determines the retransmission interval (which varies with load etc.) and other aspects of network management. For a complete network, one has to procure the components of LAN, network software and hardware and communication software (e.g. Netware 2.2, super LAN, MS net). Now, the basic components of Ethernet are discussed. Next thing is that how does the Ethernet is the accessing technique known as CSMA/CD (Carrier sense multiple access/collision Detection). There are many different types of CSMA technique [12], the technique adopted in Ethernet is 1-persistent CSMA/CD. The problem of the non-persistent strategy is that after the current transmission and line is idle. The alternative to this technique is “1-persistent” where

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the nodes continuously sense the line and transmit data as soon as it is free. The CSMA/CD is a simple and straightforward way of providing every user a chance to transmit whenever it has something to do. The concept behind CSMA/CD may appear to be derived from a technique used when people are talking in a mass gathering or meeting. If no one is talking, one people may start talking. If two or more people start talking at the same time, collision occurs, and both stop and wait for some random time before again starting to talk. In Ethernet if any node wishes to send data to another node on the network, the source listens to see if the line is free (quite/idle). This is called carrier sensing. If the cable is idle, source node starts transmission. Some times it may so happen that two or more station accidentally may start transmission at the same time. The collision is also possible in some other cases. For example if two nodes separated by a distance of propagation time t, both start transmission at an interval of time t, there will be collision. When collision will occur, transmitted data will be corrupted. A mechanism to detect collision is used by adopting the technique of “listen-while-transmitting”. In this scheme at the source node, the transceiver’s transmitting unit while sending the data, the receiving unit is listening to the data that is being sent. If the transceiver detects that the data received by receiving circuitry do not match with that transmitted by transmitting circuitry, it senses the occurrence of collision and accordingly sends a message to the controller of node. If there is match, the transmission process is allowed to go on. On receiving a collision-detection signal, controller stops sending data, and sends a burst of noise on the line(Jamming) to assure that the other nodes sending data listen a collision. All collision-detecting stations back off on detection of collision. The controller than waits for a random time before at empting for retransmission. For this a random generator is used. However, the mean wait is initially equivalent to an end-to-end round trip delay on the cable (which is about 2µ see for 500 meter co-axial cable). However, in case of second time collision, the controller doubles the previously generated random number there by ensuring double of the mean delay of first collision and so on (doubling operation) on repeated collision. Usually random generated is counted from assigned number to zero for measure of delay. The doubling operation is allowed for a prescribed number of times, which are usually 16. After that the controller sends an error message to the host (system manager) notifying the occurrence of multiple-collision. Due to this collision and retransmission scheme, 100 percent channel utilization is not achieved. Ethernet, however, come close to 100 percent due to CSMA/CD technique, which polling and other techniques cannot achieve. The minimum Ethernet packet size (64 bytes) and maximum Ethernet cable segment length and propagation time, used together, guaranteed, that by the time the last bit of information is transmitted, the source node can accurately detect a collision i. Any other node attempt transmission at the same time. However, if the utilization rate of the cable is low (i.e. load on the network is low), collision is rare, and the mean delay time rarely exceeds its minimum value of one end-to-end round-trip delay. When utilization is high (i.e. traffic load becomes heavy), collision becomes more common. Due to this feature, controller dynamically changes the retransmission interval. This is why doubling operation is there in use. When data is being transmitted, all nodes “hear” the data. On examining the first (after preamble) 6 bytes (address field) of the data packet, nodes may determine whether the data is destined for itself or not. If the message is for itself, it passes the message to the users’ device through controller. Otherwise it ignores the message usually. But why is CSMA for Ethernet? Because of distributed nature of the random accessing technique, they are well suited to LANs where simplicity of operation and flexibility are most important. Besides, since a large bandwidth is available in LAN, LAN under such accessing technique can be operated at a relatively low

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loading avoiding unstable [13] conditions. However, the performance of CSMA/CD is inversely proportional to the end-to-end propagation delay [14]. Thus Ethernet for OA can use CSMA/CD most appropriately.

2.3 Ethernet IC (integrated circuit) chips The following is a brief Ethernet IC chips [6, 15] that may be used to design an Ethernet : Vendor

Controller/Interface Chip

Intel

82586 (controller) 82501 (interface) DP8390 DP8790 DP8341 DP8342 7996 7990 8003 8023 LANCE

National Semi conductor

Advanced Seeg Tech AMD/Mostek/Motorola

2.4 Application of Ethernet Ethernet historically and traditionally is used in office automation (OA). Today some organizations are experimenting with video on Ethernets as well as high-resolution graphic access technique by which disk-less workstation may access shared disk structure. Ethernet are also used in laboratories and industries, robotics applications, factory automation, process control and many other non-office applications but in rare cases. Only consideration for adoption of Ethernet in non-office applications is its tolerance of interference from electrical motors, electro-magnetic radiation and other sources of distortions. But its use of CSMA/CD accessing technique (which is probabilistic in nature), a node in the network may have to wait for arbitrarily long period to send a message. Moreover, IEEE 802.3 standard does not have priorities in accessing scheme. This makes it unsuitable in which important message should not be delayed for unimportant frames to pass. These two factors reserve the application of Ethernet in manufacturing factory automation (MA) and in real time process control system. However, while in office the typical required response time is 02 to 10 second, in factory and process control the same is respectively in the ranges of 0.5 to 02 sec. and 0.1 to 0.5 sec. Ethernet can meet a response time of 02 to 10 sec. Ethernet is hence best for OA. For MA, LANs covering IEEE 802.4 standards are suitable.

2.5 Limitation of Ethernet The limitation of the Ethernet from application point of view due to non-deterministic (probabilistic) accessing has already been discussed. Next is that Ethernet does not perform well under heavy load condition. Due to randomness both in data arrival and service, tests have shown that Ethernets can utilize only 90 to95 percent of available resources, under a full load condition. Maximum throughput of 10 MBPS, 500 meter Ethernet with a propagation aped of 2 X 108 m/sec is only 9.96 MBPS [6, 16]. Ethernet does not guarantee of delivery of

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message, as there is no scheme of sequence number checking, missing message re-transmission requests and other such facilities.

3. MODIFICATION OF ETHERNET 3.1. Improving Ethernet for MA The problem of load balancing [17] in CSMA/CD technique can be achieved to a large extent if each station on getting a transmission access is restricted to transmit only a fixed ore-assigned number (pay, P) of packets (non-exhaustive mode) [16]. After transmitting P packets, the station has to back off, for a time, which must not be less than the time required for a bit to end-to-end of round of the bus. After passing off this time, the station can check the carrier further, and the process repeats. Priority in CSMA/CD can be achieved by assigning each station a priority number. Any station when transmitting data, may transmit its priority byte after say each q packets (q>p). Any other station which is desirous to send urgent message, if sees that transmission is going on, may check the priority is less than its priority, it will distort the priority. The on going transmitting station not getting back the proper priority byte will stop, immediately transmission to allow the higher priority station to access. However, if checked priority is greater than its priority, it has to wait for free carrier. A modified and deterministic Ethernet is already there in France-defence department [5]. This is of course a proprieratory item.

3.2 Ethernet for Data and Voice ISDN (Integrated Service Digital Network) is becoming more and more attractive to communication engineers. In spirit with goals of ISDN, a concept of ISLN (Integrated Service Local Network) [17] was introduced. But why? Statistics show that about 15 percent of their office time is spent by senior managers on telephone, and not more than 3 percent of the same is used in handling data oriented jobs. Besides, “real managers don’t use terminals” [19]. But today, of course, they do. Therefore in a complete and cost effective OA system, the integration of voice and data is an essential requirement. In any organization, why shall be there one PAPX (for telephone) and one LAN (for data). However, early problem of LAN design was to communicate data, but the real problem is to provide users’ requirements of both data and voice communication. The pioneer vendors of Ethernet can examine whether Ethernet can be extended to cover ISLN requirement in either of two technique [19] : (i) conventional voice + data upto facsimile or (ii) upto full moving video.

4. EXTENDED ETHERNET A number of Ethernet segments may be connected together (Fig. 1) via repeater or bridge [5,12,20]. A repeater consists of some sort of microprocessor (like Intel 8088, Motorola MC 68000) and memory etc. They are standalone units. They repeat everything what is received from any segment to other segment and vice versa. They connect two Ethernet segment via transceiver. Bridge, on the other hand, store and forward the intended data only from a source segment to a destination segment. Bridge is made of some sort of processor, storage, buffers and a set of software.

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User device

Segment-2 TR

Terminal Controller

TR

CI

Repeater

Controller Interface

maximum 2.5 meter TR

CI

Transceiver cable (max50 m/15 ft)

Printer Top

Terminal

TR Original segment

Terminal TR

User device

TR

CI Controller Interface

TR

terminal either (may be twister wire pair, Co-axial cable, Fibre or radio link)

Terminal Service Like DELN

TR = Transceiver PC

User

TTY

Single Ethernet Segment, max. 500 m/1500 ft

Bridge

TR

TR Segment-3 TR

TR

Gateway

CI

WAN or MAN

Plotter

Fig. 1

5. CONCLUSION A number of important considerations of Ethernet have been highlighted. Ethernet is seen to be very effective for OA. If the next generation of Ethernets is to be developed, they must be done in a direction to extend the application to MA utilizing the proposed suggestions in paper.

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References 1. 2. 3. 4. 5. 6. 7. 8. 9. 10. 11.

C. David Tsqo, “A local area network architecture review”, IEEE Communication Magazine, Vol. 22, No. 8, pp 7º, Aug.’1984. D.D. Clark, K.T. Pogran and D.P. Reed, “ An introduction to local area network”, Proc. IEEE, Vol. 66, No. 11, pp. 1497-1517. Nov. 1978. John E. McNamara, “Local area Network”, Prentice Hall of India, Ch. 1, 1991. Stephen P.M. Bridge, “Low cost local area Networks”, Galgotia Pub. Pvt. Ltd. Ch 1, 1990. Bill Hancock, “Designing and implementing Ethernet Networks”, QED information Science, Inc, 1989. Paul J. Fortier, “Handbook of LAN Technology”, McGraw Hill Inc., NY, 1989. James Martin, “Computer networks and distributed processing”, Prentice Hall, Inc, Ch. 26, 9181. John F. Shoch, Young K. Dalal, David D. Redell and Ronald C. Crane, “Ethernet”, Advances in Local Area Networks, IEEE Press, NY. pp 29-48, 1987. Timothy A Gonsalves, “Measured Performance of the Ethernet” Advances in Local area Network, IEEE Press, pp. 383-387, 1987. William L Schweber, “Data Communication”, McGraw Hill, Intl. Ch. 11, 1988. Neil Willis, “Computer Architecture and Communications”, Paradigm Pub. Ltd., U.K., Ch. 14, 1988.

3.11 Wireless LAN Wireless LAN offers wireless data communication for a limited geographical area. Wireless LAN is like wireless PBX for data. It solves ‘My Problem’] of any organization utilizing the 80/ 20 rule of communication. Wireless LANs are meant for private or organizational uses where wired communication is impossible or impractical or not desirable or expensive ( examples are historic building, trading floors, manufacturing floors, conventions etc.) and/or where some sort of mobility is required (examples university environment, conference room, hospital environment). Wireless LANs are aimed at data rates of 1 Mbps or more. Basically there are two forms of wireless LAN : radio LAN and Infrared LAN. IRLAN is less popular than radio LAN due to : IRLAN can cover wide area but it is almost twice in cost of that an equivalent radio LAN. IRLAN requires license for use of spectrum. Radio LAN uses ISM spectrum for which license may not be required. Unlicensed radio LAN s are available in USA in the bands of 902928 MHz, 2.4-2.4835 GHz and 5.725-5.85 GHz. IEEE committee for standardization of radio LAN has proposed to use 2.4 GHz band (discussed later). IRLAN is only for point to point communication. Radio LAN is much more flexible and can be used in multiple users communication. IRLAN is short ranged (if it is made wide ranged, it becomes costly), radio LAN is long ranged. Architecture wise, both radio LAN and IRLAN assume one of the two basic technologies : infrastructure or ad hoc. Infrastructure topology is the most common in radio LAN. Under infrastructure network stations (computers fitted with adapters/transceivers) communicate with each other under a coverage area of an access point, as well they communicate with any other station in the network through backbone wired network. Backbone network is accessed via access points. In IEEE 802.11 standard, access points are known as base points; and backbone network is known as distribution system. Access point is a combination of transceiver and data bridge. Each access point provides a certain coverage area. Number of access points required for an infrastructure network, thus depends on the required coverage area. Infrastructure topology is useful in covering a building or campus or an institute under radio communication.

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In ad hoc network stations independently communicate with each other and there is nothing like access point for communication through backbone network. Ad hoc network can be either temporary or semi-permanent. Semi-permanent networks are used for a few months and useful for companies, which move frequently. Field construction companies, military camps on war days may use semipermanent ad hoc networks. Temporary networks are used for a day or for a few hours of business. They may be used in sharing files, databases in a company meeting or convention. There are two important standards wireless LAN. IEEE is developing IEEE 802.11 standard, which is proposed to be used in USA. HIPERLAN (High Performance Radio LAN) is the standard developed by European Telecommunications Standard Institute and is for use in Europe. HIPERLAN standard has already been ratified by CEPT. IEEE 802.11 draft standard defines three different physical layers : a) 2.4 GHz ISM band with frequency hopping spread spectrum radio, b) 2.4 GHz ISM band with direct sequence spread spectrum radio and c) infrared light 2.4 GHz ISM band has been allowed both in USA and Europe for IEEE 802.11 version LAN; whereas Japan has allocated the band 2.471-2.497 GHz for IEEE 802.11 LAN. Japan has allowed such a narrow band in 2.4 ISM in order to provide radio LAN in medium data rates of 256 kbps to 2 Mbps where spread spectrum technique is used. Japan has allocated another band near 18 GHz for high rate of 10 Mbps or more radio LAN where QAM (Quadrature Amplitude Modulation), QPSK (Quadrature Phase Shift Keying) are used. In frequency hopping system, 79 and 23 different frequencies are used respectively in USA/Europe and Japan for data transmission under IEEE 802.11 scheme. In direct sequence the processing gain is proposed to be 10.4 dB in IEEE 802.11 draft standard. Frequency hopping system can support large number of channels compared to direct sequence scheme. Frequency hopping is also having superior performance when interference is high. However, direct sequence is simpler in design and implementation. Service wise, IEEE 802.11 has proposed to serve asynchronous and time sensitive (synchronous/isochronous) services. In radio LAN when an access point is shared by all stations, all stations use same hopping/sequence pattern. As such there is always a fair chance of interference and collision. Hidden node problem of radio network on the other hand has a tendency to increase collision. When two transmitters send data to a single receiver, single receiver can hear to transmissions; but transmitters cannot hear each other. This is known as hidden node problem of radio system that depends on physical sensing of carrier. Thus a good medium access control (MAC) strategy is essential In radio system. In IEEE 802.11 standard, MAC is CSMA/CA (Carrier Sense Multiple Access/Collision Avoidance) rather than CSMA or CD (Collision Detection) used in Ethernet. Radio Technique does not allow the collision detection mechanism. In CSMA/CD technique, when a station senses a free carrier it backs off transmission for a random amount of time. Thus if more than one station detects free carrier at the same time , due to random back off periods, collision may be avoided. For tackling hidden node problem, in IEEE 802.11 scheme, two controls frames, RTS (Request to Send) and CTS (Clear to Send) are used. Theses are like RS-232-C transfer protocol. HIPERLAN differs from IEEE 802.11 on number of accounts. IEEE 802.11 does not support multi-hop communication. No access point or station can act as a data router or relay point. HIPERLAN does not support multi-hop communication by the way of cellular architecture. It is targeted for higher data rates than IEEE 802.11 and may support 23.5294 Mbps. That is why a large and dedicated band of order of 150 MHz (5.150-5.300 GHz) near 5 GHz another band of 17.1-17.2 GHz near 17 GHz are allocated to HIPERLAN. HIPERLAN is also aimed to be indistinguishable from wired LAN of Ethernet and to support some sort of isochronous services. For modulation, Gaussian minimum shift keying is used. A (31,26) BCH mode is used for error control. It aims to achieve BER (Bit Error Rate) of 10-3 or less for fair service. MAC in

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HIPERLAN is different from both CSMA/CD of IEEE 802.11. In HIPERLAN accessing scheme, if a station senses free medium for 1700 bit times, it can transmit immediately. If not, channel accessing is done through three phases of prioritization, elimination and yield. HIPERLAN MAC can reduce chances of collision to a less than 3 per cent. IRLAN works on IEEE 802.3 and IEEE 802.5 protocols. IRLAN is based on line of sight technology, and hence it can support high data rates up to 10 Mbps for Ethernet configuration and up to 16 Mbps for token ring configuration. IRLAN is costly and hence some vendors are hopeful to go through the technology. An association of vendors has made their own standards for IRLAN. The IEEE standards for different LANs are 802.3 for CSMA/CD Bus LAN, 802.4 for Token Passing Bus LAN, 802.5 for Token Passing Ring LAN and finally 802.11 for WLAN (Wireless LAN). Of course in general IEEE standards, 802.3, 802.4 and 802.5 is commonly known as 802.x; and these standards are for wired LANs. As of today two basic transmission technologies those are in use to set up WLAN (Wireless Local Area Network) are: Infrared light at THz wavelength and Radio wave at GHz (2.4 GHz in the license-free ISM-Industrial, Scientific and Medical band). Infrared technology uses either diffuse light reflected at obstacle like furniture, walls etc or directed light if line of sight path exists between the sender and the receiver. Simple transmitter may be light emitting diodes or laser diodes; and the receiver can be a photodiode. But most of the wireless systems use radio waves. IEEE 802.11 LAN can use both Infrared and Radio wave, but HIPER LAN1 uses only Radio wave. A comparison of Infrared and Radio wave transmission technology is given in the table (7). Table 7: Comparison of Infrared and Radio waves Transmitter Receiver r

Data rate

Shielding

Infrared Technology

Very Simple

Very low as due to low bandwidth of Infrared. 115Kbps to 4 Mbps is the data rate

Infrared can be easily shielded. It can not penetrate obstacles like walls etc.

Radio Wave

Not simple as in Infrared

Higher than Infrared

Shielding is not so simple.

Like other 802.x standards, the standard 802.11 covers only physical layer and MAC sub layer. IEEE 802.11 supports three different physical layers: one layer on using infrared, and another two layers on using basically 2.4 GHz ISM band available free on world wide. ISM bands are: 902 to 928 MHz, 2,4000 to 2.4835 GHz and 5.7250 to 5.825 GHz. Radio LANs operate in the high UHF and low microwave range. Infrared LANs do transmission just below visible light. At physical level three different wireless specifications are: Infrared LANs, Frequency Hopping Spread Spectrum (FHSS) LANs and Direct Sequence Spread Spectrum (DSSS) LANs. FHSS and DSSS LANs belong to radio LANs. FHSS LANs are specified to support data rate of 1 Mbps with a faster specification of 2 Mbps. DSSS LANs are specified for 1 Mbps and 2 Mbps also. FHSS is a spread spectrum technique that allows for the coexistence of a multiple number of networks in the same area by allowing different networks the different hopping sequence. Under IEEE 802.11 standard, 79 hopping channels for North America and Europe; and 23 hopping channels for Japan are specified each with a bandwidth of 1 MHz in 2.4 ISM band. A particular channel is identified by a

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pseudo random hopping pattern. The maximum transmitter power is 1 watt EIRP (Equivalent Isotropic Radiated Power) in US and 100 mW EIRP in Europe. In DSSS, the separation is done by codes rather than frequency. Except this, all other like bit rate and transmission power remain same as in FHSS. The frame formats of physical layer of 802.11 are shown in Fig. (5). The figures in the bracket in the fields refer to the size of the fields in bits. In FHSS frame, the synchronization field is a bit pattern of 010101. The star Frame delimiter( SFD) is 0000110010111101. PLW refers to PDU Length Word i.e. length of payload including 32 bit error control CRC bits at the end payload. It ranges from 0 to 4,095. PSF is for signaling. Out of its 4 bits only one bit is specified to indicate either 1 or 2 Mbps. HEC is a 16 bit header error check field for which ITU-T CRC-16 standard is used. In DSSS frame, 128 synchronization field is made of only scrambled 1 bits. 16 bits start frame delimiter is 1111001110100000. Signal refers to bit rate. Service field is reserved for use. Length is used to indicate payload size with CRC field. HEC is used to check error on header with IUT-T CRC-16 standard. The MAC data frame of IEEE 802.11 is as shown Fig. (6). The figures in the bracket in each field refer to the size of the field in bytes. Frame control is used for several reasons like protocol version and the type of the frame etc. Duration ID indicates the virtual reservation mechanism. Address 1 to 4 which has 46 bits in each is used as they are done for 802.x LANs. Sequence control is used for acknowledgement and error and flow control. CRC is used as it is done 802.x LANs. Synchronization

SFD

PLW

PSF

HEC

Payload

-on (80)

(16)

(12)

(4)

(4)

(Variable)

(a): For FHSS Synchroni zation (128)

SFD (16)

Signal (8)

Service (8)

Length (16)

HEC (16)

Payload (variable)

(b) For DSSS Fig. 5: Physical frame format of IEEE 802.11 radio WLAN Frame

Control (2)

Duration ID (2)

Address 1 (6)

Address 2 (6)

Sequence 3 (6)

Address Control (2)

Data 4 (6)

CRC (0-2312) (4)

Fig. 6: Data format of IEEE802.11

The world is rapidly shifting towards wireless and faster network. In such a rapidly changing scenario, let us see how one of the oldest local area networks, namely Ethernet is keeping pace with the changes. Ethernet dominates as a LAN (Local Area Network), as it is time tested highly reliable, scalable, elegant and low cost network. IEEE 802.3 Ethernet is the established corporate LAN technology, and most of its implementations are with IEEE 802.3u or 100 Base T that defines a 100 Mbps data rate using four pairs of twisted wire pair wiring or Ethernet cable. Tree of the Ethernet is shown in Fig. (7). The Ethernet was originally wired network. It follows the IEEE standard 802.3 for logical link control by which the several nodes can share the single physical medium. The physical layer implementation is made with wires.

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Ethernet Wireless

Wired

802.3

802.11

Conventional Ethernet 10 Base5 Thick Co axial 10 Base2 Thin Co axial 10 Base T UTP

802.11b (11Mbps)

Fast Ethernet

802.11a (125 Kbps–54 Mbps

802.11g (54 Mbps)

100 Base T4 (CAT 3 UTP) 100 base Tx(CAT 5 UTP)

Gigabit Ethernet 1000 Base LX 1000 Base SX 1000 Base CX 1000 Base T (CAT 5+) 10 Gigabit Ethernet under IEEE 802.3ae

Fig. 7: Ethernet as grows.

The IEEE standards for different LANs are 802.3 for CSMA/CD Bus LAN, 802.4 for Token Passing Bus LAN, 802.5 for Token Passing Ring LAN and finally 802.11 for WLAN (Wireless LAN). Of course in general IEEE standards, 802.3, 802.4 and 802.5 is commonly known as 802.x; and these standards are for wired LANs. IEEE 802.11 mainly provides connectivity to corporate LAN. It is very costly for home LAN.

3.12 IEEE 802.11 Architectures In IEEE 802.11 LAN standard, there are two different configurations of a network: ad-hoc and infrastructure. In the ad-hoc network, there is no fixed structure to the network and no fixed access point. There are no fixed points and the computers are brought together to form a network “on the fly” as shown in Fig. (8) and usually every node is able to communicate with every other node. A good example of this configuration is the unscheduled meeting where officials bring laptop computers together to communicate and share information to arrive at a decision. In this type of configuration it is difficult to fix the type of the nodes, but algorithms such as the spokesman election algorithm (SEA) may be used to “elect” one machine as the master of the network with the others as slaves. To know who’s who in the ad hoc networks, a broadcast and flooding method may be used. The infrastructure network (Fig. 9) uses fixed network access points with which mobile nodes can communicate. These network access points may also be connected to landlines to widen the LAN’s capability by bridging wireless nodes to other wired nodes. As and when service areas overlap, handoffs can occur. The structure is very similar to the current cellular networks around the world.

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Computer

Computer

Computer

Computer

Fig. 8: Ad hoc Network.

Computer

Computer

Computer

Computer

Computer

Fig. 9: Infrastructure Network.

Wireless computing, Wireless communication and Wireless networks shall be the rule if future. In such a scenario, WLAN will play a major role. In the last few decades two important wireless technologies those emerge as viable and promising are LEO (Lower Earth Orbital satellites) and 3G (Third Generation) cell phones. But both the technologies fail to meet the expected aspirations. Here at, the WLAN has come out as an alternative. Presently under IEEE 802.11, two major WLAN standards are operating: 802.11a and 802.11b (Table 8). The first 802.11 standard is 802.11b that was approved by the IEEE in 1999. The 802.11b is the first standard that broke the wired brethren of 802.3 wired Ethernets. The 802.11b standard transports data at 11 Mbps using CCK (Complementary Code Keying) using 2.4 GHz band. The 802.11b has is a very successful track record as it is learnt that “ the sale of IEEE 802.11b wireless LANs has increased dramatically from 5000 to 70,000 units per month since early 2000.” It is also reported that: “ The growing popularity and ubiquity of WLANs will likely cause wireless carriers to lose nearly a third of 3G revenue as more corporate users begin using WLANs to connect to the Internet and office networks” Many analysts feel that “ the ease of installing and using WLANs is making it alternative to mobile 3G. In contrast to the reported $650 billion spent worldwide by carriers to get ready for 3G, setting up a WLAN hotspot requires only an inexpensive base station, a broadband connection and one of many interface cards using the 802.11b.” But the speed of 802.11b is one-tenth of wired Ethernets. Therefore the IEEE to have high speed wireless access approved the 802.11a standard concurrently. The IEEE 802.11a standard provides scalable data rates from 125 Kbps to 54 Mbps in increments of 125 Kbps with OFDM (Orthogonal Frequency Division Multiplexing) using 5 GHz band. Actually 54 Mbps is known as “turbo” rates. IEEE 802.11b standard defines only two lower levels of OSI (Open Systems Interconnection) reference model, the physical layer and the Data

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Link Layer Medium Access Control (MAC) sublayer. IEEE 802.11b uses two pieces of equipment, a wireless station, which is usually a PC or a Laptop with a wireless network interface card (NIC), and an Access Point (AP),which acts as a bridge between the wireless stations and Distribution System (DS) or wired networks. There are two operation modes in IEEE 802.11b, Infrastructure Mode and Ac Hoc Mode as discussed earlier in the IEEE 802.11 standard. The physical layer covers the physical interface between devices and is concerned with transmitting physical raw bits over the communication channel. IEEE 802.11b supports different data rates (Table 9). The problems of 802.11a are many: It does not support different devices with different speed, design and complexities. The standards 802.11a and 802.11b are not interoperable. 802.11a is presently used only in North America, and 802.11b is used in the whole of Europe and Asia. The IEEE 802.11e is tasked with a new protocol to non-guaranteed quality service in ad hoc connectivity. The IEEE task group “G” for 802.11 has now deliberating on the next generation standard for 802.11 that would transmit data at the speed of wired Ethernet. The new standard will be 802.11g. The mission of 802.11g standard is to have wireless access at the turbo speed of 54 Mbps while maintaining the interoperability. The IEEE task group “G” for 802.11 has now deliberating on the next generation standard for 802.11 that would transmit data at the speed of wired Ethernet. The new standard will be 802.11g. The mission of 802.11g standard is to have wireless access at the turbo speed of 54 Mbps while maintaining the interoperability.

3.13 GIGABIT ETHERNET Over the decades the speed of Ethernet has grown every time by a factor of 10 starting from 10 Mbps to 100 Mbps to 1000 Mbps (1 Gbps). The Ethernet that carries data at the rate of 1 Gbps or more is known as Gigabit Ethernet. The physical media initially recommended for Gigabit Ethernet is the fiber (table 10) But another IEEE committee is considering the use of UTP cable for Gigabit Ethernet called 1000 Base-T. Keeping the growth of speed, the next Ethernet will be 10 Gbps. The IEEE 802.3ae standardization is going on for 10 Gbps Ethernet. The goal is to achieve very high speed transport keeping maximum compatibility with already installed base of Ethernet 802.3. The 10 Gbps Ethernet will provide almost zero latency service to users. Thus even when coverage area is increased, the remote application and services will appear as local. Table 8: IEEE Standards for LAN IEEE Standards

Definition

802.0 802.1 802.2 802.3

Sponsor Executive Committee Higher Layer LAN Protocol Logical Link Control Medium Access Control (MAC) of CSMA/CD Bus LAN (Example: Ethernet) 10 Gbps Ethernet Data Terminal Equipment -electrical via balanced cabling for 802.3 interface MAC of Token Bus LAN

802.3ae 802.3af 802.4

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802.5 802.5t 802.5v 802.5z 802.6 802.7 802.8 802.9 802.10 802.11 802.11a 802.11b 802.11g 802.12 802.13 802.14 802.15 802.16

45

MAC of Token Ring LAN 100 Mbps Token Ring LAN Gigabit Token Ring LAN Link Aggregation MAN Working Group Broadband Technical Advisory Group Fiber Optic Technical Advisory Group Isochronous or Integrated Services LAN (ISLAN) Inter operable LAN Security Working Group Wireless LAN (WLAN) Working Group 56 Mbps WLAN 11 Mbps WLAN Next Generation WLAN Demand Priority Working Group Inactive Cable Modem or Cable TV Wireless Personal Area Network (WPAN) Broad Band Wireless Access (BBWA)

Table 9. Different data rates of IEEE 802.11b IEEE 802.11b Data Rate Specifications Data Rate in Mbps

Code Length

Modulation

Symbol Rate in MSps

Bits/Symbol

1

11 (Barker Sequence)

BPSK

1

1

2

11 (Barker Sequence)

QPSK

1

2

5.5

8 (CCK)

QPSK

1.375

4

11

8 (CCK)

QPSK

1.375

8

Table 10. Gigabit Ethernet Gigabit Ethernet

Mode Supported

1000 Base LX (Long wave laser over single mode and

Fiber diameter in micron

Maximum Distance in segment

Single

9

10 Km

Multi

50

550 m

Single

50

3 Km

Multi

62.5

440 m

1000 Base SX (Short Wave laser over multimode fiber)

Multi

50

550 m

Multi

62.5

260 m

1000 Base CX (Balanced shielded 150 ohm copper cable)

BALANCED SHIELDED CABLE

25 m

1000 Base T (UTP Cable)

UTP Cable

100 m

multimode fiber)

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3.14 Trade Off In the speed jargon, the wired Ethernet still outplays the wireless Ethernets. But Wireless LANs are also picking up speed. Trade off lies in wired LANs’ speed versus wireless LANs’ flexibility, reliability and low maintenance cost or looking at today IEEE 802.3ae versus 802.11g. Alan Mc Adams, Chair of IEEE-USA committee on communication and information policy (CCIP) said “ Gigabit Ethernet over fiber will allow the transfer of Ethernet technology, concepts and benefits from Local Area Networks to Metropolitan Area Networks and Regional Area Networks.” It is reported in the IEEE The Institute of Oct’2002 that “ The June approval of the IEEE 802.3ae standard for 10 gigabit per second Ethernet has the potential to allow Gigabit Ethernet over fiber (GEF) technology to supplant current telecommunications infrastructures with its cost, speed and distance advantages.”

3.15 Integrating Wireless Protocols In nature, only one thing is permanent and that is nature. This law of nature appears to be equally applicable to Ethernet, which is ever changing and growing.. The present scenario is that of cellular data and of wireless LAN data, respectively under protocols of TIA/EIA (Telecommunications Industry Association/Electronics Industry Alliances) IS-856 and IEEE 802.11.Whereas IEEE 802.11 meets short-range high speed data network, IS-856 is meant for wireless voice data. They are complement to each other. They may take advantage of each other and integrate to provide typical bridge (Fig. 6) to satisfy demand for access to the wireless Internet. On the other hand, the IEEE 802.11 will immensely take part in the seamless integration of total wireless access and networking in the next-G era (Figs. 10 and 11).

Computer

IEEE 802.11 access point

Wireless Station

Computer

Independent device

802.11

Fig. 10: Typical integration for a corporate connection.

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Seamless Next G

47

Mobile wireless Integrat

3G/Data WLAN 801.11a

2G/Digital WLAN 802.11b

1G/Cellular WLAN/ Conventional

Fig. 11: Seamless integration of wireless access and networking—IEEE802.11.

3.16 IEEE 802.15.4 Standard—Low Data Rate, Low Cost Wireless Home Networking Solution Due to applications of networking in almost everywhere, several attempts are being made to offer solutions that aim to be flexible, cost effective, reliable and consume less power, the features particularly so important for home or residential networking. In the wired communication, the DSL (digital subscriber loop) technology (discussed later) is one important driver. The cost effectiveness is achieved by utilizing the existing copper line in the local loop. But the wireless communication and networking has an edge over wired technologies for which a wireless local loop solution is needed. The wireless networking and communication technologies that have appeal in voice and data applications in residential or home services are among others cellular, cordless and IEEE 802.11b. The consideration of cost effectiveness and low power consumption has motivated for development of a new standard IEEE 802.15.4 for home networking with low data rate wireless solution. The initiative to develop a standard for low powered and low cost home wireless networking was taken by IEEE working group 15 in 2000. Besides home automation, the standard is poised to be applied in different services of industry like industrial control, automotive sensing (monitoring tire pressure, sensing of soil moisture, pesticide, pH levels), and disaster management (sensing and determining location of disaster) etc. In home applications, the services of the standard will be PC peripherals (Keyboard, PDA, Mouse), consumer electronics (TV, Radios, VCR, CD etc), automation (heating, air conditioning, ventilation, windows and doors lock), remote control, health monitoring, security and PC enabled

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services. These applications need data range ranging from a few kilobits per second (kbps) to 115.2 kbps. The acceptable delay or latency for these services ranges from 15 ms to 100 ms. The major features of the IEEE 802.15.4 proposed standard are: • like all other IEEE standards, the IEEE 802.15.4 refers to lower layer specification. In reference to OSI ISO 7 layer protocol, the IEEE 802.15.4 refers to DLL (Data Link Layer). DLL is split into two sub layers: LLC (Logical Link Control ) and the MAC (Medium Access Control) sub layer. The LLC is as per other specification of 802.3 etc. The IEEE 802.15.4 defines a separate MAC sub layer. • The IEEE 802.15.4 as recommended two versions of physical layers: (1) 868/915 MHz and (2) 2400 MHz • the IEEE 802.15.4 supports both star and peer to peer networks including ad hoc networks • the generic frame format of IEEE 802.15.4 is made of : frame control and sequence number that are respectively 2 and 1 bytes. The Address field is variable from 0 to 20 bytes. Payload is variable, but the full MAC frame is limited to 127 bytes. Frame check sequence is 2 bytes and it uses 16-bit CRC control. In the physical layer frame, the total header length s 6 bytes with preamble of 4 bytes, start of packet delimiter of 1 byte and physical header of 1 byte. The payload is limited to 127 bytes, being the MAC frame. • The header fields of the physical layer frame format are: 4 bytes preamble that is used for synchronization, 1 byte start of packet delimiter that is used to indicate the end of preamble, and 1 byte physical header used to specify the length of physical service data unit • The physical layers in IEEE802.15.4 uses DSSS (direct sequence spread spectrum) methods with different channel frequencies and modulation parameters. • The DSSS method is chosen in order to use low cost IC for implementation by which the cost of the system is made low • IEEE 802.15.4 aims to provide excellent battery life, low transmit power • IEEE 802.15.4 devices aim as much as 99.9 percent of sleeping time • The simplicity is the another attraction of IEEE 802.15.4

3.17 IEEE 1394 for Home Network A recognized definition of home network is “A home network interconnects electronic products and systems, enabling remote access to and control of those products and systems, and any available content such as music, video or data.” Several standards are in perspectives for application in home networks. For example, IEEE 802.11 is the most talked of standards for wireless interface. The standard has got several modification in order to meet with high speed requirements as well as other different requirements of home networking. Unfortunately the standard is still costly for home networks. Nonetheless the standard is yet to overcome many obstacles for wide spread deployment. Several modifications are proposed in the IEEE 802.11 standard for different requirements. For example, the task group “G” is proposing 802.11g that would transmit data at the speed of wired Ethernet. The IEEE 802.3ae standardization is going on for 10 Gbps Ethernet. The goal is to achieve very high speed transport keeping maximum compatibility with already installed base of Ethernet802.3. The 10 Gbps Ethernet will provide almost zero latency service to users. Thus even when coverage area is increased, the remote application and services will appear as local. But the cost factor is not considered in such modifications.

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The IEEE 1394 working group defined a standard known as IEEE 1394. Truly speaking, the standard was originated by Apple Computer Company for desktop LANs. IEEE1394 is a low cost digital interface that can work over existing copper, fiber and co axial cables too. The Broadband Home Company has used co axial cable to extend IEEE1394 interface beyond the local audio & video cluster. The solution so provided looks like a virtual IEEE1394 wire connection to other IEEE1394 networks. It supports hot plugging, thereby allowing users to add and/or remove devices when the interface bus is active. It provides both hardware and software specification for peer to peer connection at different operating speed of 100, 200 or 400 Mbps. The enhancement in speed may go to support 800, 1600 and 3200 Mbps. It supports a scalable architecture to meet with different speeds of different requirements, thereby providing a cost effective solution. That standard integrates communication, entertainment, and computing to provide a single digital interface for consumer multimedia. It supports both asynchronous and synchronous types of data transfer as required in home networks. Asynchronous transfer is related to conventional data/computer file transfer. But for multimedia application of voice and video where delay is the most sensitive issue, transport at the guaranteed delay is done by synchronous or isochronous technique that is duly supported by IEEE 1394. It supports high speed communication at low cost interface. IEEE 1394 has been recognized as digital interface by many organization for different purposes that includes entertainment, consumer applications, digital TV, Home multimedia, conventional file transfer, digital video conference etc. A typical integration of several networks including IEEE1394 in a single cable is as shown in Fig. (12).

Phones

0

2.5 MHz

TV Channels

Cable

Infrared

5 MHz

55 MHz

1000 MHz

IEEE 13 94

Ethernet

1500 MHz

Fig. 12: Typical several networks in a cable.

3.18 Paging Paging is a one way message system unlike two way interactive mode system of communication of cellular. Pagers transfers message on wireless network; and thereby support mobility. Person having a pager can be contacted anywhere at any time. Pagers are quite useful for doctors, journalists etc. Paging is basically a back up system to telephones. It enhances the productivity of telephones. They work on a simple technique. The caller may dial the paging center through usual telephone and leave the message with operator along with callee’s pager number. The operator shall send the message to the callee’s pager. The message will then be flashed out on the callee’s pager with an activating signal. There are two basic types of paging transmission standards : POCSAG (Post Office Code Standard Advisory Group) and RDS (Radio Data System). In India frequency allocation for POCSAG is 134-168 MHz whereas idle band of AIR’s (All India Radio) existing FM network is used to RDS. AIR is operating RDS paging. POCSAG is under DoT (Department of Telecommunication). Different types of pagers are available in the market like numeric, alpha-numeric and recently introduced English type. In advanced countries, two way paging is being developed.

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3.19 VSAT Satellite communication started with the pioneer work of Dr. Artheer C. Clarke. He showed that using just three satellites placed each at 1200 apart from each other and at a height of about 36000 Kms from earth surface, world wide communication is possible. The satellites placed in orbits of about 36000 kms away from earth surface are known as geostationary satellites as they rotate in their orbits once in 24 hrs’. therefore, from any point on the earth, these satellites appear stationary to any person on earth. Due to two big advantages of satellite communication over other means of communication, satellite communication has a big appeal to users. It is said that “Going for Satellite means Going for Wireless Communication.” Wireless communication is more reliable, flexible and adaptable than wireless communication. We do communication by acoustics in wireless communication. Auther saying is “Going for Satellite means going for wide area coverage.” Wide area coverage has a natural attraction. Over the years, hence, the satellite communication has diversified its area of application and technologies. One of the major technologies is VSAT. VSAT (Very Small Aperture Terminal) is a cost-effective technology meant for networking computers and terminates mainly for the purpose of data communication. The VSAT network may be a wide network and extended in any remote location easily. The basic components of VSAT networks are: 1. a geological satellite. 2. a master earth station or hub. 3. micro earth stations(VSAT stations or nodes) VSAT node is made of VSAT ports, VSAT controller and VSAT antenna. The size of the antenna is 1.2 m × 1.8 m. Basically two types of topology are used in VSAT communication. Star and Mesh. Star topology uses TDM-TDMA (Time Division Multiplexing–Time Division Multiplexing Access) CDMA(Code Division Multiplexing Access) and mesh topology uses DAMA–SCPC (Demand Assigned Multiple Access—Single Channel Per Carrier). VSAT communication is broadcast communication. VSAT nodes cannot communicate directly to each other. They communicate with each other via master earth station or hub. Naturally VSAT communication is known as two-hop-communication. This means a VSAT signal from node has to travel at least 36000 × 2 x 2 = 144000 kms to reach another node. The delay for which shall be around 480 msec at least. The quality voice and video communication do not allow more than 80 msec delay between transmitter and receiver. Delay is not a issue for data transportation. VSAT communication is, hence, most suitable for data communication. The characteristics of VSAT are: 1. cost effectiveness is a big advantage of VSAT communication. An STD call between Delhi and Mumbai can be around Rs. 40 whereas a VSAT call may be about Rs 10. 2. Reliability and flexibility are always present in VSAT communication as it is wireless communication. A leased telephone line can have at most 90% up time. A VSAT line shall have around 99.5% up time. Due to wireless, ease of expansion of VSAT is there. This is how flexibile is VSAT network. VSAT(Very Small Aperture Terminal)communication is useful for huge organization like DVC,ONGC, IOC, BHEL, etc., as a means of cost effective date communication system for within the organization VSAT is a small dish antenna 60 cm or 120 cm which communicates

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with central hubs and terminals via satellites VSAT is cheaper then conventional earth station communication using satellites. Power budget calculation [26] shows that in order to meet with required bit energy to noise ratio a large antenna is essential for covering a wide area. Cost increases with antenna size. For intra organizational communication a small antenna is justified. DoT has allocated extended C band for VSAT communication in India. Nowadays VAST can also provide cost effective telephony and fax services. VAST is a low speed 1200 BPS data communication system and employs TDMA for accessing.

3.20 Mobile Satellite Service MSS (Mobile satellite Service) is a form of cellular like wireless communication. With terrestrial wireless networks, wireless communication is either not economically viable (examples are :remote areas, semi-hill areas etc.) or may not be physically possible (overseas, over large mountains). In such cases, wireless communication via satellites is an alternative proposition. Satellites can be of three types: GEOs (Geo-stationary Satellites), MEOs (Medium Earth Orbital Satellites) and LEOs (Lower Earth Orbital Satellites). Orbital altitude of GEOs, is 35786 km; whereas the same for MEOs and LEOs are respectively of the order of 10,000 km and 1000 km. INMARST, INTELSAT5, INSATAs are examples of GEOS. One example of LEOs is Odyssey with 12 satellites at an altitude of 10600 km. Odyssey is proposed to use CDMA technology and ground based switching for voice communication. Project 21 is another example of MEOs. In project 21, 10 satellites are used and they are placed at an altitude of 10500km. Project 21 is proposed to use TDMA technology to support both voice and data. Examples of LEOs are Iridium with 66 satellites oat an altitude of 765 km., Globstar with 48 satellites at an altitude of 1389 km. and Ellipso with 24 satellites at an altitude of 429-2903 km. They are respectively proposed to use FDMA/TDMA, CDMA and FDMA/CDMA technology. Iridium and Globstar are to support voice and paging. A good account of GEOS, MEOS and LEOs can be seen in[29,34]. GEOs have two major advantages : they are costly systems (due to high transmitter power and large antenna size) and round trip propagation delay is about 270 ms. Large round trip propagation delay is unwarranted in voice and in real time interactive communication. LEOs and MEOs can overcome the problem. In these systems many satellites are required to be placed in the orbits. As the satellites are not geo-stationary, for continuous communication, hand off operation among satellites are required. MSS can therefore support mobility. In MSS, MEOs, LEOs are base stations and they are on motion. Here lies the difference between cellular communication and MSS. In MSS, actually base stations are assumed as mobiles. Another disadvantage of LEOs and MEOs is their short life span. HEOs (Highly Elliptical Orbital Satellite) may also be used in MSS for wireless communication. A good account of MSS in personal communication is found out in[30,31]. Satellites Communication (GEOs/MEOs/LEOs) In general “going for satellites” means going for two important philosophies - going wireless and going for large area, even upto whole world coverage. These greatly influenced the use of satellites in communication. Till date, the major satellites involved in communication are GEOs (Geo Stationary Satellite). GEOs are placed at orbit of high 35800 Kms away from earth surface, and therefore they move around the earth once in 24 hrs. Thus GEOs look stationary at anywhere relative to earth, and communication using fixed antenna is possible. A world wide coverage with just three GEOs is possible, if they are placed at equidistant apart of 1200. But there are number of disadvantages with GEO based communication: (1) GEO has poor elevation at higher latitudes and no coverage of the polar regions, (2) mobile communication using GEO under INMARST (International Maritime Satellite Organization) is possible to

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provide voice, data, telex and facsimile services to ships; but for this, there remains the requirements of very high power for both terminal and the space craft, (3) the large distance between earth and GEO causes a high propagation loss of about 200 db and a time delay of about 350 msec in one way. Such long delay is not acceptable to 80% of users. MEOs are placed at orbit height of about 10,000 Km or above; whereas LEOs are placed at orbit height of 2000Km or less. By this LEO overcome the disadvantages of GEO based communication pointed out above. Besides, there some major differences between LEO based and GEO based communications. # The communication in LEO is done through a constantly moving and tracking switching network and antenna rather than fixed system of GEO. Mobile communication in LEO is based on the relative mobility. LEO systems move and moving users appear stationary. For example in Iridium system, the LEO speed relative to earth is 26,676 Km/ hr, whereas average mobile speed is around 90 Km/hr. # The GEO based communication is single hop (earth–satellite–earth) communication; while LEO based communication is multi hop communication. # Under LEO, the communication across the world is low cost. For example while a typical GEO can provide about 10,000 channels for global services, the LEO can provide 7000 channels for regional services and 35,000 to 70000 channels for global services. This typical means the cost per channel for global services in LEO is about one half of that of the for global services in GEO. LEO is effective for global services rather than regional services. # LEOs are smaller than GEOs. The mass of LEO satellite range from 50 to 700 Kg (whereas that of GEOs range from 1800—2000 Kg). Therefore economic multiple launching of LEOs is possible. The variety of services offered by the satellites was divided into three groups by ITU (International Telecommunication Union). These are: (1) Fixed Satellite Service (FSS), which offers radio communication services between fixed location in earth through one or more satellites. (2) Broadcast Satellite Services (BSS) which provide direct reception of satellite broadcast by public and/or community and (3) Mobile satellite Service (MSS) which provide a communication between mobiles through one or more satellites. As in past twenty five to thirty years, FSS and BSS shall continue to be served with GEO mainly. LEOs shall dominate MSS. It is said, “The LEO and MEO systems offers an innovative approach to providing service to a country, a region, or to the whole world. Instead of transmission to and from a fixed point in sky (as for geostationary satellite systems) the user transmits to and receives from a network of lower altitude satellites, that move overhead with some satellites disappearing from view as others come over the horizon. The system can provide service to all parts of the world as the low altitudes satellites pass over different parts of the earth.” LEO Systems LEOs are classified into two groups: “Little-LEOs” and “Big-LEOs”. The little-LEOs group consists of satellites, which are small in size and low in weight. Little LEOs are expected to provide services of only low bit rates of the order of 1 Kbps (kilo bits per second) and they are placed near orbit height of around 1000 km. Naturally they are used for non-voice services. The frequency band allocated for mobile satellite services (MSS) under little—LEO group are: 148-150.50 MHz (uplink) and 137-138 MHz (down link). Big-LEO group of satellites are expected to provide near-toll-quality voice service and other related services like paging, data communication, facsimile, and position location. BigLEO group contains MEO (International Circular Orbit) satellites. The important three BigLEO systems are: Globalstar, Odyssey and Iridium.

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Iridium Iridium system was proposed by Motorola to provide global services of voice, data, fax, paging, RDSS; and was scheduled to operate in 1998. The cost of the system is about US$ 3.4 billion. The system is composed of 66 (77) satellites with 11 satellites in each of 6(7) polar orbits placed at the orbit height of 780 Km above earth surface. Satellite shall provide 3168 cells out of which only 2150 cells shall remain simultaneously active to provide global coverage of mobile/ cellular telephone service. In the system the same frequency band 1616–1626.5 MHz shall be used for both uplink and downlink communication on time-shared basis. Message on one telephone to another is transmitted from mobile to satellite using 23 GHz (22.55–23.55) intersatellite link until the satellite viewing the destination mobile is reached. The system uses FDMA (Frequency Division Multiplexing Access) and TDMA (Time Division Multiplexing Access) on uplink and downlink respectively. The connection to the terrestrial network is done via earth station gateway. Voice circuits per satellite are 1100. Voice service rate is 2.4 Kbps. Data service rate is 7.2 Kbps. Modulation technique used in the system is OPSK ( Quadrature Phase Shift Keying). Footprint diameter of each satellite is 4700 Km. and therefore satellite visibility is 11.1 minutes. Satellite life span is rather less and it is 5 years. Satellite antenna type is fixed and six feet in size. Beams per satellite is 48 and therefore total beams in the system are 3168. Feeder uplink and downlink frequency 27.5-30 GHz and 18.8-20.2 GHz. Minimum and maximum one way propagation delay are respectively 2.6 msec and 8.22 msec. Airtime charge per minute is US$ 3.0. Iridium System is working but not to the level of satisfaction expected before launch. Globalstar Qualcomm proposed Globalstar LEO system to provide services of voice, data, facsimile and RDSS. The Globalstar system shall use 48 satellite in 8 polar orbits. The orbit height is 1400 Km above earth. It provides global coverage and can work with existing PSTN (Public Switched Telephone Network). Calls are granted through satellites only when access is available to the terrestrial all network. The PSTN can be used via gateways for long distance communication. The system does not support intersatellite link. The gateways to the PSTN shall use 6.5 GHz and 5.2 GHz respectively for uplink and downlink communication. Access technology for MSS is CDMD (Code Division Multiplexing Access) via L-band (1610.0–1626.6 MHz) and S-band (2483.5–2500.0 MHz) for uplink and downlink communication. The modulation technique used in the system is QPSK. The system can support 2000 - 3000 voice circuits / satellite. The voice and the data service rate in the system are 2.4 to 9.6 KBPS respectively. Minimum and maximum one way propagation delays are respectively 4.63 msec and 11.5 msec. Mobile terminal cost is about US$ 750. Air time charge per minute is 30 cents. The satellite foot print diameter is 5850 Km. The satellite visibility is 16.4 minutes and lifespan is 7.5 years. The satellite on a orbit mass is 450 Kg. The system cost is US$ 1.8 billion. The satellite antenna is fixed and size is 03 feet. Feeder uplink and downlink frequencies are respectively 5.091 - 5.250 GHz and 6.875 - 7.875 GHz. The satellite output power is 1000 watt. Beams per satellite is 16 and total beams are 768. Comparing Iridium with Globalstar, a report says globalstar has capital costs (at $1 billion) one-half Iridium, circuit costs one-third Iridium’s and terminal cost (at $750 each) onefourth Iridium’s. With no intelligence in space, Globalstar relies entirely on the advance of intelligent phones and portable computer devices on the ground; it is the Ethernet of satellite architectures. Costing one-half as much as Iridium, it will handle nearly 20 times more calls.

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The advantages of Globalstar stem only partly from its avoidance of complex intersatellite connection and use of infrastructure already in place on the ground. More IMPORTANT is its avoidance of exclusive spectrum assignments. Originating several years before spread-spectrum technology was thoroughly tested for cellular phones, Iridium employs time division multiple access, an obsolescent system that requires exclusive command of spectrum but offers far less capacity than code division multiple access. “It is said that” Iridium’s voice service cannot complete with GlobalStar’s cheaper and more robust CSMA system. “It is also reported that “Iridium satellite together use 80% more power than Globalstar’s, yet employ antennas nearly twice as larger and offer 18.2 times less capacity per unit area”. Odyssey TRW proposed a system known as odyssey to provide voice, data, facsimile and RDSS services on global basis. In the system, 12 satellites and 3 polar orbits are used. The orbit height is 10370 Km above earth surface, and therefore this system is better known as MEO system. The orbital period of satellites is 359.5 minutes and visibility is 94.5 minutes. The satellite mass is 2207 Kg. Footprint diameter of each satellite is 10540Km. The access technology of the system is CDMA, and modulation technique is QPSK. The system operates at L and S brand. The mobile uplink and downlink frequencies are respectively 1601.0–1626.5 MHz (L brand) and 2483.5–2500.0 MHz ( S brand). The system supports to 3000 to 9500 voice circuits per satellite. Voice and data services are made with respectively 4.8 KBPS and 2.4 KBPS. Delay are respectively 34.6 msec and 44.3 msec. Airtime charge in the system is US$ 0.65 per minute. Satellite antenna type is steer able. Uplink and downlink feeder frequencies are 29.1– 29.4 GHz (ka - band) and 19.3–19.6 GHz (ka band) respectively. The system supports 61 beams per satellite, thereby supporting total 732 beams. Satellite output power 6177 watt. Ellipso Ellipsat proposed a LEO satellites system known as “ELLIPSP” to provide voice, data, facsimile, and RDSS, using 15 (9) satellites placed in 3(1) polar orbits. The orbit height is 7800 Km over earth surface and provides coverage over entire northern hemisphere and to southern hemisphere upto 50 south latitude. It uses L and C-band for communication. The mass of satellite on orbit is 300Kg. The system supports voice and data at 4.2 KBPS and 0.3 to 9.6 KBPS respectively. The satellite life span is 5 years. Air call charge per minute is US$ 0.50. Access technology is CDMA. ICO Hughes proposed ICO system to provide services of voice, data, fax, paging, massaging and position location employing 10 satellites in 2 orbits placed at 10355 Km. The system is MEO rather than LEO. The system supports voice and data at the rates of 4.8 KBPS and 2.4KBPS respectively. The satellite life time is 10 yrs. Air charge is US$ 1 to US$ 2. The system covers service all over world. Orbit period is 358.9 minutes and satellites visibility is 115.6 minutes. The down-link and the up-link frequencies for MSS are 1980.0–2010.0 MHz and 2170.0– 2200.0 MHz respectively. Satellite antenna type is fixed. Feeder uplink and downlink frequencies are respectively 5.091–5.250 GHz (C-band). The system supports voice and data service at the rates of 4.8KBPS and 2.4 to 9.6 KBPS respectively. Minimum and maximum one way propagation delays are respectively 34.6 msec and 48 msec. Air time charge per minute is 2.00.

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Teledesic Teledesic system of LEOs is a different class. Difference stems from the application point of view. The system is aimed at providing wireless broad band access and computer networking. Little LEOs are equivalent of paging. Big-LEOs like iridium, globalstar and ICO, are equivalent of fiber. The system comprises 840 small satellites in proposed 21 orbital planes and 20000 super cells on the earth in order to provide broadband-on-demand service by 2002 for 99% of the earth. The orbit height is 700Kms. Teledesic system is expected to use ha—band of frequencies, between 17 GHz and 30 GHz. And antennas of size 66 cms. The Teledesic system is Giga Band system. A comparative study says” in the long run Iridium could be trumped by Teledesic. Although Teledesic has no such plans, the incremental cost of cost of incorporating an “L” band transceiver in Teledesic, to perform the Iridium functions for voice would be just 10% of Teledesic’s total outlays, or less than $ 1 billion (compared with the $ 3.4 billion initial capital costs of Iridium). But 840 linked satellites could offer far more cost-effective service than Iridium’s 66. Iridium’s dilemma is that the complexities and costs of its ingenious mesh of intersatellite links and switches can be justified only by offering broadband computer services. Yet Iridium is doggedly narrow band system focused on voice. The evolutionary process of development of personal communication shall go on using existing cellular, cordless, satellites, wireless data networks, WLL ( Wireless Local Loop) VAST (very small Aperture Terminal), wireless centrex / PBX, and other GMS (Third General Mobile System/Cellular) and MSS (Mobile Satellite Service) etc. But the use of MSS in personal communication may be revolutionary of evolutionary, which remains to be seen. The MSS under different LEO projects-both big LEOs and small LEOs, is believed to be a high hope of implementing personal communication.

3.21 Wireless ATM ATM (Asynchronous Transfer Mode) technology is believed to be the only suitable presently available technology for integrated services-present and future, time sensitive and time insensitive, voice, video and data. Thus ATM can support multimedia and can be service or application independent technology for transport and switching. Therefore, a future direction for development of wireless ATM is initiated. WAND (Wireless ATM Network Demonstrator) is such a project of European unions. HIPERLAN standards of 5 GHZ shall aim towards wireless ATM. In USA, a high speed multimedia network using ATM is under development. It is targeted to operate at 25 Mbps using 25 MHz channel in 5 GHz band (5.15-35 and 5.7255.875 GHz). On the other hand, work has started to provide air interface to internet.

3.22 Changing Scenario of Internet Internet is going to have several changes and version like IPv4 to IPv6, VoIP, Internet2, Wireless Internet and under sea /Cable Internet. But most immediate is from IPv4 to IPv6 and VoIP. 3.22.1 IPv6 Presently Ipv6, Internet works on IPv4 (Internet Protocol Version 4) as defined in RFC791. By the middle of 1990s, by the time of which the IPv4 became about 15 years old, it was recognized that there are several limitations in the IPv4. Table (XI) lists the major studies on the run up of IPv4. Two important limitations are the inadequate address space available with 32-bit address space of IPv4 and inability of the IPv4 to support real time services or time-sensitive services. The 32-bit address space is not sufficient to cope up with the growing Internet users.

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Since it is estimated that the Internet has been growing by a factor of two every year, the underlying principles and assumptions based on which IPv4 was designed are going to be invalid. What was duly sufficient for a few million users or a few thousands of networks will no longer can support a world with tens of billions of nodes and hundreds of millions of networks. Inability of IPv4 to support real time services was the stumbling block to realize Internet telephone. IPng (Internet Protocol Next Generation) initiative (RFC 1752) was then, started by the Internet Engineering Task Force (IETF). By 1996, the IETF proposes IPv6 (Internet Protocol Version 6) under IPng initiatives, which is supposed to solve the problems of IPv4 including the two major limitations mentioned above. IPv6 is therefore the future replacement of IPv4. From the experience over IPv4, it was felt that new version should take care of: More addresses, Reduced overhead, Better routing, Support for address renumbering, Improved header processing, Reasonable security and Support for mobility. Under the IPng initiatives the main techniques investigated were: • TUBA that refers to TCP (Transmission Control Protocol) and UDP (Users’ Datagram Protocol) with bigger addresses • CATNIP that means common architecture for the Internet. The main idea is to define a common packet format that will be compatible to IP, CLNP (Connectionless Network Protocol) and IPX (Internet work Packet Exchange). CLNP has been proposed by OSI (Open System Interconnection) as a new protocol to replace IP, but never been adopted because of its inefficiency • SIPP (Simple Internet Protocol Plus) that proposes to increase of the number of address bits from 32 to 64, and to get rid of unused fields of IPv4 header As none of the above three was seen to be suitable. As such, a mixture of all these three along with other modifications was suggested in RFC 1883. The RFC 1883 suggested the modifications as below: • Expanded Addressing in suggesting 128 bits for address that may allow more levels of address hierarchy, increased address space and simpler auto configurable addressing • Improved IP header format by dropping the least used options • Improved support for Extensions that will bring flexibility in operations • Flow Label that will make the real time services possible over Internet Based on the experience gained in operation of IPv4 over about 20 years, the design of IPv6 has considered four major simplifications: • assigning fixed format to each header. This ensures the removal of header length field that is essential in IPv4 • removing header checksum. The main advantage in removing header checksum is to diminish the cost and the time delay in header processing. This may cause the data to get misrouted. But experience has shown that the risk is minimal as most of data pack is encapsulated by the packet checksum at other layers like MAC (Media Access Control) procedure in IEEE 802.X and in adoption layer of ATM (Asynchronous Transfer Mode) etc. • removing the hop by hop segmentation procedure • removing TOS (Type Of Service) field that IPv4 provides, since experience has shown that this field has ever been set by applications. On the other hand, IPv6 has considered two new fields, flow label and priority. These are included to facilitate the handling of real time services like voice, video and high quality multimedia etc.

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Thus the IPv6 was finally come up with packet format as in the Fig. (13). The final specifications of IPv6 were produced in RFC 1883. The new features of IPv6 are: Version (4 bits)

Priority (4 bits)

Payload Length (16 bits)

Flow label (24 bits) Next Header (8 bits)

Hop Limit (8 bits)

Source Address (128 bits) Destination Address (128 bits) Variable length TCP pack (which is TCP header + Payload)…….. Fig. 13: IPv6 Packet Format

• A fixed and streamlined 40-byte header: IPv6 is having fixed header bytes like that in ATM (Asynchronous Transfer Mode) cell. This makes the node processing delay to minimize, and thereby becomes more suitable for real time services like voice, video and multimedia. • Expanded addressing capabilities: A 128-bit address space in IPv6 instead of 32 bit as in IPv4, is believed to ensure that the world won’t run out of IP addresses. The 128 bit address size gives rise to a total of 256 × 1036 different addresses. It is expected the Internet under IPv6 to support 1015 (quadrillion) hosts and 1012 (trillion) networks. The Internet under IPv4 can support maximum 232 hosts. Therefore the IPv 6 address space is about 64 × 109 times more that that of IPv4. This is why it is expected that future and exponential growing demand for Internet connection be met with IPv6. • New Address Class: Besides unicast and multicast, IPv6 has the provision of anycast addressing. Anycast address allows a packet addressed to an anycast address to be delivered to any one of a group of hosts. • A single address associated with multiple interfaces • Address auto configuration and CIDR (Classless Inter-domain Routing) addressing • Provision of extension header by which special needs like checksum, security options may be introduced. • Flow labeling and priority: Flow level and priority headers are used to comfortably support the real time services. By assigning higher priority to the real time packets, the necessity of time sensitiveness is restored. Data packets and for that purpose time insensitive packets are assigned low priority and serviced by the best effort approach. As per RFC 1752 and RFC 2460, this new feature allows “ labeling of packets belonging to particular flows for which the sender requests special handling, such as a non-default quality of service or real-time service.” Hence video and audio may be treated as flows whereas traditional data, file transfer and e-mail may not be treated as flows. • Support for real time services • Security support that could be eventually seen as the biggest advantage of IPv6. Today, billion dollars business is done over Internet. To keep the business secure, public crypto system has emerged out as one of the important tools. IPv6 with its ancillary security protocol has provided a better communication tool for transacting business over Internet • Enhanced routing capability including support for mobile hosts. IPv6 as such is not simple extension of IPv4, but a definite improvement over IPv4 in order to meet growing demand of Internet connectivity and the services of real time communication via Internet.

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The functions of IPv6 headers that is of base headers of fixed 40 bytes are: • Version field (4 bits). It contains the version number. Versions are 4 and 6. For version 6, this field is 6 (i.e. 0110). The various assigned values for IP version label are shown in table (12). But it must be remembered that just putting a number “6” or “4” does not make the corresponding IP packet. For the corresponding IP packet the proper format is required to be made. • Priority (4 bits). The bits in the field indicate the priority of the datagram. The priority levels are 16 from 0 to 15. The first 8 priority levels (from 0 to 7) are for the services that provide congestion control. If the congestion occurs, the traffic is backed off. These are suitable for non-real time services like data. The different priority levels under the first 8 levels are: 0 that defines no priority, 1 that defines background traffic like Netnews, 2 that defines unattended transfer like e-mail, 3 remains reserved, 4 that defines attended bulk transfer like FTP (File Transfer Protocol), NFS, 5 remains reserved, 6 that defines interactive traffic such as Telnet, X-windows, and 7 that defines control traffic such as SNMP (Simple Network Management Protocol) and routing protocols. The higher 8 priority levels (from 8 to 15) are used for services that will not back off in response to congestion. Real time traffics are examples of such services. The lowest priority level of this group 8 refers to traffic that most willing to be discarded on congestion and the highest priority level15 is for traffic that is least willing to be discarded. · Flow level (24 bits). It is proposed to be used to identify different data flow characteristics, which will be assigned by the source and can be used to label packets. The packet labels may be required to provide special handling of packet by IPv6 routers, such as defined quality of service (QoS) or real time services. The combination of the sender IP address and the flow label creates a unique path identifier that can be used to route the datagrams more efficiently. The field is still being experimented. Flow is actually a sequence of packets coming from a particular source and destined for a particular destination. A flow may require a special handling by routers. Each flow is uniquely defined by the combination of the source address and a non-zero flow label. The flow label can be from (000001)H to (FFFFFF)H in hex. The packets having no flow label are given a zero label. All packets in the same flow must have same flow label, same source and destination addresses and same priority level. The initial flow label is obtained by the source by pseudo random generator, and the subsequent flow numbers are obtained sequentially. • Payload length (16 bits): The field indicates the total size of the payload of the IP data gram that excludes header fields. It can define up to 65,536 bytes of payload. • Next header (8 bits): The field indicates which header follows the IP header. The next header can be either one of the optional extension headers used by IP or the header for an upper layer protocols such as UDP or TCP. The field defines the type of extension header. For example 0 defines IP information, 1 defines ICMP (Internet Control Message Protocol) information, 6 define TCP information, 44 defines fragmentation header, 51 defines authentication header and 80 defines ISO (International Standard Organization) /IP information. Each extension header again contains an Extension Header Field and a Header Length Field (Fig. 14). When there is no other extension header, the next header will be TCP and hence the next header field will contain 6.The length of the base header is fixed 40 bytes. The extension header gives the

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functional flexibility to the IPv6 datagram. Maximum six extension headers can be used. The extension headers may be source routing, fragmentation, authentication and security etc. IPv6 has currently defines six extension headers: (1) Hop by hop option header, (2) Routing header, (3) Fragment header, (4) Authentication header, (5) Encrypted security payload header and (6) Destination options header. If one or more extension headers are used, they must the order in which they are presented above. For example, if Authentication header and routing extension header are to be used, the extension header fields must follow as: (1) main IPv6 header, (2) routing extension header (3) Authentication header and (4) TCP header with data. Each extension header must have one 8-bit next header field. For all extension headers except the fragment header(as in case of fragment header the flags and offset is 16 bits fixed), the next header field is immediately followed by a 8-bit extension header length that indicates the length of current extension header in multiple of 8 bytes. In the last extension header the next header field contains the value 59. The example that we considered earlier, the next header in main IPv6 packet will contain the routing extension header, the next header field in the routing header will show the authentication extension header, and the next header field of the authentication header will contain the value 59. • Hop Limit (8 bits): This field indicates the maximum number of hops that the datagram is allowed to traverse in the network before it reaches its destination. If after traversing this maximum number of hops the data gram does not reach the destination, the datagram is discarded from the network. The field is used to avoid the congestion that may be caused by the datagram. Each router decreases the hop limit by 1 while releasing the datagram to the network. When the hop limit reaches 0, it is deleted. The hop limit of IPv6 is exactly what is called Time To Live in IPv4. The new name of Hop Limit has been given as the name suits better to its function. • Source Address and Destination Address (Each 128 bits): Both the addresses can be called IP address and are described in RFC 2373. IP address that defines the original source of datagram is called source address. The IP address that defines the final destination of the datagram is called the destination address. The three main groups of IP addresses are: unicast, multicast and anycast. Unicast address defines a particular host. A unicast packet is identified by its unique single address for a single interface NIC (Network Interface Card), and is transmitted point-to-point. A multicast address defines all the hosts of a particular group to receive the datagram. The anycast address will be addressed to a number of interfaces on a single multicast address. The anycast packet therefore goes to the closer interface and does not attempt to reach the other interfaces with the same address. A multicast packet, like anycast packet has a destination address that is associated with number of interfaces, but unlike the anycast packet, it is destined to each interfaces with that address. Unlike IPv4, IPv6 addresses do not have classes. But the address space of IPv6 is subdivided in various ways for the purpose of use. The sub division is done based on leading bits of addresses. The present division of IPv6 address space is as shown in table(XIII). The IPv6 address space is huge enough. So a portion of the IPv6 is reserved for computer system using Novell’s Internet Packet Exchange (IPX) network layer protocol, as well as the Connection Less Network Protocol (CLNP).

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It is found that several fields present in IPv4 are no longer present in the IPv6; and notably among them are: • Checksum field. The main issue of designing IPv6 was the fast processing of packets. This results in designing with fixed header fields and removing the redundant fields. The error check is done at upper layers namely TCP/UDP. As such the check sum field further at IP layer was assumed as redundant and accordingly it was removed from IPv6. Again with check sum at IPv4 packet, the error checking at every node was essential. It was a very time consuming and costly thing and duly unwanted at IPv6. • Options field. Dropping of options field has made the IPV6 a fixed header packet. Of course if required the IPv6 packet may use next header field for the purpose of header extension. • Fragmentation. The IPv6 version has dropped the fragmentation and reassembly feature at intermediate routers. The data is fragmented for packetization at the source only. The reassembly is done at destination only. If a IP packet received by any intermediate router is found as too large to be forwarded on the outgoing link, the router simply drops the packet; and in turn send a ICMP error message of “Packet Too Big” to the sender. Sender on receiving the ICMP error message of “Packet Too Big”, retransmit the data with smaller packet size. Actually the fragmentation and the reassembling the datagram at routers is a time consuming matter; and removing these from routers’ functions to end users’ functions, makes the network to speed up. ICMP (Internet Control Message Protocol) ICMP for version IPv4 is used by hosts, nodes, routers and gateways to communicate network layer information to each other. ICMP is specified in RFC 792. ICMP information is carried as IP payload like TCP or UDP information. ICMP messages are basically used for error reporting among others (Table XIV). An ICMP message is made of a type field and a code field and also the first eight bytes of the IP datagram for which the ICMP message is to be generated in the first place so that the sender can know the packet that caused the error. A new version of ICMP defined for IPv6 in RFC 2463. The new ICMP has the reorganized existing types and codes as well as added new types and codes. The added new ICNP type includes “ Packet Too Big”, and “unrecognized IPV6 options” among others. Auto configuration and multiple IP addresses IPv4 address structure is a stateful address structure, which means that if a node moves from one subnet to another the user has either to reconfigure the IP address or to request for a new IP address from DHCP (Dynamic Host Configuration Protocol). With DCHP, an IP address is leased to a particular host or computer for a defined period of time. But IPv6 supports a stateless auto configuration whereby on moving from a subnet to another subnet a host can construct its own IP address. This is done by host on adding its MAC (Media Access Control) address to the subnet prefix. IPv6 also supports multiple addresses for each host. The addresses can be either valid, deprecated or invalid. With valid address new and existing communication may be done. With deprecated address, the existing communication may be done. With invalid address no communication is done. Address Notation Like IPv4, the IPv6 has special notation for representing the IP addresses. The IPv6 address is represented by hexadecimal colon notation. The 128 bits are divided into eight sections

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each of two bytes in length. Each of the eight sections is represented in four hex digits (or a pair of hexagonal numbers separated by a colon. A pair of hex means a byte) and is separated by a colon. One example is: AB12:0978:CF56:00FE: 1234:127E:CB65:7890 The notion allows to drop leading zeros. This means and for example 0045 can be just represented as 45, and 0A456 can similarly be represented as A456, and 0000 as simply 0. The notion also allows removing a zero leaving a colon, and therefore for example 2456:AC67:0:0:67:D4E5:A456:A678 can be written as 2456:AC67::67:D4E5:A456:A678. The stated double colon notation can be used at the beginning or at the end of an address but only once. The double colon at the start indicates leading zeros and that at the end indicates contiguous zeros at the end. If more that one location double colons are used, it will not possible to know how many zeros are there at a particular double colon location. This is why double colon notation is used only once. By counting the other bytes, the number of zeros at the single double colon location can be found out. IPv6 and IPv4 address compatibility For a long interim period, the IPv6 and the IPv4 have to coexist. During this period, an IPv4 address can be converted to an IPv6 address by pre pending 12 bytes of zero. For example, an IPv4 address 126.34.67.10 will be converted to an IPv6 address as 0:0:0:0:0:0:0:0:0:0:0:0:126.34.67.10 or::126.34.67.10. Similarly a host having an IPv4 address as 128.67.56.9 may be mapped (read as IPv4 mapped IPv6) could have an IPv6 address as ::AC45:128.67.56.9. The different special notations of version 4 and version 6 will make them separable. Version (4 bits)

Priority (4 bits)

Flow label (24 bits)

Payload Length (16 bits)

Next Header (8 bits)

Hop Limit (8 bits)

Source Address (128 bits) Destination Address (128 bits) Next header

Header length

Variable header fields …

………………………………………… Next Header

Header length

Variable header fields………….

………………………………………….. Variable length TCP pack (which is TCP header + Payload)……. Fig. 14: Illustration of use of Next Header Fields

Table 11: Reports of different studies on IPv4 address space run up Study group

Recommendation

Two leaders of IETF Address Lifetime Expectations (ALE)’s recommendation

IPv4 address space would be exhausted in 2008 and 20018 respectively

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Final recommendation of ALE in 1994

IPv4 address space will be exhausted at some time between 2005 to 2011.

American Registry for Internet Numbers (ARIN)’s report in 1996

All class A Address has been assigned; 62% of class B address and 37% of Class C address have been assigned

Table 12: Different IP version labels Value

Key

0

Description Reserved

4

IP

Internet Protocol (RFC 791)

5

ST

ST datagram Mode (RFC 1190)

6

SIP

Simple Internet Protocol (IPv6)

7

TP/IX

TP/IX: The Next Internet

8

PIP

The P Internet Protocol

9

TUBA

TUBA

10-14

Unassigned

15

Reserved

Table 13: IPv6 address space subdivision based on prefix assignments of bits Prefixed bits

Use of address space

0000 0000

Reserved

0000 0001

Unassigned

0000 001

Reserved for NSAP application

0000 010

Reserved for IPX application

0000 011 0000 1 0001

Unassigned Unassigned Unassigned

001

Aggregatable Global Unicast address

010 011 100 101 110 1110 1111 0

Unassigned Unassigned Unassigned Unassigned Unassigned Unassigned Unassigned

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1111 10 1111 110 1111 1110 0

Unassigned Unassigned Unassigned

1111 1110 10

Addresses for Link Local use

1111 1110 11

Addresses for Site Local use

1111 1111

Multicast addresses

63

Table 14: Selected ICMP messages ICMP type

CODE

Remarks

0

0

Echo reply (to ping)

1

0

Destination Network unreachable

2

1

Destination Host unreachable

3

2

Destination Protocol unreachable

4

3

Destination Port unreachable

5

6

Destination Network unknown

6

7

Destination Host Unknown

7

0

Source quench (Congestion Control)

8

0

Echo requested

9

0

Router advertisement

10

0

Router Discovery

11

0

TTL expired

12

0

IP header bad

3.22.2 Voice over Internet or Internet Telephony Internet has established itself as the most important and the single most tool of global information age. It was developed for transporting packet data, a non real-time service. But today, Internet telephony has emerged as an important technology. Internet telephony is supposed to carry real time and jitter-free voice over Internet. Active and hectic researches are being carried over the subject of VoIP (Voice over Internet Protocol). Generally speaking the use of Internet for all real time services, like voice, video, and multimedia is being explored. Table (15)[12] shows a growth estimation of VoIP traffic. There are several motivations[11] for transmitting voice over IP. These are: (1) long distance calls at low cost and may be of low quality, (2) cheaper two in one service, (3) Use of PC as a true multimedia terminal, (4) one connection for all services, (5) local exchanges can support telephone with Internet as backbone and without high investment in expensive back bone infrastructure, and (6) use of packetized voice allows voice compression that in turn decreases transmission time and cost. Earlier, telecommunication traffic or telephony connections outnumbered the data traffic. The future is to see the explosion of data traffic. When there will be crossover is debatable, but sooner or later data

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traffic will dominant the telecommunication traffic. “ Consequently, now should be the time for datacom to act as a carrier for telecom.” But Internet, as such cannot be used to carry real time service as it was designed to carry data and as the characteristics of real time services like voice and video are different from data. Table (16) shows the different characteristics and different requirements of voice, video and data The need to deploy Internet for the real time services like voice and video, have lead to redesign some features of Internet. The important two features related to this emerging issue are: (i) redesign of IP datagram format, and (ii) to use RTP (Real Time data transfer Protocol) and IP for carrying voice over conventional IP datagram and Internet. It is believed that with deployment of Ipv6, VoIP will be reached. Table 15: Projected growth of IP telephone (A) As per[12] Voice IP Traffic 1998

310 million minutes

1999

2.7 billion minutes

2004 (expected)

135 billion minutes

(B) As per [16] Year

Average unit (millions per year)

Unit growth rate (%)

Yearly revenues (millions)

Yearly revenue growth rate (%)

2000

3987.2

256

388.75

209

2002

22,386.2

162

1511.07

136

2004

167,896.2

114

8814.55

88

2006

587,636.9

75

22036.38

46

Table 16: Characteristics of different services Voice

LAN data

Transactional Data

Video

Predictability

Constant/On-Off

Bursty

Highly bursty

Constant/Bursty

Bandwidth/ Bit rate

Very Low to Low

Medium to High

Low to Medium

High

Delay/Jitter

Sensitive

Tolerant

Tolerant

Sensitive

Loss

Sensitive/ No recovery

Sensitive but can recover

Sensitive but can recover

Very sensitive/ No recovery

Error/Integrity

Can tolerate

Can not tolerate

Can not tolerate

May tolerate

Technical problems of voice packet transmission over Internet PSTN (Public Switched Telephone Network) based on circuit switching provides voice service with guaranteed quality of service. This is not the case in case of voice service provided by Internet that acts on packet switching. Many technical challenges the voice packet faces while

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in transition over packet switching network like Internet. These include packet loss, packet transfer delay and jittering delay. Voice communication is involved with human interaction. As such, a few losses of the voice packets could be tolerated due to human intelligence and perception involved in recovery. But too much loss of the voice packets may seriously degrade the voice quality. Moreover, PSTN is a reliable voice service provider whereas Internet is not, as because Internet is datagram based. Table 17: End to end voice packet latency delay Delay source

Typical value (end to end or Phone to Phone) in ms.

Recording

10-40

Encoding/Decoding (CODEC)

Each 5-10/Both together 10-20

Compression/Decompression (SPEECH)

Each 5-10/Both together 10-20

Internet Delivery

70-120

Jitter buffer

50-200

Average

150-400

Delay is the more serious issue for real time interactive services like voice. By delay it is meant that the time difference between the time the sender releases the packet to the network and the time at which the receiver receives the packet from the network. Delay refers to: (1) total transfer delay of a packet that includes coding/decoding delay, propagation delay, transmission delay, node processing and queue delay, switching and routing delay; and (2) jittering delay that refers to the phase delay between two successive packets. Typical delay from different sources are as in Table (17)[12]. If the total delay exceeds a certain value, customers may get irritated to the service. A statistic says that a delay up to 80 msec between the caller and callee is acceptable but beyond it causes irritations to the users. The total delay is a variable quantity, and it varies from packet to packet. The jittering delay is very serious issue. If the phase lags between the voice packets at the source and destination varies, the service quality degrades. The phase lag between packets differs from the source end to the destination end because the total transfer delay varies from packet o packet. Due to jittering problem, a sending voice “I shall go home” may be received as “I shall go home”. Compared to the transmitter, the phase delay between “i” and “shall” has increased and that between “shall” and “go” has reduced to zero at the receiver. While the total delay could be limited by increasing the bit rate capacities of the link and by adopting efficient routing technique among others, the jittering effect can not be solved so simply. There are several techniques to reduce the affect of the jittering problem. One such technique is known as accelerating and de accelerating. In fact the jittering problem is due (Di + 1 – Di) which is finite and a variable. Here, Di + 1 and Di are both variable quantities and represent respectively the total transfer delay of (i + 1)th packet and ith packet. To avoid the jittering effect, it is required that Di + 1 – Di = 0. In the accelerating and de accelerating technique, at the receiver end a variable delay (say Wi for ith packet) is caused to each packet such that Di + Wi = K, a constant for all packets (i.e. for i = 0, 1, 2, 3…..) before delivery of the packets to the terminal equipment for play back. By the process, the variable delay caused by the network between two successive packets is made zero as (Di + 1 + Wi + 1) – (Di + Wi) = 0. This ensures that the phase delay between packets at the transmitter remains same at the receiver. The scheme is illustrated in the Table (18). As illustrated in the table, the success of the technique depends on the choice of K.

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Table 18: Illustrating of accelerating and de accelerating technique to cope up with the problems of jittering Instant at which a packet is released at the transmitter (xi) in ms th.

Variable delay with which the packet reaches the receiving node in ms (Di)

Variable delay (Wi) caused at the receiving buffer (100–Di) in ms (K has been chosen as 100 ms)

Delay with which the packet is delivered to the terminal device (xi +100ms)

Packet-1

0

80

20

100

Packet-2

10

70

30

110

Packet-3

15

85

15

115

Packet-4

25

100

0

125

Packet-5

30

110

–10

130

(packet-4 is the marginal case. Packet-5 is the failed case. Both could have been avoided had the constant K been chosen more than 110 ms in this case. So the success of the technique depends on the choice of fixing K) VoIP is going to be a dominant service issue of IP. VoIP has several motivations as we discussed earlier. PSTN supports only toll-quality sound (4 KHz sound), and not suitable for high-fidelity sound. VoIP can support higher grades of sound. This will be another major driving factor for VoIP. But there are several issues that need to be resolved before VoIP is used. Standards are still not finalized, although H.323 of ITU is being projected as a possible standard. H.323 may be under new version 2 be used for interoperability between different service networks like PSTN and Internet to support voice. The standard H.323 is for multimedia or videoconferencing. The audio G.7xx standard of H.323 may be many based on choice of xs. The choice of xs will define the intelligibility of the voice service provided. 3.22.3 Ipv6 for real time services The conventional packet switching is not appropriate to carry real time services. There are many reasons for this. For example HDLC or SDLC packets are variable in size. To synchronize and identify a packet, flags are required to be located. To avoid occurrence of flag byte in the payload, stuffing and de stuffing are done. These cause huge node processing delay, and hence packet transfer delay. ATM was proposed as the replacement of packet switching to support real time services. The problems of conventional packet switching were solved in ATM by making ATM packet, called cell simpler. The simplicity in ATM is in two respects: (1) shorter cell and (2) fixed size cell. This philosophy was extended to design Ipv6 datagram to replace Ipv4 datagram so that IP can carry real time services. IPv6 has a simple and basically fixed header format. The overhead bits of Ipv6 are less than that of Ipv4. The overhead bits in Ipv4 is 12 bytes in the header format of 20 bytes (8 bytes are for address), whereas the overhead bits in Ipv6 is 8 bytes in the header format of 40 bytes (32 bytes are for address). IPv 6 proposes to provide QoS (Quality of Service) service support to real time services like voice and video. The flow level and priority in the header of Ipv6 facilitate the support of real time data. Ipv6 has an efficient header format compared to Ipv4.

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3.22.4 Wireless Internet Two proposals for further development of the Internet are: (i) under sea super speed Internet and (ii) wireless Internet. A proposal for a global optical-fiber under sea cable network called Project Oxygen has significant industry support and financial backing. This project is called “the best of bandwidth on demand” project as per the company release. Experts say “Project Oxygen is the most ambitious communication project in the 20th century…..The Internet and video transmission are the major drivers for the expansion…..a global optical fiber network could erase the boundaries between Internet and the traditional communications, and shift the profit model from voice service to data and video.” Construction of the under sea network began in September’98. In the first phase, the cable shall be stretched over 158000 Km in 74 countries with three major network management centers in USA, Spain and Singapore. The major transatlantic and transpacific links are likely to be operational by 2000. Phase two shall start in 2002 and cover the whole of the world. The speed of cable is projected at 1920 GBPS with minimum capacity of 640 GBPS. It is reported that with under sea Internet, a video-based Internet shall come with over 10,000 video channels. The growth of wireless technology is immense. At the same time Internet traffic is growing exponentially. These have motivated for wireless Internet access research and development. The proliferation of the Internet-enabled wireless devices[56] has also excited the wireless Internet project. The wireless Internet access to mobile subscriber of UMTS (Universal Mobile Telecommunication Systems), GSM and other 3G and 2G technologies; and even with wireless ATM, are studied in literatures[46,54,57]. The modification of RSVP (Resource Reservation Protocol)[58] is directed to implement wireless Internet that can support for different broadband services. The wireless Internet will provide a scalable global mobile system for different services.

4 LOCAL LOOP TRANSPORT TECHNOLOGY 4.1 Fiber-free optical or Optical Wireless Communication and Networks It is often said that science is back to basics. Old science is science of light. Modern science and technologies are now finding and exploring the viable and potential applications of light. The two pillars of the information technology, namely computer and communication see their brighter future in use of light. The future computer technology is heavily dependent upon optical storage because of getting higher density, lower error and higher speed etc. Even the secure transportation of computer data is seeing a high hope with quantum transportation and computing that uses nothing but a single photon characteristics. The communication technology prefers fiber optics as the best choice as transmission medium because of fiber’s several advantages like low noise contamination, and very high bandwidth etc. But the cost of fiber optics is considerable. So if optical communication is made possible without fiber? Grapes the technology and use it. That is what is “fiber-free optical communication” or “free-space optics” (FSO) or “optical wireless”(OW) as known in industries. FSO appears nothing new, but the clever and intelligent application of ancient basic technology of use of light for communication purposes. Remember the early men used to use light signal or smoke signal for messaging with no cost as light travels through air with no cost. FSO technology has the attraction that “it’s cheaper to beam data through the air than to build infrastructure with wires.” Light through free-space or air provides high speed transport over short distance and that too at no cost. This transmission medium may be used with proper transmitter and receiver to realize FSO. Thus for economic advantages the free-space optics

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technology may be used for Gbps (giga bit per second) transport over metropolitan or city distances. The appealing other advantages of FSO are: no cable cost, no cable installation, trenching & digging cost, no cable maintenance cost, and no link failure (virtually link availability is almost 100%!). It is said that ““Free space optics really only provides a very limited application when you consider five 9s of reliability. Some of the free space optics companies will tell you that the five 9s are outdated and that they actually have trials with alternative operators that are just going to three 9s and four 9s” and “Five 9s is probably the greatest myth that exists today in the world of telecom.” Free-space optics is the hybrid of the optical and the wireless technology, presently the two most important carrier technology of communication. FSO offers free-for-all transmission medium. A study says “FSOs also offer lower deployment costs and reduced installation time compared with metro fiber builds. Business cases we have seen start at one-fifth the cost of metro fiber and can be six months faster to install in some metro areas.” As the name implies, FSO uses optical laser technology to transmit data across open spaces and uses the property of straight line propagation of the light beam. The low-power infrared beams that do not harm the eyes are used in FSO technology to transmit data through the open space between transceivers. The transceivers are mounted on rooftops or behind windows (Fig. 15) which are in line of sight with each other over the distances of a few hundred meters to a few kilometers. The part of the electromagnetic spectrum above 300 GHz that includes infrared is unlicensed and available free of cost. The FSO technology then is to ensure only that the radiated power does not exceed the standard defined by the International Committees. Usually the equipment works either at the 850 nm or the 1550 nm laser. Lasers of 850 nm are much cheaper than those of 1550 nm. But the safety regulations permit the lasers of 1550 nm to operate at higher level than that of the 850 nm laser. The FSO with 850 nm laser thus suitable for moderate distance whereas FSO with 1550 nm is favored for distance of kilometer ranges.. Actually 1550 nm has two fold power advantages and five fold distance advantages over 850 nm laser but about ten fold cost disadvantages compared to 850 nm. Table 19 gives a comparative study. A few major applications of FSO are in the areas of metro network extension, last-mile access, enterprise connectivity, dense wave division multiplexing services, SONET ring closures, wireless backhaul, back up, disaster recovery, service acceleration, storage-area network and LAN interconnectivity. FSO may be deployed to extend the existing Fiber Ring of MAN (Metropolitan Area network) by connecting with other networks. This may compete with SONET (Synchronous Optical Network) network. FSO may be deployed in the last-mile access in the sense that it may be used in high speed links that connect Internet service providers or other networks with end users. It is reported that “domestic service providers and foreign carriers are using FSO not only as a broadband backup but also as a viable last-mile technology. For a technology that depends on straight lines, free space optics is taking a circuitous route to espectability.” FSO may be used as redundant back up in lieu of a second fiber link, particularly over short distance communication. This has a clear advantage. Consider the Sept’11 disaster. Had there been FSO, some means alternative communication could have been available in case of fiber failure. A report goes on saying “While FSO will never defy the laws of physics, it can provide a valuable last link between the fiber network and the end user-including as a backup to more conventional methods. A key example was the Sept. 11 tragedy, when carriers learned that having a backup fiber optic network was of little use if both fibers went dark.” AS a backhaul, FSO may be used to carry cellular telephone traffic from towers back to fixed wire PSTN (Public Switched Telephone Network).FSO may further be used to provide immediate or instant service to customers while their fiber link is being laid.

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FSO or OW has another important application in the last mile solution for broadband services. This application is otherwise known as bridging technology. To support broadband services to residential customers, the problem of the last mile made of twisted wire pair exist. The clever utilization of last mile has made the access rates to vary from 128 Kbps to 2.3 Mbps. One important technology of the clever utilization is the DSL (Digital Subscriber Line) technology which provides access rate at 144 Kbps. With OW technology the access rate is believed to increase to Mbps. This a great offer of OW technology. FSO technology is believed to change the optical communication and “Optical networking technology is radically changing the foundation of carrier backbones, boosting Internet bandwidth exponentially while slashing costs dramatically.” But FSO is not free from disadvantages. FSO link may suffer from weather conditions, for example the Fog may hamper the link operation. Till date no standard is available for FSO operation. The vendors have to do a lot to utilize the technology’s viability and consequent products’ marketability. Let us hope for the best for this old technology. It is concluded with a few observations of some industrialists and members of academic: 1. Optics Alliance and chief technical officer for vendor fSONA said: “To have alternate paths using free space optics is getting much more interest from carriers.” 2. “People are realizing that if they have two fibers, they’re not necessary protected if it’s a correlated event and they both go out,” said Steve Mecherle, chair of the Free Space 3. Michael Sabo, senior vice president of sales and marketing for vendor AirFiber, said “FSO is earning a place as more than a fiber backup. Billions of dollars have been spent on longhaul fiber builds out on the trunks. This technology fits the last-mile kinds of applications to fill in all the leaves of those networks.” 4. “Qwest uses FSO in commercial deployments because they are the vast majority of the users of Qwest’s broadband network. We’re pleased with the technology, but we cannot [speculate] about its future deployment in the Qwest market,”Qwest Communications. 5. “Nevertheless, fiber doesn’t go everywhere, and it can’t always be deployed quickly. In all those cases, FSO is a superb alternative,” Werne , CEO, Utfors, A Sweden broadband carrier 6. Ken Corriveau, Tribal’s IT director: “You could rent dark fiber, but that would take forever to figure out in the city. You could rent a T-1 or DS-3, but both of those are 30 to 90 days out.” 7. “In Madrid, 80% of business users are within 500 meters of fiber,” said Paul Kearney, Alua’s (a carrier in Spain)chief technical officer. He further said : “We plan our [FSO] network by using very short ranges to be within the weather limitations.” 8. “In general, the technology has a lot of future for the carrier networks—if it’s marketed well,” said Gartner’s Tratz-Ryan. And therefore to many, ”at least FSO cleared the first hurdle on its circuitous obstacle course” Table 19: Comparison of lasers used in FSO Laser in FSO

Typical Cost

Typical Data Rate

Typical Coverage distance

850 nm

US$ 5000

10-100 Mbps

A few hundred meters

1550 nm

US$50000

Upto Gbps

1-2 Kilometers

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T T

TR

T T

TR

T MAN/LAN TR

TR = Transceivers Free space or air ______ Fiber link

Fig. 15: FSO operation.

4.2 DSL Technologies: ADSL and VDSL Over more than a century, a vast analog telephone network is existing world over. The telephone line in general and the last miles, in particular is made of twisted copper wire pair that is suitable for voice communication. Due to information technology, the need was arisen to transport many diverged types of information. Information relates to many different applications and services, viz. voice, video, data, image, facsimile etc. The information age is motivated by a new culture of value added communication where communication of video, data, image, facsimile and graphics etc has become imperative besides, basic communication of the voice or speech. The characteristic of diverse services, voice, data, video, image, graphics and facsimile etc are quite different from each other. Therefore for each of the services logically there is a requirement to have each one’s nature based communication system. Obviously, such a proposition is not techno-economically viable and sound. The only other economically viable alternative left was to find techniques to use existing vast copper cable of telephone network for the value added services. Even after evolution of fiber optics, a vast copper line is still existing that can be guessed from the following statistics of 1997[35,36] 1. In USA, the unloaded twisted wire pairs up to 18000 ft (between central office and customer - local loop) account around 70% of all loops. 2. In USA, the Loaded loops (>18000ft) account around 15% only of all loops, 3. In USA, the derived loops up to 12000 ft with unloaded twisted pair connected with FTTC / DLC, Fiber - To - The - Curb or Digital - Loop - Carrier, accounts around 15% only,

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4. The world picture in this respect is 600 million unloaded twisted copper wire pair versus 6 million hybrid fiber/coaxial lines, i, e the ratio is 100:1, 5. The annual growth of telephone network in 1990-95 in Africa, Arab States, Latin America and Asia Pacific was respectively 8%, 9%, 10% and 27% 6. Around 1000 million telephone subscribers exist in the world in 2003. Actually the varied services like video conferencing, video on demand, fast access to Internet and interactive multimedia services require higher bandwidth than that of voice. Therefore new technology and signal processing are prime needs if the copper is used to carry these services in the last miles. xDSL (Digital Subscriber Line) is the unique technology that supports more than one services like voice, video and data simultaneously over a shared access line of copper. The DSL is established as a scalable service that provides quality service delivery and at the same time provides a cost effective local loop infrastructure. The DSL appears to be an efficient solution for providing multimedia services. In order to provide value added services, broadband services and multimedia services using existing unloaded telephone lines, over the last few years communication engineers developed a number of techniques. These are: Modem culture and xDSL technology. XDSL technology[38-40] includes: HDSL (Higher-rate Digital Subscriber Line), ADSL (Very-high-rate Digital Subscriber Line), G.lite (splitter less ADSL—this is also called UDSL, Universal DSL), SDSL (Symmetric DSL), VDSL (Very High Rate DSL), IDSL ( ISDN DSL), RADSL (RateAdaptive DSL) etc. 4.2.1 Modem Versus xDSL Using modem, the copper wire provides the data services. For example Internet access with dial up facility is done through the modem. As of today the modem speed is 56 Kbps. The speed of 56 Kbps is not sufficient to support high quality broadband services. Moreover modems occupy the entire 0-4Khz bandwidth allocated to voice, thereby preventing simultaneous services of voice and data over copper of local loop. Within last few years, the slogan of communication technology has become “Speed is the ultimate.” Technology is being developed in pace to serve with the demand of more and more data rate, namely, from bits per second (bps) to Kbps to Mbps to Gbps and finally, to Tbps with WDM, Fiber amplifier, solution and fiber optics communication in hand. High bit rate communication is not possible with copper twisted wire per. Alternative may be to use optical fiber link at loops. This may be a long run solution, but xDSL technology was developed out of this race of more and more fast data communication but using copper cable. The oldest technology of communication digital data through along twisted pair cable of telephone loop, is modem technology. Bit (oldest modem, example is V.21/Bell 103) to as high as 33.6 Kbps (as predicted in V.34 extended standard). With standardization of V.43 modem with 28.8 Kbps, it has been postulated to provide low graded multimedia service to customers through POTS. But there is a big bug in modem technology. A 3 KHz voice line (local analog loop) with 30 dB signal-to-noise ratio can have maximum bit rate of about 30 Kbps as per Shannon theory. Thus using modem technology to carry data over analog telephone line is handicapped by the above speed constraint. This is the reason that modems sometime do not work at the vendors’ advertised speed. There is also report of 56 Kbps modem technology, which can well fit to carry multimedia and Internet services to customers’ promises using telephone lines. But the 56 Kbps technology does not communicate data in between two modems. It communicates data between a modem and a digital ISP (interface signal processor) system which creates a reduced noise like environment. Therefore use of 56 Kbps modem technology to transport high bit rate services to customers, promises may be the limit.

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It was already mentioned that local loops of copper twisted pairs designed for caring voice signals, are not suitable to carry high speed digital data. Local copper loops are primarily designed to carry voice traffic. Voice traffic is relatively short duration and on an average of 3 minutes. Internet traffic is on average of 30 minutes duration. The impulse noise and pulse dispersion of copper loops is the main obstacles in carrying data at high speed. But with growing World Wide Web culture and demand of multimedia services like video-on-demand, boosting of capacity of copper twisted pair local loops, by using some alternative technology of modem, was felt essential. This gave the birth of xDSL technology in general and ADSL technology in particular. It is often said that ADSL is for boosting the capacity of installed copper and fiber optics link. In xDSL technology, special circuits and software called transceiver are used. Transceiver software perform the function of encoding/decoding or modulation/demodulation by which serial binary digital data streams are converted into signal suitable for transmission through analog copper twisted pair link. Transceiver also performs the other functions like equalization, signal shaping and processing, and amplification to compensate for signal attenuation and phase distortion. The other important function performed by transceiver is error detection and correction of data. 4.2.2 ISDN versus xDSL technology ISDN (Integrated Services Digital Network) was developed to provide integrated and simultaneous services of voice, data and low speed video at a basic rate signal of 144 Kbps. The payload of 144Kbps consist of two B channels each 64 Kbps and one D channel 16 Kbps. The DSL signals was first coined to carry 144 Kbps of ISDN over copper loops of 18000 ft or less. This was made with 2BIQ four level line code. The 2BIQ code provides baseband signal spanning from zero to voice frequency band. In this mode of ISDN, voice is served in digital mode using PCM (Pulse Code Modulation) and B channel at the rate of 64 Kbps; but ISDN does not support POTS(Plain Old Telephone Service). Data at the rate of B channel of 64 Kbps (which is much higher than the maximum permissible rate in MODEM culture—about two folds) is served in ISDN. Therefore, why to go for xDSL ? Reasons behind going for xDSL technology are two. First, xDSL technology provides much higher data rate than ISDN. With growing web culture and demand of multimedia services, bit rate of the order of a few Mbps become common. Services like video on demand can not be meet with 64 Kbps or even of 64*2 = 128 Kbps of ISDN technology. Second, ADSL and VDSL are different from ISDN in the respect that unlike ISDN, they retain the service of POTS while providing high rate data service. 4.2.3 ADSL technology ADSL technology has become most appropriate technology out of all xDSL technologies. HDSL is a variant of ISDN technology which provides data communication at the bit rate of about 784 Kbps (T1 carrier) over twisted copper paid loop upto 12000 ft. like ISDN, HDSL uses 2 BIQ line code. ADSL technology was developed mainly to provide multimedia service like video-ondemand service and growing Web service. The characteristics of these two service are quit asymmetric in nature. For Web accessing and / or interactive video two ways communication is essential. Out of the two ways communication downstream (towards the subscriber) communication requires much higher bandwidth then upstream (towards central exchange/

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office) communication. This is because, typically Web surfer is more interested in downloaded on uplink request. ADSL technology[37-43] offers higher data rate of say 6 Mbps for downstream data and lower data payload of say 640 Kbps for uplink data using copper installed loop of telephone. In addition, ADSL provides POTS or conventional voice service. As the service nature is asymmetric, SDSL technology got lost to ADSL technology. Due to the asymmetric nature of ASDL technology, it provides an interesting technological benefit. When many wires are squeezed together in a cable, cross talk is inevitable due signal overlapping. In case of downstream data, signal amplitudes are same because they all originate form the exchange. Due to the same amplitude, there is no effect of destruction of weak signal by strong signal. For uplink data, signal may originate from different customer premises, which are the different locations. Therefore signal reaching through wire pairs of a cable may greatly varies in amplitude. But as the cross talk increases with frequency, problem is tackled by limiting upstream data and keeping it at low end of spectrum. This is exactly what is done in ADSL. ADSL technology increases capacity of installed copper link of telephone to 6 Mbps. In the technology data traffic and voice is carried simultaneously. It carries data in digital form and voice in analog form, unlike ISDN which carries both in digital form. ADSL System POTS splitter/filter preserves the 4 KHz spectrum for POTS service; and prevents hampering of POTS service due to any fault of ADSL equipment. The rest available bandwidth of 10 KHz is used for ADSL data communication at the rate 6 BPS for every hertz of available bandwidth. Fig. (16) portrays the operation of ADSL system. The transceiver software of ADSL uses an advances modulation technique known as discrete multitone (DMT) technology. The ANSI T1E1.4 has standardized DMT as the line code for ADSL. DMT divides bandwidth 10 KHz to 1 MHz in 256 independent subgroups each of 4 KHz width. Each of the sub channels referred to as tone, is QAM modulated on the separated carrier. The carrier frequencies are multiples of basic frequency of 4.3125 KHz. The DMT is used in ADSL technology as because it has the unique ability to overcome typical noise and interrupts in the local loop twisted wire pair cable. The ADSL frequency spectrum is shown in Fig. (17). The available spectrum ranges from about 20 KHz to 1.1 MHz. The low 20 KHz is reserved for voice services under normal POTS. To perform bi directional communication, ADSL modems divide the bandwidth in one of two ways: (1) FDM where non overlapping bands are used separately for upstream and downstream links, (2) echo cancellation where for both the upstream and the downstream the overlapping bands are used but separation is made by local echo cancellation technique. Echo cancellation technique is bandwidth efficient. Advanced forward error correction techniques are used to tolerate error bursts as long as 500 msec. A comparison of different DSL technologies is given in Table (20). ADSL is about 400 times faster than most sophisticated modem and 60 or more times faster than ISDN. However ADSL down stream speeds depend on the loop distance as shown in Table (21). But typical coverage distance is about 4 km. Over distances, the natural degradation of data rate occurs. To provide services to customers beyond 4 km, an embedded rate adaptive mechanism may be used.

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POTS Splitter Copper Local Loop

Computer

ADSL Modem

Local Switching Exchange

Network

Pots Splitter

Processing Circuit

Network

Local Exchange

Fig. 16: ADSL System

Pots band

Guard band Up stream band

Down stream band

4 KHz

138 KHz

30 KHz

1.104 KHz

Fig. 17: Frequency Spectrum of the ADSL

The arrangement may be coupled with growing ATM (Asynchronous Transfer mode) network, which is predicted to be a network for multimedia services. Recent advances in ADSL technology promises to transfer data at the rate as high as 50 Mbps to the customers over a short distance of twisted copper pair from FTTC. This advancement is termed as VDSL. ADSL technology and WDM technology support the predicted cyclic nature of analog digital transmission. Table 20: Comparison of DSL Technologies Service / Network

Data Rate

POTS (Plain Old Telephone System) with Modem ISDN ADSL VDSL HDSL IDSL SDSL

28.8 - 56 kbps 64-128 kbps 1.544-8.448 Mbps for downstream 16-640 kbps for upstream 12.96 - 55.2 Mbps 784, 1544, 2048 kbps 128 kbps 800–2000 kbps for downstream 64 - 200 kbps for upstream

RADSL

1.544–8 Mbps for downstream 64 kbps–1.544 Mbps for up stream

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Table 21: Down stream speed versus distance of ADSL technology Distance in feet

Speed in Mbps

18,000 16,000 12,000 9,000

1.544 (T-1 carrier) 2.048 (E-1 carrier) 6.312 (DS-2) 8.448

The major applications of ADSL technology are: (1) Information highway to wide community, (2) High speed to Internet access, (3) Distance learning by the process of video conferencing etc (4) Video on Demand, (5) Video telephony. ADSL was standardized by the ITU-T in recommendation G.992.1 in 1999. The splitter less ADSL known as ADSL lite was recommended in G.992.2. In the ADSL lite the use of splitter in the customers’ premises are avoided at the cost of lower transfer capacity as 1.5 Mbps and 512 Kbps respectively for downstream and upstream. 4.2.4 VDSL Technology Very high speed or high rate DSL technology is the most recent and important addition to the DSL technologies. The technology is believed to provide the bridge between today’s existing copper infrastructures with near future’s future’s entire fiber infrastructure. VDSL modems [140-43] are placed in the customers’ premises and at the end of fiber installation. The end of fiber installation is the neighborhood or exchange point where the fiber link terminates. With the technology, very high speeds are possible on the copper link spanning about 1.5 km between fiber end and customers’ premises with as high as 15 Mbps total in both directions and over a short distance of 300 m or less with 52 Mbps. VDSL offers about 100 times faster tan normal modems. The proposed VDSL can use up to 30 MHz bandwidth compared to 1.104 MHz of ADSL and 300, 580, 1100 kHz for HDSL. VDSL supports two service classes : Asymmetric known as Class I service and Symmetric known as Class II service. Asymmetric service type is compatible to ADSL technology and primarily aims to meet residential customers. Symmetric service aims to serve business purposes. VDSL is supposed to provide broadband services to both business and residential communities on existing copper infrastructure. Data rates of VDSL is at Table (22). VDSL system VDSL is aimed to be coupled with FTTC (Fiber To The Curb) and FTTB (Fiber To The Building)/FTTH (Fiber To The Home), the technologies that uses fiber in part of the local loop. In that context the VDSL reference model is shown in Fig. (18). Table 13: Typical VDSL data rates Service Class

Upstream data rate in Mbps

Downstream data rate in Mbps

Spanning Distance in m

Asymmetric

6.4

52

300

3.2

26

900

1.3

13

1500

26

26

300

13

13

900

6.5

6.5

1500

Symmetric

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Customer Premises

Copper Link

Fiber Link

Central Office/Exchange

VDSL

VDSL Transceiver at NT

Transceiver at ONU

NT = Network Termination ONU = Optical Network Unit (a) System Reference LT = LINE Termination

Splitter

Splitter Copper wire

PSTN/ISDN

NT = Line termination

Network Interface

PSTN/ISDN

(b) VDSL reference model Fig. 18: VDSL System/Model

Attenuation and Cross Talk The subscriber loop is made of copper wire of different gauges. A number of pairs are grouped together in cable bundles. The attenuation of the signal in copper wire depends on the dielectric used, gauge, type of twisting, and length. But attenuation usually increases with both frequency and the length. That is why the data rates in ADSL and VDSL falls wit length as pointed out earlier as well the distance coverage is lower in VDSL than that of ADSL It may be noted tat NEXT is not attenuated by the line transfer function. That is why NEXT is more harmful than FEXT. In both ADSL and VDSL, by FDM technique the effect of NEXT is made lower. But both cause the data rates to fall with lengths. Both the technologies, ADSL and VDSL are believed to provide wider broadband services to residential and business users using the existing copper link of last miles. However future research directions will aim to tackle the issue of falling rates with length.

5. MULTIMEDIA COMMUNICATION AND CONFERENCING STANDARDS It is said that in the future, multimedia shall be the rule and monomedia shall be the exception. Multimedia is a tele-service concept that provide integrated and simultaneous services of more than one telecommunication service, namely, voice-world, video-world and data-world. Truly, multimedia is supposed to provide such service in real time and in interactive mode. Typical examples of multimedia applications are teleconferencing, videoconferencing, telemedicine, telemarketing, teleshopping etc. Multimedia is fast emerging as an important tool of information technology and as a basic tool of tomorrow’s life. Multimedia proposes to simulate human-like communication and services in an environment of “you see as I see” and “you feel as I feel”. Virtually reality is envisaged in multimedia services. Multimedia transferred your message in your way. Multimedia

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is believed to prosper with the general human trend from “nice to have” to “value to have” to “essential to have”. With multimedia a society with “plug and play”, “look and fell” and “point and feel” and “point and click” shall emerge. In near future, we shall have multimedia cities and centres. It is often said that in near future multimedia shall be the rule and the monomedia shall be the exception. Interactive multimedia is a service, which provides simultaneous access, dissemination, transportation and processing of more than one information service like voice, video and data in the interactive mode and in the real time environment. Multimedia is to integrate three communication worlds, namely, telephone world, data world and video/TV world into a single world communication. multimedia application shall comprise more than one information type, namely the non real time service of data, images, text and graphics, and the real time service of voice and video. Future world of information and communication shall be converged to multimedia application and shall provide comfort, competition, mobility, efficiency and flexibility. As per Fred T. Hofstetter “Multimedia is the use of a computer to present and combine text, graphics, audio and video with links and tools that let the user navigate, interact, create and communication.” Technologically multimedia shall be service of services and nontechnically a community of communities”. Multimedia shall enable people to communicate and access at any time at any where at reasonable costs with acceptable quality with manageability. Location of man, materials and machine resources shall be irrelevant in business in the era of multimedia. It is said that “It makes no sense to ship atoms when you can ship bits.” “Virtual reality with virtual presence in virtual worlds, virtual cities, business enters, virtual schools and virtual rooms will emerge in the next future ……… For example, virtual reality at short notice allows collaboration between changing partners on specific tasks, sitting at virtual writing tables without real offices and addresses other than the network. Transactions in this enhanced telecooperative working environment would be electronic analogies of the normal world.” Faster work flow, comprehensive 24-hour service, remote operation and maintenance, easier trouble shooting, life long and leisure time activities, less travel, less cost and more fun shall be the important attraction of the multimedia world. Multimedia communications provide a chickenegg benefits to information world, and have acceptance at all levels: (1) contact acceptance, viz., service availability, user-interface, (2) economic acceptance, viz., less cost, more benefits, (3 ) content acceptance, viz. quality, and (4) social acceptance, viz., desirability, privacy.

5.1 Standards A great challenge is to standardize broadband services and system for the purpose of deployment. In fact, the deployment of seamless integrated mobile broadband services will greatly benefited from the standardization process[48]. In order to define any standard, the International Telecommunication Union (ITU) usually forms a study group. This study group submits recommendations for standards pertaining to the assigned functions. A list of different study groups along with their assigned functions, made by ITU for 1997-2000, is given in Table (23). SG9 and SG16 respectively deal with television and sound transmission, and multimedia services and systems. The low bit-rate (kilo bits per second-kbps) audio coding standards specified by ITU for multimedia application are listed in Table (24). The standards G71X and G72X are mainly used in different multimedia applications. MPEC-I (removing picture export group) audio coding decoding is applied in H.310 multimedia conferencing standard.

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Table 23: ITU Study Groups SG 1 SG2 SG3 SG4 SG5 SG6 SG7 SG8 SG9 SG10 SG11 SG12 SG13 SG14 SG15

Service definition Network and service operation Tariff and accounting principles, economic and policy issue Telecommunication management network (TMN) and network maintenance issue. Protection and policies against electromagnetic environmental effects. Outside plant Data network/open system intercommunications Features and characteristics of telemetric system TV sound transmission Software aspects of telecommunication systems Signal and Protocol End-to-end transmission performance of network and terminals Network aspects in general Modems and transmission techniques Transport networks systems and equipment’s

SG16

Multimedia services and systems.

Table 24: Standards of low bit rate audio coding for multimedia communication Standard G.723,D G.723,1 G729,A G.729 G.711 (PCM/POTS) G.722 (Broadcast quality) G.723 (Low bit rate POTS) G.726 G.728 MPEG.1 layer (CD audio)

Bit rate In kbps

Frame size in mg/cc

Algorithms delay in m sec

Required RAM size with 16 bit words.

5.3 6.3 8 8 56

30 30 10 10 —

37.5 37.5 15 15 —

2.2k 2.2k 2k 2.7k —

48-64

—

—

—

5-6 32 16

— — —

— — —

— — —

32-256

—

—

—

Different video coding standards for multimedia services are listed in Table (25) along with bit rate and applications. H.26X standards are used for videoconferencing and MPEG-I is used for video-on-demand. H.26X standards are mostly used in multimedia videoconferencing standards like H.320, H.324, H.323 and H.310

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Table 15: Video-coding standards for multimedia applications Standards

Bit rate

Typical Multimedia Application

H.261 H.263 MPEG.1 MPEG.2

64kbps-1.92Mbps 15kbps-34kbps 1.2Mbps–2Mbps 3-15Mbps

Videoconferencing (N-ISDON-64) Low rate videoconferencing Video on demand Temperature-Diagnostic video on demand

Table 16: Multimedia conferencing and terminal standards Standard

Network

Video Coding

Audio Coding

Data Standard

Multiplexing

Control

Remarks Application

H.320 (1990)

N-ISDN

H.261

G.711 G.722 G.728

T.120

H.221

H.242

Multimedia conferencing with G.711

H.324 (1996)

PSTN/ GSTN/ POTS

H.263 H.261

G.732.1 G.729

T120

H.223

H245

Multimedia conferencing with H.263 and G.723.1

H.322 (1996)

LAN internets packet switching

H.261 H263

G.711 G.722 G.728 G.723.1 G.729

T.120

H.225.0

H.245

Multimedia conferencing H.261, G.711

H.322

Isoethernet

H.261

G.711 G.722 G.728

T.120

H.221

H.242

—

H.321

B-ISDN/ ATM

H.261

G.711 G.722 G.728

T.120

H.224

H.242

—

H.310

B-ISDN/ ATM

T.120

H.222.0 H.222.1

H.245

Multimedia conferencing with H.262, MPEG.1, H.222.0

H.262 MPEC.1 MPLG.2 G.711 H.261 G.722 G.728

Table (26) is a comprehensive list of different multimedia standards, their network performs, video coding, audio coding, and data standard multiplexing standard, control standard and applications. The standard H.324 may be used to provide videoconferencing, putting to work the existing telephone network. H.323 may be used for the same over LAN (local area network) and H.320 may be used over N-ISDN using nx64 kbps channel, whereas H.310 may be used using BISDN/ATM. The table also lists the users terminal requirement for different multimedia standards.

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5.1.1 H.320 multimedia conferencing standard H.320 is the narrow band (< 2 Mbps) conferencing standard meant for conferencing over telephone networks such as ISDN with bandwidth typically in the range 384 kbps. H.320 family of standard is to serve video conferencing with any H.320 compatible terminal irrespective of whether it is stand alone video conferencing unit or video telephone or PC based system. H.320 is often treated as a de facto standard of video conferencing. H.261 is the video coding standard has a lot of similarity with H.320. The H.261 standard has a lot of similarity with MPEG technique, and uses the DCT transformation technique with motion compensation and Huffman coding [see-Box 4] to active compression. But unlike MPEG, it has rate control to cope up with variable video bandwidth within the rate of 40kbps to 2Mbps. H.261 standard supports two picture sizes: the larger one is called CIF with pixel size of 352X288 and the smaller one is called QCIF with size of 176 x 144. H.320 terminals are having H.261 video code. The audio coding of H.320 standard can be any one of three: G.711, G,722 and G.728. The G.711 is equivalent to a -low and m-low PCM such coding. It supports 3.1 KHz audio at 64, 56 or 48 kbps. The G.722 provides higher quality audio with 7KHz bandwidth using 64kbps. The G.728 is equivalent to a -law and m-law encoding and supports 3.1 KHz bandwidth of 16kbps. The H.221 is the framing standard. The audio and video bit streams are multiplexed together to create a frame that is to be sent. The H.221 define how the frame is achieved. Each frame is made to the 80bytes of information. Each frame creates 8 sub-channels with each bit within each byte allocate to a sub channel. They are numbered 1 to 8. First seven channels are used to carry video and audio data. The 8th channel is used not only to carry data but to carry other codes also. The other codes are: FAS: Frame alignment signal. BAS: Bit-rate allocation signal. ECS: Encryption control signal. These sub-channels are called service channels. The standard H.230 provides frame synchronization control and audio-video signal indication control. H.242 is for achieving capability exchange, mode switching and frame reinstatement. The H.243 is multipoint control standard. The video conferencing is not only point-topoint but multipoint too. To control multipoint conferencing, multipoint control unit is required (MCO). H.243 is a standard for MCU. The T.120 is for communication of all forms of data between two or more multimedia terminals. The H.233 is security coding. It is used to define a method of encrypting data. Different wireless combination have been investigated in Japan[45] to define multimedia terminals: PDC (Personal Digital Communication System) + PHS ( Personal Handy phone System), PDC + 3G, 3G + 4G, 4G + MMAC ( Multimedia Mobile Access Communication), 3G + 4G + MMAC. Such investigation will definitely lead to wireless + integration to coexist to implement true personal communication.

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BOX 4 The Huffman code is a compression code designed by Daceid A Huffman in 1952. It is a simple improved code over Shannon-Fanon code. In order to illustrate Huffman code, let us say we have an original body of data which reads only source triple as in table to present some message. The probability of occurrence of any source triple in the message is also shown. According to the Huffman coding, the corresponding compressed codes are shown in the table. The average size of the compressed code under Huffman coding becomes: 1 X 0.4 + 2 X 0.2 + 3 X 0.2 + 4 X 0.1 + 4 X 0.1 = 2.2 bits per code. Whereas the code size of the original source code is 3 bits per code. Source Triple

Probability of Occurrence

Corresponding compressed word

000 001 010 011 100 101 110 111

0.25 0.25 0.125 0.125 0.0625 0.0625 0.0625 0.0625

11 10 011 010 0011 0010 0001 0000

There are several disadvantages to Huffman coding. First, to design the code, one must know the probability of occurrence of any code in the original block of data. What shall happen if the probability is not known a priori? And what shall happen if probabilities pattern changes over time? Second, Huffman coding is not unique in nature. The code is also block code. But the redundancy under this code is either minimized or optimized

6. UTN PERSONAL COMMUNICATION To offer UTN services, in USA, a band of 160 MHz near 2 GHz has been allotted. Personal communication shall mature with UTN. Frequency band allocation for different services of personal communication is shown in Table (27). Cellular communication is the early personal communication. Personal communication shall coverage to and merge with total wireless, total service independent and total UTN based communication. If we consider the growth and development of wireless communication at the present rate, total wireless (to a constraint) may be achieved within next 5 years. Total UTN service may need another 5-10 years. Integrated and application oriented communication needs new technology and new integrated terminal which shall be affordable to mass customers. Technology is with us with ATM. Integrated terminal is under development. Computer integrated telephone is the first such device. It may require another 5-10 years to commercially develop an integrated terminal for voice, video and data. Therefore a matured PCN (Personal Communication Network) is expected within next 510 years. As per forecast made in literatures, narrow band personal communication service may cover 750000 km2 and 1500000 km2 in USA in next 5 and 10 years respectively. Indian communication is lagging behind international communication by several years. As per account this lag is a uniform 4 years since 1986. GSM started in Europe in 1988, India adopted the in 1994. ISDN started in Europe in 1990, India has adopted only in 1996. On the ATM, BISDN and UTN aspects, India is yet to open its chapter. Hence it is evident that Indian lag is more

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than 5 years and even may be 7-10 years in PCS/PCN. India is lagging behind its neighbors like Singapore, Taiwan and Hong Kong. Table 27: Frequency bands of different PCN services Service

Frequency Band

Cellular

800-900 MHz Ex : GSM - 890-915MHz 935-960MHz

CT-2

864/944 MHz

Cordless

46/49 MHz

Satellite/VSAT/MSS

C band

Narrowband PCS (FCC)

900-940 MHz

Broadband PCS (FCC)

1850-1890 MHz 1930-1970 MHz 2130-2200 MHz

7. FROM 2G TO 3G 2G (second generation) technology for mobile connection started around 1990s and it was revolved around GSM cellular communication that is mainly for voice communication. 3G were then expected to be deployed around 2000 and were targeted towards: • implementing anywhere and any time mobile connection with low cost and flexible handheld devices • implementing wireless data access particularly with wireless Internet connection. This was motivated by the exponential growth of Internet access. Users are prone to get Internet access anywhere and anytime with hand held devices • implementing high data rates at 2Mbps whereas previous GSM or 2G offered to 10 to 50 Kbps • implementing high speed multimedia or broadband services causing shift from voice oriented services to Internet access (both data and voice particularly with technology of VoIP), Video, Music, Graphics and other multimedia services • use of spectrum around 2 GHz whereas spectrum allocation for 2g was 800/900 MHz • global roaming to support global communication • flexible network to support existing and future changing requirement • a mobile multimedia services that will be able to transmit data, voice, video, image etc over variety of networks like point to point, point to multi point, broadcast, symmetric and asymmetric etc. The key benefits of 3G will be: delivery of broadband information direct to users and global access with a unified single radio interface. Several major challenges are to be overcome to implement 3G: wireless Internet for exponential growing users will be difficult to implement till IPv6 is implemented, global roaming with single number as proposed in PCN, fixed access with technologies like ADSL with high data rates of 12 Mbps has become competitor as that of IEEE 802.11 b WLAN in wireless local data interface, low cost flexible devices are yet to mature.

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7.1 Beyond 3G Mobile comprehensive broadband integrated communication will step forward into 4G (fourth Generation) all mobile services and communication. The 4G technologies will be migration from other generation of mobile services with an aim to overcome limit of boundary and achieving total integration. The evolutionary approach towards a wireless information age proceeds as in Fig. (19)[44,47,59] in comparison with other technologies as progress in pace to pace. The key characteristics of 4G systems will be: higher transmission capacities per user, larger frequency band, higher traffic densities, and integrated services. The technical challenges behind the expected technology lie with the associated different technologies as discussed earlier.

2G GSM, PDC, IS95

IG Analog cellular

3G : UMTS, CDMA

Towards wireless communication society Multimedia Content, High Bit Rate and IP Transport

802.11b WLAN

WLAN

Circuit Switched Networks

Wired Internet

802.11a WLAN

Broadband Internet/ DSL

4G Total Wireless, Seamless coverage & Integration, Anytime & anywhere communication

Wireless/Mobile Local Area Integration

Broadband FTH (Fiber to Home)/Fiber to Business

Fig. 19: PCN Evolution /Migration and other technologies as progress in pace to pace

The motivation behind aiming 4G information society are many: high speed transmission, next generation Internet support (Ipv6, VoIP, Mobile IP), high capacity, seamless integrated services and coverage, utilization of higher frequency, lower system cost, seamless personal mobility (LEO), adoption and integration of fixed and wireless support (ADSL/VDSL/ WILL/FSO), mobile multimedia (Standards), efficient spectrum use, QoS service, flexible and re configurable network and end to end IP systems. The convergence of local fixed wired network including wireless home or local network with broadband fixed and coming up ad hoc wireless networks will shape how we will communicate in next decades that may include[49.60]: Complete unification and integration of all and every services, Single communication number for each and every services, and Freedom to communicate any time any where. All these provisions are required to be meet with simplicity, cost effective, reliability and flexibility. The problems to be solved in achieving the expected results are: lack of bandwidth, lack of standardization, high error probability of wireless links,

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multiplicity of different systems & operators, and cost reduction. The problems are being addressed. The research in tackling the high error probability of wireless links has reached the expected directions[50-53] with BEC (Backward Error Control) technique. The research[55] in this context for optimizing Internet access over IEEE 802.11b has demonstrated with frame level FEC (Forward Error Control) technique.

8. e-BUSINESS AND e-COMMERCE–A DURABLE APPLICATION OF IT Noble Laureate of Chemistry, Ilya Prigogine once told: “We can’t predict the future, but we can prepare it.” Certainly the future will be what we make of it with today’s single most technology, IT (Information Technology). It is a technology of the networks, the telecommunications and the computers. IT will make an impact in all aspects of our life. It is believed to bring about a profound change. The success and the effectiveness of the changes will be measured with proper perspectives in future. But it has undoubtedly brought an unprecedented change in business and commerce, by giving time, space and volume continuity. Today business and commerce have no geographic boundary, no volume restriction, and no time limitation. They provide just-in-time and just-on-scale solutions. On the scale of effectiveness, the business is measured by the low of the loss (W) of process cost. If P is the process cost, OPE (Overall Process Efficiency) is the efficiency factor of business, W = P - (P X OPE). With application of IT and its derivative like KM (Knowledge Management), OPE increases causing W to fall. Besides, IT gives the competitive advantages to the business activities. Information technology believes that: Investment + Web technology + Users = Big Profits. In such a scenario, ebusiness and e-commerce have eventually been emerged as the sound strategies for business and commerce. e-business and e-commerce will be facilitated by different technologies like global-reach Internet, WWW (World Wide Web), E-mail, Electronic publishing, Multimedia systems and communications, Interactive video, Image recognition and processing, Voice recognition, MSS (Mobile Satellites Services) and Personal Communication among others. The use of border less Internet is increasing following Moore’s law that estimates the doubling of the performance of silicon every 18 months. Even Internet growth may go well beyond Moore’s law. Gilder’s law may be more accurate estimate for growth of Internet traffic. Gilder’s law predicts the doubling of packet on the network every few months, and that few months may be in the range 4 to 9. It is estimated that the Internet traffic will increase by 1000 fold in the next ten years. As of today, about 50 million users use Internet with about 16 million servers in more than 140 countries. Internet is the best facilitator of electronic mode of business and commerce. Today, e-business and e-commerce refer to the business transaction over Internet. E-Business and e-commerce mean to doing business over wires or over Internet or using Information Technology. They are changing the rules of traditional business pattern, and making new rules and means for fast and border less business. The confusion on the difference between e-business and e-commerce is standing. e-business defined in most of the literatures reflect that there is actually no difference between these two. Yet two things appear to be somewhere and some how different.

8.1 E-Business E-business refers to the operation of the business objectives through and using IT. It may also be defined as business activities over digital infrastructure or doing business over wires. As per Colin, Director of the integration division of CNS, UK the e-business refers to the issue of supply chain integration. “ An ideal scenario is when a customer places an order. All of the

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suppliers and agents involved in the transaction are contacted electronically. Every system involved in the supply and delivery of that product is linked to every other system, hence talk of ‘zero latency transactions’ whereby there is no waiting for someone to do something because everything happens at the speed of light.” A report says, ” E-Business relates to how you and your customers place orders and ensure efficient delivery. E-commerce is the financial aspect of doing business. Both aspects will affect your operations sooner or later.” The economists usually identified four types of e-business: • Business-to-business (B2B). This refers to transaction between one business house to another. For example, the transaction between a large organization and their suppliers falls in the category. B2B is the most common business model. One example of B2B e-business is MetalSite.com • Business-to-customer (B2C). This refers to online retail activities. For example, software, journals and books sold over Internet using web sites. • Customer-to-business (C2B). The example of this is the booking of railway tickets or air tickets on any agent’s computer that has the network or the Internet connection. C2B is just the reverse of B2C. • Customer-to-customer (C2C). Online auction is the best example of this type of transaction. One example is eBay.com Currently e-business is mostly confined to B2B. Other areas of business are of course coming up.

8.2 E-Commerce E-commerce is basically financial transaction via computer networks, between people and organizations. E-commerce is a financial part of e-business.. Harvard Academic, Jeffery Rayport defined e-commerce as “selling real products for real money.” Eddie Rabinovitch observed “ Not surprisingly, the expected pay off of e-commerce projects is, of course, the bottom line: money. However, despite the prevailing notion of access to global markets as the most important competitive advantage enabled by e-commerce, most companies expect of e-commerce ways to reduce spending rather than increase profits. Let’s for a moment think about the rationale of the previous statement, which is also going to answer another e-commerce question: ‘ why is business-to-business (B2B) market considered by many experts several magnitudes more important than business-to-consumer (B2C)?’ Well, it’s probably easier to convince a CEO to spend $100,000 on a solution that will demonstrably ‘save’ $1 million than to spend the save amount on a solution that might ‘make’ $1 million....... Making money on the Internet is still quite dicey. But it’s not too difficult to demonstrate that B2B e-commerce will save money by improving efficiency and therefore reducing expenses for transactions between companies.”

8.3 Problems for e Commerce and e Business Money is the ultimate motive, if not sole of the business. Thus there will not be any compromise on financial transaction. E-business has to deal with flexibility, interoperability, scalability, performance and security of business; e-commerce has to deal firmly with security of the transaction e-commerce improves quality of service and performance. Security of e-commerce is required at two levels: Confidentiality and authenticity. Failure at the security of e-commerce virtually means the failure of the e-commerce itself. Financial transactions are done by several modes: Electronic cash, Electronic cheque, and Electronic transfer and payment advice etc. Security of the transaction in electronic payment system is the key to e-commerce. Public key techniques, Digital Certificate are useful security measures of e-commerce.

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9. KNOWLEDGE AGE AND MANAGEMENT As we move forward and as more and more human-IT interaction plays role in shaping society, an all inclusive knowledge society turns into. Knowledge Management has become a central issue of the knowledge age. What is then Km (knowledge management)? In a theory, “KM is seen as a logical extension society in that its purpose is to cope with explosion of information and capitalize on increased knowledge in workplace.” According to Peter, “The successful companies, in the knowledge management terms are the ones that have looked at the business processes rather than seeing the solution revolving round the company intranet.” According to his research, “the main reasons for using knowledge management techniques are to be competitive. Through globalization, there are a lot more competitors coming into markets quickly. Therefore, you need to do more in order to appear different. Another layer of knowledge is how to integrate things in the organization so that this process makes the organization look different.” His research suggests, “Content management is important.” He mentioned a figure that “ out of 1000 pages of a marketing intranet, 873 pages were not used, the reason being that they were out of date.” KM is not dumping data on the intranet, but for sharing of knowledge and information. British Telecom (BT) is recorded on saying the following reasons for sharing knowledge: (1) knowledge is the basis of services, (2) knowledge helps to cope up with changes, (3) knowledge sharing is the natural next step to information sharing. Alain J Godbout[61] analyzed the concept of KM from views of Peter Drucker, Nonka, Tom Davenport, and the American Quality and Productivity center among others. Following Peter Drucker, he viewed that the KM process “is a question of proper vision, organizational networks, educated decisions and best use of lessons learned as the key to organizational learning.” He further said: “In a sense, knowledge management is a form of application of sound management practices to an object: human resources which are the carrying vector of knowledge.” Tom Davenport and the American Quality and Productivity Centre is believed to emphasize more on explicit knowledge, and “their emphasis is to focus on means of optimizing these holdings (the explicit knowledge of organizations is contained in information holdings), improving the methods of formalization and increasing the use or usability of the available knowledge.” Referring to Nonaka’s model of mental process, Alain said: “ knowledge management is a form of sound management practices to another object: information resources with a different carrying vector of knowledge.” Taylor[62] views knowledge management as a “process of ensuring that the organization’s knowledge needs are met and exploiting the organization’s existing knowledge assets.” DiMattia and Oder [63] defined KM as “ KM involves blending a company’s internal and external information and turning it into actionable knowledge via a technology platform.” It is to note that in the definitions, sometimes the knowledge and the information are used interchangeably. In our eastern philosophy, the knowledge management can be seen in unique terms. As per Bhagbat Gita any action has two components: Karmayog that proceeds along path of action and Sankhyog that proceeds along the path of knowledge. To our philosophy, nature is the best manager. Again the first law of management of nature is the principle of least action and least time that aims to “accomplish most with least effort” and least time. In nature, everything follows the least path of action. Apple falls from a tree to ground on straight line, waters falls down in straight line, and light travels on straight path etc. Business or Organizational Management being a action of man where man is a part of nature, any man as manager desires to accomplish any action with least path and time for which Information Technology is with us today by our own creation. One can see in Fig. (4) how the information technology, which is due to the marriage of Computer with Communication, is tending to be like nature type technology.

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Over time the gap between human axis and technology is reducing. Therefore KM is an action to achieve goals along the path of knowledge with least action both mental and physical, or otherwise to do management so far done absolutely by man by technology in order to go along a path of least action, the path of nature by expanding intelligent technologies like brainy computers and personal communications. Swamiji made a following few comments over nature, man and knowledge: “Nature with its infinite power is only a machine.” “All our knowledge is based upon experience…. All human knowledge proceeds out of experience; we can not know anything except by experience.” “Man is man so long as he is struggling to rise above nature, and this nature is both internal and external.” These observations of Swami Vevekananda imply that man by earns knowledge from experience, and he applies his knowledge to be creator of nature, which is not impossible so long nature is assumed a machine. It will be pertaining to mention here that Tagore told that everything in nature follows a rule. This supplements my views that the KM is a step of human effort where he attempts to be his known creator.

9.1 KM-Conflicts and Confusion Knowledge management appears to be a collection of organizational knowledge in machines whereby the collected knowledge can be shared instantly at anywhere, at any time and by any body for the managerial purposes, be it for the policy decisions or for the routine works. But can the collected knowledge be ever creative? Or the human knowledge, which is ever creative, can be collected? Human knowledge has an ever alive and changing creative dimension. The creativity of human knowledge brings invention and innovation in framing organizational problems and solving organizational problems. Human and computer have their own different merits, and merely are not supplement to each other. Therefore the goal of the KM to capture and keep the knowledge of the employees leaving the organization will how far be successful is not without doubt. The objective of the KM is to share the knowledge at intra organizational and inter organizational level for arriving at a decision. The sharing provides a number of solutions and practices done and/or in operation in the organizations so that some of them could be deployed for the current need. All such sharable solutions and practices are definitely pre-programmed and heuristics in nature. A preprogrammed solution is fitting to a stable environment. A hostile environment may be of innumerable types caused by the wicked environment. How far a pre arranged solution is applicable to processes and problems of wicked environment? One recent example is the fight between ICC (International Cricket Council) and BCCI (Board of Cricket Control of India) over match referee’s report against six Indian cricketers during their second cricket test against South Africa. No pre-programmed knowledge was available for providing a solution. The problem was first of its kind and dragged into a wicked environment. Only human creative empowerment has saved the situation. Yogesh Malhotra[64] authoritatively analyzing the KM in inquiring systems duly highlighted the stated limitation of KM. He showed that out of the four inquiry systems, namely: 1. Leibnizian systems those “are closed systems without access to the external environment: they operate based on given axioms and may fall into competency traps based on diminishing returns from the ‘tried and tested’ heuristics embedded in the inquiry processes”. Example: as per some mathematical models (ESPN ratings, Pepsi ratings ..), the best team and players of test/ one day cricket.

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2. Lockean inquiry systems those “are based on consensual agreement and aim to reduce equivocality embedded in the diverse interpretations of the world view.” Example: Selection board meeting for a cricket team 3. Kantian inquiry systems those “attempt to give multiple explicit views of complementary nature and are best suited for moderate ill-structured problems.” Example: Result of a final match 4. Hegelian inquiry systems those “are based on a synthesis of multiple completely antithetical representations that are characterized by intense conflict because of the contrary underlying assumptions.” Example: Which party is to form government when no party has got majority in any Indian Parliamentary Election! The KM may have the significant role in Lockean and Leibnizian systems as they are “suited for stable and predictable organizational environments”, but the KM will have limitations in applying to other two systems as they “are better suited for wicked environments.” The wicked environments are characterized by discontinuous change, and the information technology has a trend to create wicked environment, it is not yet clear how the KM will suit to information technology driven present and future world. 5. The one of the main features of the KM is sharing of knowledge for improving business process and activities. The expectation and the results from knowledge sharing in many cases, particularly in the environment of competition, however cause havoc. In one final examination the topper of the class and the second topper sat side by side. The topper wanted to share the answer of a problem which he correctly got as, say 60. The topper when asked the second topper, although the second topper got 60 as answer, yet just to confuse the topper he told him that the answer was 50. The topper being confused scrapped out that answer, and tried another; but before its completion the time was out. Consequently, in the result the topper went down to the second position and the second topper moved up to the first position. This shows the possible consequence and counter productive feature of knowledge sharing particularly in competitive business environment. This phenomenon of knowledge sharing may be called “calamity of knowledge sharing.” The calamity may also occur when sub standard knowledge is shared. 6. The more serious conflict of knowledge sharing lies in its very definition. If knowledge is power, if knowledge is saleable, and if knowledge brings prestige, power and authority; why one should share his or her knowledge? The very basics of knowledge do not support the knowledge sharing. This being the case, the KM itself lies under a cover of confusion. Thomas H Davenport described [49] this phenomenon, as “sharing and using knowledge are often unnatural acts.” He felt that “sharing and usage have to be motivated through time-honored techniques-performance evaluation, compensation for example …..Lotus Development, now a division of IBM, devotes 25% of the total performance evaluation of its customer support workers to knowledge sharing. Buckman Laboratories recognizes its 100 top knowledge sharers with an annual conference at a resort. ABB evaluates managers based not only on the result of their decisions, but also on the knowledge and information applied in the decision-making process.” The other type of problem of same nature also exists in the organization. An employee who is an expert in obsolete technology may do not like to share knowledge of expert of new generation due to several reasons like ego, inferiority complex, and fear of being out classed. This phenomenon can be analogically compared with electric circuit as illustrated in Fig. (20). The organization likes to attain at a knowledge level, K. It has a storage capacity, C. But the organization offers a resistance. This

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resistance delays the organization to attain at the knowledge level K. Until and unless the offered resistance is removed by organizational process of transformation, the conflict will exist and resist the implementation of KM. The organization resistance (R) restricts the flow of knowledge.

C >> Storage Capacity of Organization

K

R >> The Organization Resistance (Physical, Mental and Cultrual Resistance)

Fig. 20: An analogy of a conflict

7. KM involves two words: Knowledge and Management. Who is for whom or who will rule to whom is a big question. Does KM mean the management of organization by the knowledge or does it mean the management of knowledge of the organization or a hybrid? This confusion is pictorially illustrated in Fig. (21). 8. Lester C Thurow documented some factual conflict that is existing in the USA: The information technology has been projected as a high productivity in nature. But the Lester C Thurow studies claimed that “ Financial services in the United States have had negative productivity growth for the last ten years. Every year productivity is falling about 1 percent.” His studies on office automation show that offices still use paper in the same ways for the last 500 years. The paper less office or automated office still remains a far cry.

Knowledge

Management

Knowledge

or

Management

OR BOTH ?

Fig. 21: A conflict in picture

Knowledge management is the technology-based management. Therefore its impact and consequences will change with technology and technological trends over time. However it will be not wrong to define KM as a management using computer and communication or for that purpose if we write : KM = MC2 The technology in general and information technology in particular follow a few empirical laws. In that light, we can analyze and predict the future technologies and hence future KM.

9.2 What is there after Knowledge Management Knowledge age has emerged in pace with information technology that innovated information age a few decades ago. The rapid transition is unprecedented in the history of technology

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applications. Thus logical speculation is: what is next to the knowledge age? One incident reported in the Indian may through some light to it. The great Akbar once asked his naba- ratnas: “what moves fast?” When eights of nine ratnas pointed towards Royal Horse, the ninth ratna, Birbal got an edge over others by saying “Our Mind, Sir.” We at least find a technology area where the trend is to achieve something like speed of mind, and this is nothing but communication. From the trend of communication we have no hesitation (and I am sure all will agree to it) to conclude that it is the speed of communication that is growing leaps and bound. We have seen the age of kilobits per second, and mega bits per second, and presently in the age of gigabits per second, and are seeing a tomorrow of tera bits per second. This is an indication that after knowledge age, the next age may be the age of mind or the age of conscious. The universe is made of non-living and living things. Their comparison in terms of level of intelligence, conscious and communication power is made in table (28). S Ranade, a great admire of Aurobinda told [65]: “Knowledge by identity will change current science completely. Particularly physics and biology will see radical changes. The wave-particle duality and the mass-energy equivalence will be seen in the light of the more basic substance of consciousness” and then he defined [65]: “consciousness is awareness, awareness of yourself and of others. In the human being both exist. In the animal, there is only awareness of others, not awareness of itself, it is a more limited awareness. In plants the awareness is even less. In the crystal it is still less, but nevertheless it is there.” If the crystal is having awareness, it is surely possible that “the next century will be the century of consciousness” and “you can focus your body consciousness on a point outside the body.” Will the “Will power” or “Mind Power” of Iswar Patuli depicted by great Bengali Novelist Sarat Chandra prevail upon the society, organization, culture and economy at the fragile end of knowledge age? Mother’s in the historical declaration[66] made on April’24 1956 said: “The manifestation of the supramental upon earth is no more a promise but a living fact, a reality. It is at work here, and one day will come when the most blind, the most unconscious and even the most unwilling shall be obliged to recognize it.” Perhaps that will be in the age of consciousness that is next to knowledge age. The collaborative views on this prediction is one important research found in [68]. Table 28: Comparison of different entities in universe in terms of sense and communication Non-living things Living things

Apparently no sense and no communication. Dr Ranade sees otherwise Plants

Limited sense and no communication

Animals

Low level sense and communication

Human beings

High level sense and communication

10. AGE OF DIGITAL DIVIDE Tagore once told “we have only one country in this universe, and that is world”. Rabindranath Tagore’s such a powerful philosophy may ultimately be realized if to-day’s tenet of “one world one village is implemented in true sense in future. To achieving this, a trend has already been initiated the world over. Privatization, Liberalization and Globalization are replacing liberty, fraternity and equality all over the world including the countries of third world. It does not mean that library and fraternity have no relevance in to-day’s society. They are ever alive and

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their universal appeal shall ever remain for the noble human society, but to day they are not all in all. Privatization and universalization shall be the other social partners with them. This is a wave brought forward by different emerging technologies, which are often interactive, interdependent and diffusive. Information technology, computer, communication, microelectronics, Genetic engineering, Biotechnology, Space technology are a few to name worthy. Developing world in general is far lagging behind the modern technological evolutions and revolutions. Besides the developing countries are hardly having capital to deal with such fast, rapid and perpetual changes. Developing world in general is labor intensive rather than capital intensive. Therefore, debate on the ability, suitability and the acceptability of liberalization is going on and will continue to go on for some more time in the developing countries. Initial mismatch and inertia are parts of life and the fact is that the society never denies mobility. The society ultimately accepts technological changes, which might be off-touch to the society even a few years back. And irony is that delayed such acceptance is done in quite haphazard and irregular ways. What has happened to the deployment of computer in government sectors in India today is anybody’s guess. This is a lesson that the third world always forgets. Consequently the third world continues to lag behind International trend, and losses money as, there is hardly any planning for technological up gradation and applications. We can sight a figure to justify this point. Telecommunications lines of India are 66% digitized; where as figures of Brazil and Hungary are respectively 35.7% and 41%. But the faults’ figures are 218 faults per 100 lines in India and 2 faults per 100 lines in USA and Japan. In Table (29), the percentage share of information technology for America, Europe and Asia, and that of the e-commerce buyers are shown. It is noticed that in both terms, the position of Asia is very poor. Table 29: % share of IT and E-commerce buyers % share of information technology in 1995

% share of E-commerce buyers in 1998

America

45.5

72.57

Europe

30.9

22.8

Asia and pacific

23.7

4.6

Better is not the sole dimension of competitive advantages; faster is equally another important dimension. Thus it will be a sound strategy for the developing country to take part in the globalization with out any further loss of time, but with intelligent, selective, judicious and strategic applications of globalization process, uses of and innovation with few technologies. Analyzing the problems of Third world in depth Dr. Colombo observed “The ability of developing countries to derive all the benefits of the new technologies faces one stumbling block right from the start. Although rapidly and seemingly effortlessly permeating the economic and production systems of the world, these technologies are not available “off the peg”. They have to be absorbed, metabolized, mastered and controlled. Their application calls for a pre-existing capability to insert new ideas, new practices, and new elements into a flexible system. This does not simply exist in the vast majority of the developing countries. Furthermore, it is essential that as the new technologies are introduced into the socio-economic fabric of the third world, they do not impair or destroy existing local cultures—we must equally concern ourselves with safeguarding the richness of the world cultures, mankind’s “cultural genoma”. Despite these problems it is strongly believed that the intelligent application of the new technologies in the developing countries can indeed speed up process of economic growth”.

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10.1 Gap Studies In history of social studies one important component of research deals with the findings reason and cause of growing gap between rich and poor; and for that purpose to suggest measures and steps to reduce the gap. But the fact remains that the gap has not been reduced even after thousands of such studies and the implementation of their recommendations those including those of some noble laureates. A few research findings report[67] 1. If the present growth trends in world population, industrialization, pollution, food production, and resource depletion continue unchanged, the limits to growth on this planet will be reached sometime within the next 100 years. The most probable result will be a sudden and uncontrollable decline in both population and industrial capacity. 2. It is possible to alter these growth trends and to establish a condition of ecological and economic stability that is sustainable far into the future. The state of global equilibrium could be designed so that the basic material needs of each person on earth are satisfied and each person has an equal opportunity to realize his or her individual human potential. 3. If the world’s people decide to strive for this second outcome rather than the first, the sooner they begin working to attain it, the greater will be their chances of success. To us those conclusions spelled put not doom but challenge - how to bring about a society that is materially sufficient, socially equitable, and ecologically sustainable, and one that is more satisfying in human terms than the growth-obsessed society of today.” Whatever gab and whatever challenge is to be met revolves around three factors : i) economic and social gap, ii) education gap and iii) status gap between agriculture and industry.

10.2 Problems of Agriculture Sector The existing economic and social gap between rich and poor is primarily due to two avalanche affects: (a) In agriculture sector, negative avalanches is “produce and perish” and (b) In business sector, positive avalanche is “produce and flourish”. The only solution to bring the balance is that the prices of agriculture produce must be raised at those of business produces by strict control of governments. Education is an investment not only in terms of money but also in terms of time and human resources. Parents have noticed that the boys/ girls after getting school level education become useless/worthless/resource less rather than resourceful in terms of earnings in the family. They neither get job nor by that time skillful for laborious jobs including agricultural jobs. Had these boys not been sent to schools rather been engaged from the childhood in agriculture related sectors; they would be more useful for earnings for the family. This clearly demonstrate that the education till not is sure with guaranteed minimum income to family, the poor family does not like to take risk of spending mainly time and money in education. Mac Bridge Commission report that the farmer and the agriculture producers must have the direct market knowledge to get actual price of the produces. This is believed to be possible only with IT.

10.3 Case of Industries The state of West Bengal in India has achieved a considerable amount of rural economic growth in the last two decades. The average income of the rural people has increased and the social

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security of rural people has been established on the solid footings. The disparity in income among the rural people has decreased considerably. An all around development of rural people and society has been noticed. However this development is due to land reforms and “barga” system sincerely implemented by the Left-front government of W. B. in their 25 years of rule. By the process of land reforms and barga system, the agricultural workers or farmers are given confidence that they will never be thrown out of work and land they do cultivate. This confidence has led to generate among farmers the more sense of belongingness and sincerity in their work. This has reduced the victimization and the injustice meted out to them in terms of payment or no payment earlier by the Land-Lords; which in other ways has caused the agricultural productivity to increase and loss of agricultural working days to decrease as well as the agricultural disputes between labor and owner to lessen. The barga solution is our own and is not something copied from the developed nations. The economic and productivity failures in all sectors namely agricultural, industrial and banking is mainly due to disputes between labors and owners. Thus if such disputes in agriculture sectors are overcome by the barga system; it is logically extensible for other sectors like industrial and banking too. In this paper we propose an “industrial barga” system for Indian Industries. We have achieved something unique by our own system of bargas in agriculture sector. Similarly the industrial barga not prevailing elsewhere dose not mean it is inappropriate in India. In Indian environment where economic disparity is huge and where labor is cheap and for which victimization of labor is easy; the industrial barga will be the right solution. The proposed industrial barga aims to provide share of production and profit of industries with labor, management and owner as in agricultural barga. There may be several means of implementation. The Industrial barga will not be easy to implement. With IT age, the difference gap is easily to meet with. What is need of the hour is the strategy and goodwill for the application in right perspectives.

11. CONCLUSIONS The goals of both the near and the far futures of IT is Fig. (22). In the field of computer, the major challenge of the 21st century will be the designing of bio/brainy computer. The basic science has been searching, since the days of its journey, the design if any behind the universe as well the theory of birth of the universe; and possibly a new “Theory of Everything” as Prof Hawking’s prediction made in 1980 of achieving his famous “Theory of Everything” by the end 20th century is proved wrong. The debate on deterministic vs. probabilistic nature of universe or whether the nature is a machine or not, is oscillating. In such a scenario the debate on possibility of designing brainy computer only be a logical extrapolation; and definitely will take long time to answer. On the other hand, future all wireless, anywhere and any time communication is relatively non-debatable issue and expected to be achieved, although not without overcoming many obstacles. Even small deployment like IEEE 802.11 based WLAN faces many obstacles[69]. Other than systems and standards, two inherent problems of future communication need to be properly addressed: higher error probability of all wireless links and information security. Whereas the error control is basically a technical issue, the security of information has several dimensions. The requirement of security for a durable application of IT, namely e commerce and e business was illustrated earlier. It is reported[70] that “The increasing frequency of malicious computer attacks on government agencies and Internet business has caused severe economic waste and unique social threats.” As per the second law of thermodynamics the open systems cannot bring order without making its surroundings disorder.

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The security measures that are for bringing order to information process, inevitably brings disorder to its surroundings that may again itself be the source of hackers or security breakers. This is a manifestation of chaos and complexity. Of course the men create problems only to solve it afterwards. Is the nature likes to see man dancing between problems and solutions? Are we then leading us to a state of chaos and complexity[71]? This is compounded by the fact that “computer system and network security is increasingly limited by the quality and security of the software running on constituent machines. Researchers estimate that more than half of all vulnerabilities are from buffer overruns, an embarrassingly elementary class of bugs”[72,73]. The steps to go out of chaos and complexity will be the major challenge for investigation in 21st century. High Speed Computing, Autonomous Computing Optical Computing, Quantum Computing Chemical/Bio/Intelligent Computing Seamless Power + Intelligence

In Neat Future Information Age to Knowledge Age Knowledge Society, Knowledge Factory, Knowledge Workers, Knowledge as Wealth

In Far Future Age of conscio usness

3G Mobile to 4G Mobile, Cellular, GSM, PDC, PHS, Paging, UMTS, FSO, xDSL, Next Generation IP and VoIP, Wireless Ethernet-IEEE 802.11, Wireless Home Networking IEEE 802.15.4, Wireless Internet, LEO, Multimedia Standard, Wireless ATM, PCN Seamless Mobility, Coverage, and total Integration

Fig. 22: IT in 21st century

Entering into the knowledge age is the inevitable consequence of the application of networks in the business, organization, government, society and economy. The entry needs to break several hurdles. The issue of the acceptability of knowledge economy with non-material wealth, knowledge along with the new status of human resources as knowledge workers, and the concept of sharing knowledge for organizational benefits are a few areas to be addressed. The quantification of the knowledge and the exchange rules of knowledge for the purpose of sale and business of and with knowledge are the technical challenges and need serious

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investigation in this century. The consciousness as Penrose told “is the phenomenon whereby the Universe’s very existence is made known.” Thus in the age of consciousness, the man’s desire to be the master of nature with which this paper started, may be realized. Will it be really! The constructive and judicious application of IT may lead to overcoming the consequences of “digital divide.” Several studies[74,75] have suggested for the application of IT in Education & Training, Telemedicine and Diagnosis, E-Government, Rural Information Sharing for the purpose of food conservation & sale, and Entertainment among others for deriving maximum benefits in the developing countries. Like digital divide, another negative application of IT is like what happened on 11th September in USA. Analyzing the 11th September issue, a famous research work[76] has reported to examine the issue for developing a system dynamics for positive application of technology. This is a new direction of research in application of technology. The same direction may be extended to remove digital divide.

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Fernando J. Velez et al., “Mobile Braodband Services….”, IEEE Communication Magazine, Vol. 40, No. 4, April 2002, pp. 142-150. William Webb, “Broadband Fixed Wireless Access as a Key Component of the Future Integrated Communications Environment”, IEEE Communication Magazine, Vol. 39, No. 9, Sept. 2001, pp. 115-121. Shyam S. Chakraborty et al., “An Adaptive ARQ Scheme with Packet Combining for Time Varying Channels”, IEEE Comm Letters, Vol. 3, No. 2, Feb. 1999, pp. 52-54. Shyam S. Chakraborty et al., “An ARQ Scheme with Packet Combining,” IEEE Comm Lettters, Vol. 2, No. 7, July 95, pp. 200-202. C.T. Bhunia, “ARQ Techniques: Review and Modifications,” Journal IETE Technical Review, Vol. 18, No. 5, Sept.-Oct. 2001, pp. 381-401. C.T. Bhunia, “A Few Modified ARQ Techniques,” Proceedings of the International Conference on Communications, Computers & Devices, ICCCD-2000, 14-16, Decedmber 2000, I I T, Kharagpur, India, Vol. II, pp. 705-708. Hossein Izadpanah, “A Millimeter Wave Broadband Wireless Access Technology Demonstrator for the Next Generation Internet Network Reach Extension”, IEEE Communication Magazine, Vol. 39, No. 9, Sept. 2001, pp. 140-145. Luis Munoz et al., “Optimizing Internet Flows over IEEE 802.11b Wireless Local Area Networks……..” , IEEE Communication Magazine, Vol. 39, No. 12, Dec. 2001, pp. 60-66. Vipul Gupta and Sumit Gupta, “Securing the Wireless Internet”, IEEE Communication Magazine, Vol. 39, No. 12, Dec. 2001, pp. 68-73. Jeyhan Karaogue, “High Rate Wireless Personal Area Networks”, IEEE Communication Magazine, Vol. 39, No. 12, Dec. 2001, pp. 96-102. Geng Sheng Kuo et al., “ Dynamic RSVP protocol”, IEEE Communication Magazine, Vol. 41, No. 5, May 2003, pp. 130-135. Shidong Zhou et al., “Distributed Wireless communication System….”, IEEE Communication Magazine, Vol. 41, No. 3, March 2003, pp. 108-113. Yungsoo Kim et al., “ Beyond 3G: Vision, Requirements,and Enabling Technologies”, IEEE Communication Magazine, Vol. 41, No. 3, March 2003, pp. 120-123. Alain J Godbout “Information Vs Knowledge”, Robert Taylor, “Knowledge Management”, Robert m [email protected] <mailto:[email protected]> S. DiMattia et al., “Hope or Hype”, Managing Knowledge, Macmillian Business, UK, 2002. Yogesh Malhotra, “ Knowledge in inquring organizations”, Proc. 3rd Americas conference on information systems, August 1997. S. Ranade, “The Technology of Consciousness” Dipti Publications, Sri Aurobindo Ashram, Pondichery, 2000. Sisir Kumar Mitra, “Sri aurobinda”, Orient Paperbacks, 1976. R Sadananda, “The Limits to Growth-A Revisit, Knowledge Networks and Sustainable Development”, Proc 37th National Convention of CSI 2002, Tata McGrawHill, 2002, pp. 23-31. Sushil Mukhopadhyaya, “Whither Bio-Science?”, J IETE Tech Review, Vol. 19, No. 6, Nov-Dec. 2002, pp. 381-386. Upkar Varshney, “The status and Future of 802.11 based WLANs”, IEEE Computer, Vol. 1, No. 3, June 2003, pp. 102-104. Hassan Aljifri, “IP Traceback: A New Denial Of Service Deterrent”, IEEE Computer, Vol. 1, No. 3, June 2003, pp. 24-31. C.T. Bhunia, “Cryptography: From Classical to Quantum Age”, IT Seminar, Dept of ETC, BEC (Deemed University), Shibpur, 2001. Nancy R Mead et al., “From the Ground Up….”, IEEE Computer, Vol. 1, No. 2, March 2003, 59-63.

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D. Wagner et al., “A first step towards automated detection of buffer over run vulnerabilities’, Proc 7th Network and Distributed System Security, 2000. Michael Gurstein, “Rural development and food security…..”, SD Dimensions, FAO, November 2000. A.K. Roy, “The Dawn of an information age…….” Thought, Vol V, Issue IV, April 2001, pp. 4-7. Erica Vonderheid, “Answering a Wake Up Call”, IEEE , The Institute, June 2003, pp. 1 & 12. Arun N. Netravali, When Networking becomes……and Beyond”, IETE Technical Review, Vol. 19, No. 6, Nov-Dec. 2002, pp. 353-362. P.C. Mabon, Mission Communications—The Story of Bell Laboratories, Bell Telephone Laboratories, Inc, Murray Hill, N J, 1975, p. iv. Lester C Thurow, The Wealth of Knowledge, Harper Collins Publishers, USA, 2002. R. McGinn , A Revolution in Networking: Toward a Network of Networks, Network + Interop, Atlanta, Georgia, Oct. 21, 1998.

74. 75. 76. 77. 78. 79. 80.

APPENDIX-A Edholm’s Law The following table depicts the growth of data rates under different communication/network technologies. The data rate follows Edholm’s law that states the data rates for all three communications, namely wired, nomadic and wireless are as predictable as Moore’s law. The rates are increasing exponentially and the slower rates trail the faster rates within a predictable time gap. Table: Date rate growth of different Communication/Network Technologies Year

Wired Technology/ Standard

19751984

19851994

19952004

Nomadic Data rate

Technology/ Standard

Wireless Data rate

Technology/ Standard

Data rate

Ethernet

2.94 Mbps

Hayes Modem

110 bps

Wide Area paging

A few hundreds bps

Ethernet

10 Mbps

Modem

9800 bps

Alphanumeric paging

A few Kbps

Ethernet

100 Mbps

Modem

28.8 Kbps

Cellular/GSM

≈ 50 Kbps

Modem

56.6 Kbps

IEEE 802.11 b

11 Mbps

IEEE 802.11 g

108 Mbps

PCN/UMTS

> 2 Mbps

B3G (Beyond 3 G)

12 Mbps

MIMO

200 Mbps

Ethernet

1 Gbps

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HISTORY BEHIND COMPUTER NETWORK

The convergence between two important technologies, namely computer and communication gave the birth of Computer Communication and Network. A useful study of convergence of telecommunication and computing is literature [1]. Computer network means a network of geographically distributed many autonomous system connected [2,3] in such a mode that meaningful transmission and exchange of information become possible among them. Resource sharing and load sharing are the major two objectives of a network. To meet the objectives the networks were evolved over times. In early 1950s, peripheral devices and remote job entry points were connected to the central computer through communication links (Fig. 1). This is to use the resources optimally and economically, as then computers were very high cost machines. By sharing of resources, the cost was distributed over number of users. In 1960s the number of peripherals devices expanded rapidly, the computer power also increased by several folds and time shared computer was evolved. This made the use of separate long haul /distance communication link to each peripheral device. But the solution was most uneconomical and technically unsound. Fig. 2 illustrates the falling cost of communication and computer, but the rate of fall was more in case of computer than in the case of communication links. Moreover the design of a high bit data link is more economic than several low data rate links. For Example the cost of a 1.5 Mbps link is about six times that of a 64 Kbps link but the data rate is 24 times higher. The trade off is low cost per unit data versus high cost per unit data. This leads to the use of remote multiplexers or concentrators to connect a number of remote terminals in the same area and to use a single shared communication link to connect with central computer (Fig. 3). As the number of devices increased, the task of communication became huge. To free central computer from communication task, special processors called Front End Processors were used. The communication is automated in such system but the control of communication still remained central to the central computer. In 1970s, with advent of PCs—low cost, portable but not less capable computer, the problem takes a U turn in several aspects: automated system at remote locations, distributed control etc. This leads to the concept of connecting several geographically distributed computers—some of them doing the computational jobs and others communicational jobs

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(Fig. 4). General purpose network concept then evolves as Communication Sub network + Users’ Sub network. In 1980s and thereafter, several networks of different types were connected through bridges and gateways to develop an interconnected network (Fig. 5). Interconnecting network of different networks is known as internets (note the lower case of “i”). The special such an unique internet is called Internet (note the upper case “I” in Internet). Printer

T

T

Central processor

Terminal T

T

T R. I. F

Falling cost

Fig. 1: One central processor and a separate communication link to each device like terminal, remote job entry (RJE) points, pointer etc.

Communication cost

Computer cost

Year

Fig. 2: Falling cost of Computing and Communication with year (Cross over year 1970)

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T Printer T

T

T

T Multiplexer

Central processor

Front end processor

Printer T T

T

Terminal T controller T

T Multiplexer T

Fig. 3: One central processor but with shared communication links to devices like Terminal, Font End Processor, Terminal Controller and Multiplexer.

(Issue: Front End Processor performs the job of communication, while central processor remains in processing job)

Personal computer Terminal

Node

Node

Communication subnet

CPU

CPU

Fig. 4: General network with two subnets: Communication Subnet and Users’ Subnet.

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BOX 1 How do we see a Computer Node in the Netowork in Terms of Message in and out ?

Model of Nodes The intermediate computer node or terminal may be seen in the different models of queuing. A model of this kind has four things for specifying: the message arrival probability density, the message departure or service probability density, the number of the servers in the node (basically this is the number of the output links or the number of the computers/machines providing the service) and the buffer space in the node. For the purpose of simplicity and analysis in the computer networks, it is often assumed that the nodes are of having infinite buffer space. However such an assumption works fine as usually the analysis is of comparative nature. The different statistics are used to define the arrival and the service probability density functions and these may be: exponential probability density function represented by the symbol, M (Markov); deterministic nature represented by D or general arbitrarily type represented by G. A model of type M/M/1 means it is a node where: The message arrives at the node at the Markov Process or in the exponential probability The message departs the node at the Markov Process or in the exponential probability The number of the server is one The exponential probability density function is most suitable for a large number of independent customers. The large number of independent customers actually comprises a network. Thus it has been observed that the exponential arrival and the exponential service process best suit the computer network. The M/M/1 model is one of the appropriate models for a node. The Markov process is governed by the Poisson Law that states that the probability, P k(t) of arrival of exactly k messages in an interval of time, t is given by: Pk(t) = [(λt)k/k!] e–kt ...(1) where λ = average arrival rate of messages. We define the state of the node by the numbers of the message staying in the buffer (queue) and in the server of the node (Fig. 1). Pk defines the probability that the node is having k messages. Queue/Buffer Message arrives, 

Server M

M

M Message departs, 

Fig. 1: M/M/1 model

We define m as the average message service rate of the node. If we define C as the capacity of the server’s output link in bps (bits per second), and m as the average size of the message in bits, we have: µ = C/m ...(2) Based on our defined state of the node, the state diagram of the M/M/1 node will look like as in Fig. 2.

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NETWORK AND INTERNET TECHNOLOGY  P0

P0

 P1

P1

 P1

Pk – 1

103

P2

 P2

 Pk + 1

 Pk

Pk

 Pk

Pk + 1

 Pk + 1

Fig. 2: State diagram of the states/State transition probability.

The node is in equilibrium when from Fig. (2) we have: λP0 = µP1 λP1 = µP2 λP2 = µP3 ………...…. λPk = µPk+1 ………...…. we then have: Pk = (λ/µ)k P0 K=∝

∑ P = ∑ (λ/µ)

and

k

k

...(3)

P0 = 1

K=0

or

P0 = 1/(1 – (λ/µ)) = 1/(1 – ρ) where ρ = λ/µ [ as Σ

(λ/µ)k

= 1/( 1 – (λ/µ)) or

So,

Σ ρk = 1/(1– ρ)

Pk = (1 – ρ)

...(4)]

ρk

If N is the average number messages in the system, we know: K=∝

N=

∑ K.P

k

= (1 – ρ) Σ k . ρk

...(5)

K=0

Differentiating both sides of equation (4) with respect to ρ and then multiplying both sides by ρ, we find that Σk .ρk = ρ/(1 – ρ)2, using which in equation (5), we get: N = ρ/(1 – ρ)

...(6)

If it is assumed that the average time all the messages in the node stay is T, they by famous Little’s formula: N = λT

...(7)

Using equations (6) and (7), we get : T = 1/(µ – λ) Equations (6), (7) and (8) are used to qualitatively analyse the nodes of networks.

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...(8)

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QUESTIONS 1. ρ is called load intensity. Why ? What is its physical significance ? From eqn. (8), it is seen that for a physical realizable and stable system, µ should be greater than λ i.e. ρ should be less than 1 (Fig. 3). Actually if µ is not greater than λ, the number of the data in the system and its corresponding delay will continue to grow unbounded. Then natural question be asked if µ is greater than λ, where is the question of delay? Actually the parameters, λ and µ are statistically average. At some instant the arrival rate may be higher than the service rate when the data faces delay. =1



T delay

Fig. 3

2. In the network environment two nodes are as in Fig. (4). The nodes are independent to each other. Prove that the probability that the node N1 is having n1 messages and the node N2 is having n2 messages is given by: n P (n1 in N1 and n2 in N2 = ρ1 1 (1– ρ1) ρ2 n2 (1 – ρ2)

N1

1 l1 = total average arrival rate from other nodes/internal/external at node N1

N2

2

l2 = total average arrival rate from other nodes/internal/external at node N2

ρ1 = λ1/µ1 ρ2 = λ2/µ2 Fig. 4: Figure of the given question

SOLVED PROBLEM 1. At a node, packets arrive at an average rate of 120 per hour. The output service link capacity is 8 characters/sec. Find (a) average number of the packets in the system at steady state condition, (b) average delay per packet, (c) average waiting time in queue, (d) average number of packets in queue.

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Assume (i) M/M/1 model (ii) one packet = 144 characters. Solution. λ = 120 pkt/hr. = (120/3600) pkt/sec = 0.033 µ = (8/144) pkt/sec. = 0.055 δ λ = (a) N= = 1.46 pkts 1− δ µ − λ (b) T = 1/(µ – λ) = 44.33 sec. (c) W = T – 1/µ = 26.14/sec( W = δ/µ – λ) (d) NQ = λ . W = 0.86 packets. (NQ = δ2/1 – δ) 2. For the problem 1., what is the probability that there is 2 or less packets in the system under steady state condition? Solution. The probability is

2

∑ (1 − δ) δ

k

K =0 2

= 0.4

∑ (0.6)

k

[δ = λ/µ = 0.6]

K=0

= 0.4 + 0.4 x 0.6 + 0.4 x 0.36 = 0.784. BOX 2 How do the Nodes and/or the Computers Communicate over Share Line ? (This is Basicaalu an Issue of Users’ Subnet Communication)

Multidrop/Multipoint Link Control-Polling When a transmission path between two terminals or nodes or connectors is dedicated, the link is termed as point to point. When many terminals or nodes or connectors share a transmission link, the link becomes shared. For a shared link, if transmission between any two users is allowed at any time, the link is called Multidrop / Multipoint. Link. In order to regularize the transmission of a Multidrop / Multipoint Link, a control is required. The main goal of such regulation is to increase the link utilization of the link. Regularized control prevents the interference caused by simultaneous transmission by more than two stations. If more than two stations try to communicate simultaneously, the interference occurs making the communication the garbage. Thus a control is required to avoid the unproductive communication. In Multidrop/point Link, often one node or terminal or connector acts as a primary station. The role of primary station is to monitor the control of link regularization. The other station or nodes or connectors in the link are known as secondary stations. In a Multidrop/point link, communication between primary and any one secondary is allowed at any given point of time. However, the communications between secondary stations are only done though primary. This is in sharp contrast to the other kind of Multidrop/point communication technique used in networks like LANs (Local Area Networks). In terms of accessing methods of shared link, different techniques may be classified as in Fig. (1). Polling is a technique used in Multidrop/ point link control. Two main categories of polling are serial polling and hub polling. A variation of serial polling is known as selective serial polling. In all polling techniques, the stations are assigned some addresses that are known to all stations in the link.

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Access methods

Non-contention techniques (Deterministic)

Centralized (Hence primary station is responsible for control Example : Poling

Contention-technique (Probabilities/Stochastic) Example: CSMA/CD used in LAN

Distributed ( All stations equally take part in control) Example : Token passing used in LAN

Fig. 1

In serial polling (also known as roll-call polling), the primary station sends a poll to each secondary station one at a time and one often another. For Fig. (2), the primary station sends first a poll message to S1. The poll message contains the address of polled station as a header. The secondary stations check the header address, and thereby identifies whether the poll message is meant for the particular secondary or not. However, S1 getting the poll message accordingly identifies the poll. If the poll is meant for it, it will send a positive response followed by data if it has data to send. Along with the message/data, the secondary sends its address also. If it has no data, S1 will send a negative response, which makes a call nonproductive. After the call of S1 and after receiving its response, the primary will poll S2, then S3 and so on until Sn. After that the cycle will repeat.

S1

Primary

Sn

Fig. 2

However, the major disadvantage of roll-call polling is the waste of time due to nonproductive calls. This problem is uniquely tackled in hub polling. In hub polling, the primary initials the polling. It initiates polling by sending a poll message to S1. S1 sends data if it has to send. Other wise, the secondary S1 transfers the poll to S2 and so on. Primary identifies the senders by the appended header address of secondary in the data message. The last secondary Sn returns the poll to primary. There is other way also to implement hub polling. For example, the primary may initiate poll by polling first Sn. Sn after its response, sends poll to Sn – 1 and soon. A poll is terminated when S1 at last transfers the poll to primary. The processing job in secondary stations for hub polling is more than that for serial polling.

Sn

Si

S2

S1

Polling

Fig. 3

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1

S

2

S

;

S

m

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Analytically serial polling and hub polling may be compared. For such comparison, we shall use a parameter known as poll-scan time (PST). PST is the time taken for just polling (and not for sending data) all the secondary stations once in a cycle. For the purpose of analysis a generalized shared link system of Fig. (3) is to be considered. We assume: tp = Processing time required in primary to initiate a poll. We assume it is same for serial polling and hub polling. ts = processing time required in a secondary to process the poll message and to process the response for serial polling. th = Same as ts but for hub polling. tli = Time required both for propagation and transmit for a poll message to be transferred from primary to secondary, Si i tl = Same at tli but for a poll message to be transferred from primary to secondary Si (i = 1 to n, j = 1 to m) We also assume tli and tli for all possible is and js are same for serial and hub polling. Under these assumptions, we can calculate PST as below: For serial polling For secondary S1, poll scan time will be tp + ts + 2tl1. For S2, the same will be tp + ts + 2tl2, and so on for other secondaries. Thus (PST)s =(n + m) (tp + ts) + 2(tl1 + tl2 + …. + tln) + 2 (t11 + tl2 + …. + tlm) ...(1) For hub polling On similar grounds (PST)n = tp + (n + m) tn + 2(tln + tlm) ...(2) We further assume that all stations are equidistant apart in which case tli = tlj = t (constant). Then (PST)s = (n + m) (tp + ts) + 2t (1 + 2 +…..n) + 2t(1 + 2 +….m) as under the assumption of equidistant we have tl2 = tl2 = 2t tl3 = tl3 = 3t ………....... tln = nt and tlm = mt Then, (PST)s = (n + m) (tp + ts) + n (n + 1) t + m (m + 1) t or

(PST)s = n [(tp + ts) + (n+1)t] + m [(tp + ts) + (m + 1)t] Under the same assumption of equidistant nodes, we have (PST)h = tp + (n+m)th + 2(n + m) = tp + n(th + 2t) + m(th + 2t)

...(3)

…(4)

We can make several observations on equations (3) and (4) : 1. Condition that hub polling should be superior to serial polling is that : (PST)h < (PST)s or

tp + n [(th + 2t) + m(th + 2t)] < [n(tp + ts) + (n + 1)t + m(tp + ts) + (m + 1)t]

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th < [{(m + n – 1) tp + (m + n)ts + t (m2 + n2 – m – n)}/(m + n)] ...(5)] 2. If we assume m = 0, then system of Fig. (3) becomes that of Fig. (2) for which: (PST)s = n[(tp + ts) + (n + 1)]t (eqn. (3) with m = 0) (PST)h = tp + n(th + 2t) (eqn. (4) with m = 0) and condition for hub polling being superior will be : th < [{(n – 1)tp + nts + t (n – 1) n}/n] (Inequality (5) with m = 0) or tn < {(1 – 1/n) tp + ts + (n – 1) t } ...(6) or tn < ( tp + ts + nt) when n>>1. 3. If m = 0, n = 1, it is predicted that (PST)s should be equal to (PST)h. This can be verified from eqns. (3) and (4) provided ts = tn. For inequality (6), it is then seen that, tn < ts which is not the case as we known tn > ts. This suggests that hub polling becomes superior to serial polling as number of secondary increases. Selective serial polling is a compromise or hybrid technique. Serial polling is done by roll-call technique where under a poll cycle each secondary is polled once only. But in the selective serial polling, a particular secondary may be polled more than once, while other particular secondary may be called just once. In such technique, a pre-defined sequence table is maintained based on statistical behavior of data of the secondary stations. This reduces the delay due to non-productive call for a cycle of polling. Addressing in polling technique is done by standard protocol. One such standard protocol developed by IBM is known as BISYNC (Binary synchronous Communication). BISYNC message format is : or

SYN

SYN SOH

Where :

Header STX

Data

ETB/ETX Error

Check

SYN = synchronization Word SOH = Start of Header Word STX = Start of Text Word ETB = End of Transmission Block ETX = End of Text. For error check vertical parity is used if ASCII code is in use CRC is used for EBCDIC. However, ETB is used to terminate a block when there will be more transmission of block, whereas ETX is used to terminate the transmission of full text i.e. to indicate the termination of last block. Solved Problems. Compare serial polling with hub polling for a scheme of Fig. (3) with m = 0. Use the parameter of duration that a secondary waits before polling, for such comparison. Find the average range of such delay. Assume secondaries are having no data. For serial polling we have from eqn. (3) (PST)s = n (tp + ts) + n(n + 1)t Thus the average time to poll a secondary, ta = (PST)s/n = (tp + ts) + n(n + 1)t At any instant, a particular secondary may have to wait for polling of (n – 1) stations before a poll is offered to it or may not have to wait to get poll. As the secondaries are having no data to send, therefore the average waiting time before getting a poll is : average waiting = [{(n – 1) + 0}/2} ta

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We can find the second moment of the waiting time which is by definition is the weighted average of the squares of possible polling time. Hence, second moment = 1/n

( n − 1)

∑ ( pt

a)

2

p=0 2

ta (n – 1) (2n – 1) 6

=

n

as we known

=

∑

p2 = n(n + 1) (2n + 1)

p=0

Therefore, variance by definition is: Var = Second moment – (average)2 =

FG H

ta 2 n−1 (n – 1) (2n – 1) – . ta 6 2

n2 − 1 . ta2. 12 By definition average range is given below:

IJ K

2

=

Average ± 2 var Hence average range of delay before a secondary is polled is

n−1 ta ± 2 or, we can say :

upper limit =

and lower limit

n2 − 1 2 . ta 12 n2 − 1 3

1 n−1 t + ta 2 a 2

F GG H

n2 − 1 3

=

ta n − 1+ 2

=

n − 1 ta n − 1− 2 2

F GG H

I JJ K

n2 − 1 3

I JJ K

However if n = 1, we have Upper limit = lower limit = 0 Which is physically justified. For hub polling from equ (4), we have ta = Hence we can write 1. For serial polling

(PST) h t p + th +2t = n n

upper limit =

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FG t H

p

+ ts + (n + 1) t 2

IJ FG n − 1 + K GH

n2 − 1 3

I JJ K

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lower limit = 2. for hub polling

FG t H

p

+ ts + (n + 1) t 2

FG t /n + t H 2 F t /n + t =G H 2

upper limit =

p

n

lower limit

p

n

IJ FG n − 1 − K GH

IJ FG n − 1 + K GH = 2t I F JK GGH n − 1 + = 2t

n2 − 1 3

I JJ K

I JJ K n − 1I J 3 JK n2 − 1 3 2

Multidrop/Multipoint Terminal Control After the discovery of integrated circuit technology, in general the cost of communication system exceeds the cost of communication line. In order to reduce the total cost of communication, therefore a mean of communicating is provided by which a single transmission line is shared by many terminals or users for purpose of communications. Two broad classes of terminal controls are shown in Fig. (4). Terminal controller may be again broadly of two types: multiplexers and concentrators. Concentrator is also known as asynchronous time division multiplexers or statistical time division multiplexers. In multiplexer each input line is provided with a predefined time slot. Input data are outputted hence according to the sequence. No addressing of input data is required. However, the output capacity will be sum of the input capacity of input. So far link capacities is concerned, multiplexer is not a good choice for terminal handling because of “BA” features of the data. As such it may so happen that most of the time for a small fraction time, some of input lines may have data to send; and hence time slot meant for them will be wasted.

Controller Controller

(a) Multidrop/Shared the communication

(b) Point-to-point lines/dedicated line communication

Fig. 4

In concentrator, the output line capacity is made less than the sum of input capacity; and the inputs are not sequentially allowed to send data at a predefined sequence as in multiplexer. Therefore data of inputs must have address for identification. For a concentrator if there are n inputs each with input data rate of r bps, then output link capacity, c in bps is given as: c < nr whereas for multiplexer c = nr As for concentrator c < nr, it may so happen that when all input terminals, try to send data, there will be loss of data. This will be tacked by buffer size at concentrator. However, it will be not a serious problem for data as data is insensitive to time.

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However, if a is the mean fraction of time each input terminal is transmitting, then 0
or

β<1

or

β ≥ α]

However if β < α, there will be a case when input flow will exceed output capacity. We can analysis controller in terms of M/M/I model (BOX 1). Hence λ = αnr µ=c or δ = λ/µ = αnr/c Therefore, average number of bits in queue at the concentrator is N = δ/1 – δ = αnr/c – αnr We can consider two cases: Case 1: n = 100 r = 500bps α = 0.5 c = 50000 bps. Case 2: n = 200 r = 500 bps a = 0.5 c = 10,0,000 bps In both the cases, α = 0.5 and n = 1. If we can n as buffer size estimate, we see that as input lines increases, a small amount of buffer space per input is required. Average delay will also be small. This is a thing to be remembered while designing a system. The major problem of concentrator design is always the trade-off between output capacity and buffer size. In terms of delay, multiplexer and concentrator may also be compared. We assume M/ M/1 model for the purpose of analysis. For multiplexer (both TDM and FDM) for each time λ = r bps µ = c/n bps Whereas for concentrator: λ = nr bps (max.) (This is assumed for comparison) µ = c bps

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Thus, average delay for each line (Process) of multiplexer is n 1 1 = = T= µ − λ c/n − r c − nr Whereas the same for shared process of concentrator is: 1 1 = T= µ − λ c − nr It is seen that average delay is ‘n’ times more for multiplexer. Actually it will more than ‘n’ times as in expression of T for concentrator, in place of ‘nr’ it will be actually ‘αnr’ and as we know 0 < α ≤ 1. Example of Design of a Data Multiplexer In practical situation, how a multiplexing system is designed with main multiplexer and sub-multiplexer in asynchronous mode of multiplexing, that is to be discussed in this section. For this purpose we shall take usual technique of telemetry (a typical data) TDM. This example is chosen because it deals with physical processes / signals / channels / terminals with wide variation of data rate. Suppose we need 5 data channels with sampling rates of 6000, 1400, 1000, 650 and 450 Hz. We can use a 5–channel synchronous multiplexer with signaling rate of 6000 × 5 = 30 KHz (without consideration of frame synchronization bits etc.). But it will not be an efficient decision in terms of cost and channel utilization (for lower bits channel, we are keeping unnecessarily higher bands). A more efficient decision one will be a scheme of Fig. (5). Main multiplexer 8. For synch 1. Data channel-1 (1500 × 4 = 6000 Hz)

1.1500 Hz – Sampling frequency

2. Data Channel-2 (1400 Hz) 3. 4. Data channel-3 (1000 Hz)

2.8 Channels TDM

5. 6. Data channel-4 and 5 (650 + 450 = 1100 Hz) 7.

12 KHz Sub-multiplexer 1.750 Hz sapling frequency 2.2 channels TDM

Channel-4 Channel-5

1500 Hz /8

Clock 1500 × 8 Hz = 12 KHz

(12 KHz/1500 Hz = 8)

Fig. 5

Data channel-1 is four inputs of main multiplexer to match its input rates. Similarly data channel – 4 and – 5 are sharing one input to match their lower data rate – this is being dine though use of a sub-multiplexer.

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Data Transmission over Voice Grade Lines The world is having a huge network of telephone. A telephone circuit (or voice circuit) provides a path for voice communication at voice frequency. For example, the basic two - wire telephone link (local loop) (a link between a subscriber and local exchange) usually bear a band with of 3 KHZ. However, data needs higher band width for transmission. For data, if separate network is proposed to be developed, it will be costly endeavor. Therefore, data transmission over voice-grade line is only a suitable proposition. It is now adopted for public uses by public link or by big organization by leased link. However, binary bits /data (1 or 0) is represented by computer as pulse. Pulses are not suitable for transmission over voice line. A modem (modulator + demodulator) is used to convert the digital pulses to analog signals suitable for transmission over voice line using frequency shift keying or OPSK (Quadrature Phase Shift Keying). Fig. (7) can be seen for multipoint or point-to-point configuration while Fig. (8) can be seen for a network. P1

Modem

Voice grade line

Modem

Terminal-1

Computer P2

Modem

Voice grade line

Modem

Terminal-2

P3

Modem

Voice grade line

Modem

Terminal-3

P1 = Port

(a) point-to-point

Computer

Port

Modem Modem

Modem

Modem

Terminal-1

Terminal-2

Terminal-3

(b) Multidrop/Multipoint Fig. 7 TDMers or FDMers (Reapers/ Amplitude/ Delay equalizers)

T Modem R Local loop

Customer premises

Control offiece

Repeaters regenerators echo suppleness Cable carrier radio carrier fibre carrier satellite carrier Carrier system

TDMers or FDMers (Reapers/ Amplitude/ Delay equalizers) Control offiece

R Mode T Local loop

(a) Data transmission circuit over telephone line

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INFORMATION TECHNOLOGY, NETWORK AND INTERNET Ter Mux

Node FEP

Computer

FEB = Front end process Node/FEP = a concentrator /a STDM/with memory and processing power for switching

High speed link of voice

Modem

Modem

Ter

DEM UX

Modem

Modem

Modem

Low speed voice line

Data link

Low speed line for voice

Modem

Concentrator

Modem

Ter

Ter

Ter

Ter

(b) Data Network using voice lines. Fig. 8

LAN Gateway

Personal computer

Wide-area network

LAN

Communication subnet Terminal

Local area network

CPU Wide-area network

Bridge

CPU LAN

Fig. 5: Interconnected network with individual network like WAN, LAN via bridge and getaway.

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COMPUTER NETWORK = Communication Subnetwork + Users’ Subnetwork Communication Subnetwork deals with Switching & Routing, Topology, Link & Transmission, Node Capacity Users’ Subnetwork deals with Topology and devices.

2. Objectives of Networking In literatures many objectives are often cited. But all such objectives are each a derivative of basic two objectives, namely RESOURCE Sharing and LOAD Sharing. Any resource, mainly costly one that is connected in the network, may be shared by all the users of the network. For example, a costly laser printer or a modem in a network may be shared by the network users. All computers, nodes, hosts and terminals that are part of a network can share resources like costly machines (Printers for example), Files, Messages, Data, Modems/Fax and other hardware resources. In resource sharing, the objective like distribution of cost of machines over the number of users is fulfilled. This achieves the economic solution of any design (Cost can be minimized in hardware, as well as software used by sharing them among all users.). In the load sharing, the load of any users may be shared with other computers in the network. This may be called distributed processing: by which the processing can also be done in the geographically distributed computers connected in the network. This makes the optimal utilization of machines. This solution provides higher flexibility and reliability. In case of failures of one or more machines, the other machines can be used to care of loads of the failed machines. BOX 3 In the previous chapter we mentioned about the Metcalfe’s law that states that if there are ‘n’ computers in a network, the power of the computers in a network like the Internet is multiplied by ‘n’ square times. However we may look into the issue as follows: Assume a system of n (>1) independent computers with mutually exclusive information resources each with an amount of I. The system has total resources of an amount (n.I). When all these computers are put in a network, assume each computer shares α (< 1) part of information resources of all other computers. Thus each computer of the networks is logically having an amount of information resources equals to I + (n – 1) α I. The whole network is having now total logical information resources equals to n (I + (n – 1) α I which amount to say that due to network the information resources has increased by an amount = {n (I + (n – 1) α I} – n I = n (n – 1) α I. Thus the power is nearly proportional to n2. The cost effectiveness may also be analyzed in term of the above stated simple derivation. A computer in a network at the physical cost (say, C) of I amount of resources, can use the information resources of I + (n – 1) a I amount. This results per unit information cost to decrease from [C/I] to [C/ {I+(n – 1) αI)}]. As percentage of sharing increases (α increases), the cost effectiveness increases. When there is no sharing (α = 0), there is no decrease in cost. The sharing of resources is the issue of all goals of networking.

3. Functions and Scopes of Networks The main three functions of networks are: Data collection, Data Processing and Data Transportation. Now-a-days the first two functions, namely data collection and data processing have become the subject of data management in computer engineering. As such by the matter of right, the network function and scope become the correct delivery of data from source to destination. Thus network has the scope of:

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• Data Transmitter that deals with relevant techniques, theories and equipments • Data Receiver that deals with techniques, theories and equipments • Data Transmission link that deals with aspects, behavior and remedies

3.1 Data versus computer network We often hear about data network, and again computer network. Basically computer network is a special data network. Data network is supposed to deal with data of any base; may be binary data, octal data, decimal data etc. But computer network deals with binary data only.

3.2 Why separate data or computer network? As of today the world over there has been existing a century old huge telecommunication network for telephone. Then what is the need of separate data network? Communication is high investment, long term and highly conservative technology. But computer is low investment, near term and quite liberal technology. It takes about 100 years to make wire pair obsolete. Computer power increases by 2 folds every 18 months and cost of computer decreases by 2 folds each 2 years. So we need to use voice grade lines for transmitting data but with new techniques and technologies. On the other hand there is strong reason to go for separate data network. This is because the data characteristics are different from those of the voice. Data is known as BAD IT, where “B” refers to bursty; meaning that for a very long time there may not be any data for communication, but suddenly there may emerges a huge chunk of data for a short duration. Thus the network and transmission link should have the provision to cope up with the wide data rate variation In voice communication the data rate is fairly constant. “A” refers to asymmetric, meaning that flow of data is heavily titled towards unidirectional transport. On a single stroke of command, a destination may have to response for a long period towards source; the one example of which is the transfer of bank details from a branch bank to head quarter of the bank. In the voice communication the bi-directional flow variation is around 40% to 60%. “D” stand for delicacy. The data is delicate. It is more error prone. This is because in the data communication, the machines are involved. Unlike voice communication, human perception and intelligence is absent in data communication. So care must be taken to protect data from error. “IT” stands for Insensitive to Time. Data normally does not require real time transport or on line interactive communication. Data may tolerate delay. Thus data transport may be made with intermediate buffering and storage while being delivered in between the source and destination. In fact voice can tolerate error to some extent but not delay whereas data can tolerate delay but no error. For example delay for voice must be less than 100 ms. The error of data transport depends on Bit Error Rate (BER). If BER = 10–3, it says that on average out of transmitted 103 bits, one bit will be in error. The tolerable BER in data transport should be less than 10–2. It is the “B” and “IT” features of data that gives rise to the concept of packet and packet switching as most appropriate for data networking. A session is called bursty is λ T << 1. where λ and T are respectively average data arrival rate and average delay between the source and the destination. For data, λ T < 0.01 Considering the above stated characteristics of data, the necessary theories and techniques are developed into a new branch of networking. The main subjects include: topological

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design—the way of connecting geographically distributed physical machines in a network; protocol—a set of rules for networking thereby allowing vendor independent devices to function in networks; switching—mainly packet formation and its switching and routing; Interconnecting different networks—routers, bridges, gateways; Global single network—Internet; Different interfaces; Error control and Security of data among others. BOX 4 1. What are the problems with “B” feature of data associated with transport over telephone network, often known as circuit switched network? How may the problems be solved? The major problem of data transport over circuit switched network is the poor link utilization arisen out of the “B” feature of data. Data is bursty meaning that λT << 0.01. Assume that the average transmission delay suffered by data over a physical dedicated circuit switched link is t, the propagation delay is p, then for achieving proper service quality, t + p < T. Hence t << T or λt << λT. But due to “B” feature for data λT << 0.01. This means λt << 0.01, may be even << 0.001; hence link utilization is very poor. Assume that a telephone link, dedicated physical line of 64 Kbps is proposed to transport data. Due “B” feature, sometimes data may arrive at 1 Mbps or more; and then data is most likely to be lost. But fortunately due to “IT” feature, data may tolerate delay. So intermediate buffering & storage may be done to avoid the loss; and subsequent transmission on the 64 Kbps. This gives rise to the use of “store & forward” switching as appropriate one for data transport. But what will happen, when data arrives at a very low rate say 10 bps? The link utilization will be very poor, as the unspent bandwidth must not be used by any potential users under the concept of physical dedicated link. This problem may be solved by allowing other potential users to share the link. The concept of Store & Forwarding switching with the concept of link sharing gives the birth of packet switching most appropriate for data transport. 2. The circuit switching uses link quite inefficiently in carrying data. This is due to the bursty feature of data. We can quantitatively justify the same. Assume ? is the average arrival rate of a session in a network. The average delay of the data (in between source and destination) of the network is T. Session for which ? T << 1 is referred to as bursty session. In case of data communication ? T is typically 0.01 or less. Let tt and tp are the average transmit time of message and propagation time in the network. Then (tt + tp) T or (1 + a) tt = T where a = tp/tt Thus tt < T or λ . tt < λ T As λT << 1 then λ tt << 1

tt << 1 (1/λ) As λ is average arrival rate, 1/ λ is inter arrival time i.e. ideal time of link. Whereas tt is the transmit time i.e. on time of link. Thus one form of quantitative message of link utilization is (tt/1/ λ), which as above is seen to be << 1. On the other hand, a comparative study in between message switching and packet switching can be seen. Assume there are (K + 1) computers are in a link and the computers are all equidistant apart over K links. Also assume switching processing and propagation times are or

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negligible. Let a message of M bit with overhead bits of m are to be transported over the whole of links. If C is the channel bit rate in BPS, then time required for end-to-end transfer in message switching is: M+m M Tms = K . K as M >> m. C C Here we have assumed that the node delay = 0. In packet switching, the message is to be broken into several packets. Assume there are N packets each of bit M/N and with overhead bits of p. Then time required for the first

LM M/N + p OP , while that for rest of (N – 1) N C Q

packet to reach the destination is equal to K . packets is (N – 1)

LM M/N + p OP . Thus required transmit time in packet switching is : N C Q M F M + pI +p N + (N – 1) . G N T = GG C JJJ C H K ps

We assume here also that the node delay = 0. Now, let us find the value of N for which Tps will be minimum. For this :

dTps

K.M p M + + 2 =0 2 dN N C C N C (K − 1)M (K − 1)M =p or N= 2 p N

or Again

=−

d 2 Tps

= +ve. dN 2 Putting the value of N as derived above in tps expression we get: (Tps)min =

Assume So,

p C

F GH

I JK

M + K −1 P

M/p >> K, when,

2

(Tps)min = M/C (Tps)min = tms/K

This justifies that for proper choice of packet size, the packet switching is faster than the message switching. 3. A 8192 bit message is to be transmitted over a 10-link path. Assume packet switching is to be used and overhead per packet is 256 bits. Assume processing time, propagation delay and node delay is negligible. Find (a) number of packet into which the message is to be divided for time of transfer to be minimum and (b) the minimum time so required, (c) what is the gain over message switching. Assume C = 1 MBPS. Sol (a) Know that for minimum transfer time, the number of packets: N=

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Where K = 10, M = 8192, p = 256 N = (9 × 8192)/256 =

17.

(b) The minimum time is given as: [tps] = (c) The gain

M 8192 = 8.192/ m sec. = C 10 6

8192 sec. − 10 6 10 6 = 9 × 8192 m sec = 80.2 m sec.

Tms – [Tps]min = 10.

8192

4. (a) What value of packet size minimizes transfer time, when a message of M bits is transmitted over a K – hop links. Assume (1) p is the overhead bits/packet, (2) propagation, processing and node delay are negligible. (b) Assume that an 8 inch × 10 inch image (by facsimile) is to be transmitted by such a packet switched network where p = 256, K = 10, C = 48 kbps. The facsimile digitizes the image into 500 pixels per inch (i.e. spatial resolution is about 1/500 = 0.002 inch). Assume that 24 bits per pixel is required for color representation. What is (1) size of packet for (Tps)min, (2) the minimum time required to transfer the image. Assume propagation processing and node delay are negligible, and (3) dt C = 4800 bps, what is (Tps)min. Solution: (a) For minimum (tps)min we know, required number of packet is given as : N=

(K − 1)M P

Thus if P is the packet size for minimum (tps)min, then P=

M Mp N (K − 1)

(b) Number of bits to be transfer = 8 × 500 × 24 (horizontal) + 10 × 500 × 24 (vertical) = 216 k bits M = 216 kbits. (1) The minimum (tps)min, we know P=

Mp = (K − 1)

216 × 10 3 × 256 9

= 2478.7 = 2479 bits (2) The minimum (Tps)min

=

(3) When C = 4800 bps, (Tps)min =

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M 216 = = 4.5 Sec C 48 216 × 103 = 4800

45 Sec.

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DIFFERENT NETWORKS

Based on geographic coverage area the networks are classified as below: Local Area Network (LAN)-Covering a building, an organization Metropolitan Area Network (MAN)-Covering an area over a city or metropolitan Wide Area Network-Covering an area of a zone, region or country Based on the technique of implementation, the networks may also be classified as: Peer-to-Peer and Server based. In the peer-to-peer network, the machines and computers those are connected to the network are all equal in status. There is no centralized administrator in the peer-to-peer network. In the peer-to-peer network, there is no costly server that controls and provides resource-sharing access to other computers in the network. The peer-to-peer network is called workgroup when the numbers of computers in the network is 10 or fewer. In the server based network, the dedicated costly machines known as servers are used to administer the network. The servers are designed to provide optimal resource-sharing services, administration, and accounting of the network. The computers that are connected in the network but are not servers, are known as clients. In the networks, the clients request and get the services that are only provided by the servers. Depending on the varieties of the services provided the servers, the servers are classified as: File Server, Print Server, Mail Server, Application Server and Message Server etc. Networks

Point-to-point

Multipoint LAN with bus, ring terminal controller with pool

Switched

Physical keyboard to computer antenna to TV set Logical keyboard to printer via computer star topology

Catenet interconnected net

Integrated net

Circuit telephone ISLAN/ISDN/BISDN Message telegraph/e-mail Packet ARPA net

BOX 5 An Introduction to Local Area Networking Local Area Networking The IT revolution is mainly due to marriage of computer with communication that gives rise to a field of networking that has several components of LAN, MAN, WAN and ISDN. Let us have a through review all aspects of LAN including the discussions of the goals of LANs, their structures, protocol and design hints. We also suggest in this box several modification to improve the performance of LAN.

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Introduction A local area network (LAN) is a privately owned [1] data communication network, typically a packet communication network, limited in geographic scope [2] of from 1 km and having the capacity of data rate exceeding a Mbps [3] over an inexpensive transmission media [1-3]. A LAN may also be defined as a high-bandwidth data communication system, which permits a number of independent devices such as computers, terminals, storage, printers, plotters, monitoring equipments and gateways to communicate with each other. One should not expert that a simple device like terminal can be connected to a network. Actually, “station” [4], the basic addressable device is directly connected to network. The station is made of communication system (transceiver), access controller and sufficient computing resources. In general, a computer is a station. However, terminals are connected to the network through terminal controller. LAN also is an economic solution of using single link for communication between different stations in using the B (burst) characteristics [5] of data. It is seen that terminals in interactive mode are active for only 1 to 5 percent of time. The LAN differs from a computer fitted with low-cost and high-band width BIG-BUS, in the sense that while such computer system is connecting all components or devices of its own system for some specific purposes; the LAN connects different systems and devices installed previously or likely to be installed at present and /or in future to serve different purposes. Truly speaking, the distinction is philosophical one rather than technical, geographic or topological. The IEEE instrumentation bus may be thought of a boarder-line between LAN are BIG-BUS computer. LAN is also defined as a layered local communication network. Actually, seven layers architecture of protocols of OSI (Open System Interconnection) are used in LANs for enabling two or more devices to communicate with each other. The purpose of OSI model as developed by the ISO(International Standard Organization) is to provide an architectural model by which networks (LANs, WANs etc) will be developed that will allow flexible enhancement and reconfiguration as well as interconnectivity with any other compatible networks.

Goals of LAN The main objectives of LAN are: 1. to connect together the existing computers, terminals and peripherals located in the same building, in the same office campus, or in adjacent campus in order to allow them to intercommunicate as well as to allow all of them to access a remote network or a remote host economically. This is illustrated in Fig. (1). 2. to explore the advantage of functionally distributed computing: some of machines, terminals etc. may be allotted only specific specialized tasks such as terminal handling, file storage, database management etc. By this the implementation becomes simpler but efficient. 3. to explore the computing power of other station in case of the failure of any station. 4. to explore the sharing of computing load with idle computing station in case of overloading of any computing station, and 5. to provide a cost-effective high speed data network. The designer of long haul networks of WAN are often compelled by legal or economic reason to use the low band-width public telephone network, regardless of its technical acceptability. This problem is absent to the LAN-designers. As such they are free to use highbandwidth cable which is not at all precious resource. Moreover use of WAN causes a delay

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problem in transfer. LAN attributes low-delay data network. The 100% resource sharing is only possible with LAN. The resource sharing is one of the main objectives of networking. Imagine a costly laser printer connected to a network is to be shared by users of the network. In MAN and WAN, the sharing may not be practically feasible (Why? Can an Internet user of India share a laser printer connected at USA?). But in LAN the sharing is absolutely feasible.

Components of a LAN A network is composed of three basic hardware elements and a set of software protocols in order to control data transfer via hardware. The hardware elements are: 1. a transmission medium which may either be twisted pair (unshielded and shielded), coaxial cable or optical fibre. 2. a medium-access controller (MAC) often known as local communication controller (LCC) to implement a mechanism for control of transmission over medium, and 3. an interface to the network. However, the required set of software protocols function at various level of ISO (International Standard Organization)-OSI (Open System Interface) reference model beginning from low level transport protocol to high level application protocol [6]. These software protocol are integral part of both LAN and WAN.

Network topologies The way of interconnection used among the various nodes of the network is known as network topology. Topology is also defined as logical arrangements of physical nodes. The different types of topologies known to us are pictorially shown in Fig. (2). In the fully connected topology (Fig. 2) each node connects directly with all other nodes via distinct separate link. As such, the transfer process is effective. No routing is required as each one is connected to all others. Reliability is very high. But the required number of links in this case increases with number (N) of nodes in the system following a formula [7]: Link = N(N – 1)/2 This suggests that required number of links is so high that in terms of cost-benefits analysis, this topology is not a viable topology. Moreover, operation and maintenance become a serious problem as number of link increases tremendously with number of nodes. The unconstrained topology is the general topology. It is used basically in WAN. The advantage of this topology is that depending on the communication traffic, links can be arranged. At each node, there is routine algorithm which is to route the message. Thus each node is to have a degree of complexity. It has thus no use in LAN. The star, the bus and the ring are the viable topologies for local area networking. We can discuss the relative merits and demerits of these in terms of following parameters: 1. flexibility which measure the ability of the topology to add/or delete a node to the network. 2. reliability which measures the ability of the topology to transfer data in case of failure of some nodes. 3. technological suitability. In terms of flexibility, the first comes the bus, then the star and at last the ring. So far the reliability is concerned, the bus and the ring area comparable. However the star topology completely fail to work if the central node fails. As such the reliability of topology completely

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depends on the central node. Thus technologically star network is also not preferable to the other topologies. Moreover, the central node is to contain the routine algorithm for all nodes. The secondary nodes are of course simple. Such topology is suited to time sharing computer system. In the bus and in the ring topology, there is no need of routine decision in any node. The direction of data flow is unidirectional in ring, while that in the bus is bi-directional. Thus ring topology can support the broadcast type as well as the point-to-point type of communication; whereas bus is to support only broadcast communication. However, considering economic, technical and philosophical aspects of topology; it is suggested that the efficiency of topology should be a function of flexibility, reliability, cost and management. When the management parameter is needed, especially for tree topology it requires that the effective management theory of V.A. Graicumas [8] should be taken into account. In such case, each processor of tree should not have more than three successors. A comparative study of important topology is shown in table–I.

USE OF HUB IN TOPOLOGY The centralized node of the star topology is called hub. The hub may be passive or active. A passive hub just is a connection point. An active hub regenerates the electrical signals carrying the message and then sends the signals to the connected computers. Hence active hub requires electrical power to run whereas the passive hub does not. Using the different combinations of basic topologies, a number of practical topologies are made. Examples are: star bus and star ring (Fig. 2)

Control Structure in LAN In the bus and in the ring topology, no routine software is there in any node. So a mechanism is required for these to allow its nodes to access the medium, that is to say, to select which node is to transmit at which time. This is apparently not a problem in star Network. There are different types of MAC (medium access control) as illustrated in Fig. (3). In serial polling, one station act as a primary station. The primary station is to maintain a list of all other stations known as secondary stations. The primary station send a “call” to each secondary station by its address, one at a time in turn serially. If the polled secondary has no message to send, it sends a negative response to primary station. When data is ready in polled station for transmission, it sends a positive response to the call of primary followed by the data. The primary station then sends “call” to next station. The variation of serial polling is selective polling [10]. In selective polling, priority is assigned to heavy traffic secondary station. Instead of serially, the primary station issues poll as per an algorithm incorporating priority. The disadvantages of the technique are many as below: 1. the need of a primary station: Data transfer in between any two secondary requires the co-ordination of primary. 2. the unnecessary delay spent on the nonproductive calls which are calls for which the secondary station has no data to send. 3. the response time is much, as each secondary station is to return response to primary for future call. However, the response time can be improved as well as delay caused by nonproductive poll may be reduced to some extant in using hub polling. In hub polling, the primary station generates the first poll to the nearest secondary in any pass. The secondary station either

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sends data if it has so or transfer the poll to next secondary station and so on. At the end of operation of the last secondary in the process, the poll is again generated by the primary for next pass. In the polling, the delay increases and the response time decreases nearly with the number of nodes [11]. These can be tackled by adaptive technique such as group-polling. In this technique, the terminals are grouped together. The primary station can broadcast tom all terminals are a group simultaneously. If one member or more of any group of nodes has and data to send, it response affirmative. On receiving a positive response, the primary station sends again a broadcast call to a sub-groups of the group under affirmative response. The process continues as pictorially illustrated in Fig. (4) until the node of the nodes under the data queue is searched of . In this technique if only one terminal in a group of 2n has a message, the controller needs to transmit at most 2n + 1 calls rather than 2n calls as required in conventional polling. But if all the terminals have message then 2n+1 – 1 calls are required. However, the technique is good for tree topology. Nerveless all the polling techniques are used in tree topology. The concept of token passing is a technical improvement of hub polling. It remove the delay of nonproductive call and the necessity of primary station. In this concept, usually a 8bit control token is passed sequentially around the medium. Any station in the network seeking a token crossing it, can capture the token and then send data. After sending data, it released the token to the medium. No station has authority to send data until and unless. The said token is in its possession. Priority can be introduce in token passing technique. The high priority stations may be equipped to generate special token to give a signal to other stations, if usual token is already captured by any of them, to release the same. The priority can also be introduced in token passing using the concept of token hold time. As soon as, a token is captured by a station with data queue, it initiates a timer loading a pre-assigned value in it to act as down counter. The high priority station loads a higher value in the timer, whereas less priority has lower value. By this each station is given only a fixed time. A less priority station has to pass token after assigned time even if it has data to send. The algorithm [12] is pictorially illustrated in Fig. (5). The comprehensive studies and analysis show that the token passing is an important potential MCA in bus and ring structure. The daisy chain control requires dedicated wires to pass the control as per famous daisy chain technique. It is a typical hardware arrangement of token technique with priority. On the ground of flexibility, the token passing is preferred to daisy chain technique. The so far illustrated controls are basically sequential in nature. The family of random access control begins with ALOHA technique. In this technique, as soon as data is gathered in any station, the station sends the data via common link, alongwith the address of station for the transmitted data is meant for and the parity bit. On correct receipt of data, the receiving station sends an acknowledgement via a return channel. In this process, as there is no co-ordination among terminals, there may be collision caused by two or more stations sending data simultaneously. In such a case, an erroneous data is received by the receiving station which is to return an warning or no acknowledgement the transmitting station transmits data again. The improved version of ALOHA is a technique where collision is detected by the transmitting station so that it has not at all a LAN. For local areas networking, an improved technique of ALOHA is used. An effective strategy to avoid collision on random accessing will be to listen to the medium before transmission. This idea leads to the development of the technique known as Carrier sense multiple access (CSMA). The CSMA are of three types-non-persistent

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CSMA and p-persistent CSMA and CSMA/CD (Collision Detection). In non-persistent technique, the station with data senses the carrier. If carrier is not free, it waits for a random time before it further tries. In p-persistent CSMA, the contending station if senses that the carrier is not free, it continuously monitor the carrier and as soon as free carrier is sensed it sends data with probability p. In CSMA/CD, the sending station may detect collision and then retransmits data. The concept behind CSMA/CD is in fact one that often uses when many people are talking. In terns of throughput, ALOHA, persistent CSMA, non-persistent CSMA and CSMA/CD are of respectively around 37%, 53%, 80% and 90% [13] of the channel capacity in broadcast mode. The CSMA/CD is a good technique for accessing bus. However, the technique will be more attractive if priority is assigned to stations. It is natural that some of the stations may have high traffic and others may have low traffic. But low traffic may be of such order that some time there is no packet. In such situations, the burden of traffic station becomes tremendous. If it is assumed that the stations are of M/M/1 model, the average number of packets in system, N is given by : N = λ(µ – λ) As λ is not constant for all stations, but µ is same on the threshold point of low traffic as assumed previously, the value of N in some stations will be very high. If N is to be kept same for all station, it is required the µ will very with λ i.e. priority is to be assigned.

New Technique on Priority Assignment One technique of assigning priority is suggested as below: T = K (M – m). random generator. where T = time to be spent by station on detection of a collision, K = fixed value assigned to each station (Higher traffic station should have Low and vice-versa). M = any fixed number (constant for all station) and m = number of packets staying at the station at the instant of CD. By doing so, actually µ is made a function of N. However, in CSMA/CD, there is a chance for repeated collision. The repeated collision can be avoided if different sets of number are allowed in random generator of different nodes. For example say the random generator for node 1 is a set of 1k, 3k, 5k, 7k while that for node 2 is a set of 2k, 4k 6k, 8k etc. where k is a constant. For the n number of nodes, sets may be like Set 1 = 1k, (1 + n) k, (1 + 2n) k, (1 + 3n) k ............ Set 2 = 2k, (2 + n) k, (1 + 2n) k, (1 +3n) k ............ : : Set m = mk, (m + n) k, (m + 2n) k, (m + 3n) k ............ : : Such hybrid technique (random but fixed) to a large extend can avoid collision. But still will remain collision (due to propagation delay between stations and due to existing randomness). It will be made further better if maximum propagation delay is less than nk. But this will again increase the message delay.

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The above illustrated hybrid technique reduces the probability of repeated collision. If there are k users and each user on average transmits a packet of duration Tp, every NTp seconds (N >> k) and if packet arrival at node is of Poisson distribution, then the probability of first collision is: Pc = 1 – e–Td/Tp Where it is assumed that one or more packets are attempted to transmit in the td seconds interval and the stations in the networks are all at equidistance apart. Here ρ = k/N. The probability of chance of further collision will be: Pc1 = Pc/n where m > 1 and is to be decided by the random generator. However, if the propose hybrid random generator (random but within a given set only) is used, the probability of further collision will be: Pc11 = Pc/n. Where n > 1 and for the set under proposal it is definite that n > m. Accordingly: Pc11 < Pc1 This makes the proposed technique advantageous. In addition to above control, other mechanism such as FDMA (Frequency division multiplexing access) on broadband, STDMA (Statistical TDMA), register insertion, and message slots etc. technique are there in the process. But today token passing and CSMA/CD becomes standard MAC to venders and users. However, VDM [14-17] for point LAN is another new area. A comparative study is shown in table–II.

Simple Comparison of Polling with Token Ring It is very difficult to define performance based on fixed parameters for different control mechanism. However, we can compare the performance of serial, hub and token passing (a hub polling) in ring using the concept of cycle time. The cycle time [11] is the time required to allow access to all terminals at least once and to transmit message from the terminals. It is seen that average duration (T) of a cycle for a model of Poisson’s arrival with infinite storage: T = PT/(1 – S) Where PT = total overhead (fixed component of TC) or poll time (only for polling or granting access to terminals), and S = nM λ where M is the average duration of a message, λ is the average arrival rate and λ is the number of terminals. The approximate PT [10] of different MAC techniques are as follows: (PT)Hub Poll = P + n.H +L (PT) Serial Poll = n(S + P) + L (n + 1) (PT) Conventional Token Ring =L where P = time required in initiating a poll and processing the response in primary station (both for serial and hub), S = time to receive a poll and generate response in secondary in serial poll and H = same as S in hub polling, L = total link time.

Combined Structures Some combined viable structures of LAN were emerged out of above discussion and on the modulation scheme out of followings : (i) passed band transmission. (ii) Broadband transmission using carrier modulation technique.

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These [18] combined structures are : (a) passed band bus with CSMA/CD : Example : Ethernet (Fig. 6a). It incorporates IEEE 802.3 standard. (b) Broadband bus with CSMA/CD : It incorporates IEEE 802.3. (c) Broadband bus with token passing (Fig. 6b). It incorporates IEEE 802.4 standard. (d) Passed band bus with token passing : It incorporates IEEE 802.4 standard. (e) Passed band ring with token passing : It incorporates IEEE 802.5 standard. However mixture of techniques in LAN at will always cannot work. For example : multichannel broadband fibre optics LAN is currently impractical and true broadband ring is impossible. The analytical studies in terms of response, delay and capacity show that : (i) at a data rate of 1 M bit / sec., token ring and CSMA / CD bus performs equally well. (ii) At a data rate of on and above 10 M bit / sec., token ring has other performance. (iii) Token bus offer more efficiency than CSMA / CD [19]. (iv) Response time of token passing is superior to that of CSMA / CD technique. (v) For burst mode of data transfer CSMA / CD is better than token passing technique. A more efficient version of token ring approach are available in [020 - 22]. Moreover new accessing technique to replace CSMA is under spread spectrum modulation technique.

Engineering Aspects Choice of LAN LAN has four basic application areas, namely office automation, engineering, factory and industrial process. Different types of LAN are now available in the market [18] and new LANs are coming day to day. IT is jot of the system engineer to chose a particular application. The Ethernet is a LAN based CSMA/CD bus structure. As such if we consider the merits and the demerits of the viable LAN structure mentioned previously, we be seeing that Ethernet is not suitable for manufacturing process control. This is why MAP (manufacturing automation protocol) for production plant automation is to be based on token passing bus standard IEEE 802.4. But for the purpose of offices automation we see that Ethernet is good. However, the other parameters for design to be considered is whether the net will be for only data or data plus voice. Table-v highlights some LAN that can support both voice and data. Such integration often are considered on ground of economy. Why to support two systems-one LAN for data and another PABX for voice? However, there are also some LAN which can integrate voice, voice and data. In addition to these, many other parameters such as geographic span and cost etc. must be considered. For higher data rate, fibre optic LANs [24] may also be chosen. After selection of a LAN, it is the job of the design engineer to design the system. Here we shall discuss the selection of LAN, it is the job of the design engineer to design system. Here we shall discuss the design of a work station for Ethernet. Designing an Ethernet Workstation As per Forester Research Inc, the Ethernet covers 33% of LAN market with IBM token lagging behind with 22% in the year 1987. Dataquest claimed that 52% installed LANs are Ethernet LAN. As such Ethernet is most popular and hottest. As such we are to provide a design of Ethernet station reference.

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The block diagram of an Ethernet node/workstation is shown if Fig. (7). The designer has a broad selection to choose LCC chip and SIA chip from various chips available in the market from different vendors. Out of these, the important chips [25], each of which can perform all link layer functions of ISO-OSI reference model for LAN using sophisticated buffer management techniques as well as performs all the physical Ethernet protocol functions, are: (i) LANCE (LCC chip) from AMD/Mostek/ Motorola. (ii) DP8390 from National (Network Interface Controller). (iii) 82586 (LCC chip) and 82501 (SIA chip) from Intel. The VLSI 82586 chip is fully compatible [26] with Ethernet and IEEE 802.3. I is an intelligent peripheral IC. It can completely manage the transmission and the reception of frames over the network and performs all, Manchester coding/decoding for Ethernet. Its ease of operation, flexibility and reliability are attractive. Simple drivers and receivers are not sufficient to have a physical and electrical interface to Ethernet. For this specially 82501 Ethernet SIA chip is developed.

Interconnecting LAN with LAN and WAN Different LANs Interconnection A large number of LAN having different and/or same control and physical structures may be interconnected to allow the different station under different LAN to communicate with the other stations under other LANs (Fig. 8). The job interconnection is performed by bridge, often also known as data link relay [27]. A collection of LAN and bridges as shown if Fig. 8 may be termed an extended LAN. Bridges are actually intelligent filtering devices which stores and forward frames as per address of destination. They are not like repeaters or amplifiers in Fig. 8, the station a can send frame to M or N or U or V in the same manner in which it sends frames to node C or B and so on. Extended LANs have several properties as below: (1) The traffic on LAN-1 of Fig. 8 is unseen to LAN-1 as well as to LAN-3.Because of this filtering, load on each LAN of extended LAN is reduced. This property is known as traffic filtering. (2) Each LAN has maximum geographic span. Extended LAN increases the physical extent of individual LAN. (3) Each LAN can support a maximum number of station. (4) LANs of different structures can be connected by bridge. A bridge can allow a station of IEEE 802.3 standard to send frames to a station of IEEE 802.4 or IEEE 802.5 and vice versa. And (5) Bridge allows in extended LAN to co-exist different physical layers (base band coaxial, broadband co-axial, optical fibre etc.) of different LANs. The structure of a bridge is shown if Fig. 9. Bridge may interconnect two LANs separated at best by the coverage span of individual LAN directly. However, there are situations where a LAN needs an interconnection with remote LAN. For example, the headquarter of any industry can needs its LAN to be interconnected with its of campus factory-LAN. In such cases, the concept of half bridge, often known as long distance bridge, or long distance data link relay may be used as shown in Fig. 10. Interconnection with LAN with WAN The LAN is connected to WAN through gateway, a high level data link relay; while LAN is connected to another LAN via bridge (Fig. 11). The functional difference between bridge and

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gate way is best illustrated [28] in Figs. (12-13), in terms of ISO-OSI reference protocol. Gate way operate on transport layer of OSI model or above; and connects LAN to other networks which employs different protocol such as X.25 (packet), IBM, SNA, Decent and TCP/IP etc. Gateway to gateway connection is through X.75. Further details of gateway is best depicted in Fig. (14) [25]. Conclusion The current trend in LAN technology suggests that in future this new technology of decentralization of computing power coupled with fast communication, particularly integrating voice, video and data will emerge out as full success in implementing its goals. LAN is becoming more and more attractive due to 80/20 rule [29] which claims that in any organization 80% communication is internal and 20% is external on average. The spread and adoption of LAN in every aspects of life will ultimately lead to have a global network system, the idea mooted in ISDN philosophy and by that IT revolution will be finalized. The first generation LANs (1970–1975) are of data rate of 10 M bits/sec. These are mainly used in office automation. The second generation LANs (1975–1985) and MAN’s data rate is about 100 M bits/sec. This generation is through the standardization efforts. The third generation LANs supposed to operate at G bits/sec. These are yet to be in commercial field and to be standardized under current research. Table 1 Topology

Max Circuit Speed in MBPS

Flexibility

Distance

Reliability coverage

Complexity at interface

Cast

Bus

50

Good

Unlimited

Good

Average

Average (lower than star but higher than ring)

Ring

80

Average

Limited

Good

Low

Average (Generally lower)

Star

10

Poor

Limited

Average

Low

High

Table 2 Accessing Technique

Distance coverage

Delay node

CSMA/CD

Average

Unlimited

High

Average

High

Random

Average to high (Depends on nods)

Token passing

High

Low

Average

High

Average

Deterministic

Average

Register Insertion

High

Low to average

Low

High

Low to average

Deterministic

Average

TDM

Average

Low

Low

High

Low

Deterministic

Low

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Cost per nodes

Number of required

Bandwidth

Nature

Efficiency

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IEEE 802 Specification

Access Technique

Topology

Physical Medium with Board band

Allowed data rates Base band

802.3

CSMA/CD

BUS

Co-axial cable 1MBPS 5MBPS, 10MBPS

Co-axial cable 1 MBPS

802.4

Token Passing

BUS

Co-axial cable 1.5444 MBPS, 5MBPS 10MBPS, 20MBPS

Co-axial cable 1MBPS, 5MBPS 10MBPS

802.5

Token Passing

Ring

X

(a) Shielded Twisted pair 1.4 MBPS. b) Co-axial cable 4 MBPS, 20 MBPS, 40 MBPS

(A) Parameters Ethernet

Standard

Minimal net

Board Band Ethernet

10MBPS

1MBPS Starlan Starlan

Code No.

10 BASE2

10 BASE2

10 BASE36

10 BASET

10 BASE0S

Data rate

10 MBPS

10 MBPS

10 MBPS

10 MBPS

1 MBPS

Distance covered

500 m

200 m

3.6 km

100 m

500 m

Topology

BUS

BUS

BUS

Star

Star

Physical

Thick Co-axial Cable

Thin Co-axial cable

CATV Co-axial cable

Twister pair wire

Twister pair wire

(B) Other IEEE Specification Specification

To be used in

802.6

Metropolitan Area Network (MAN) may also be termed as “medium area network”

802.7

Broadband LANs

802.8

LANs using Fibre Optic media

802.9

LANs and Integrated Service Digital Network (ISDN)

802.10

Network Security

(C)

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Long haul network

Remote host

T

Long haul network

T

T

(a) No use of LANing Remote host

Remote host

Remote host

Gate way 2

Gate way 2

Gate way 2

T3

LAN

T3

T3

(b) Use of LAN Fig. 1: In (a) each of the terminals needs special hardware interface, communication protocol etc. There may be nine different interfaces and nine different protocols. In (b) each of the terminals needs only one interface and one protocol. Topology

Fully connected (a)

Partially connected

Constrained

Star (c)

Ring (d)

Bus (e)

Unconstrained (b)

Tree (A) combination of star and bus)

Unrooted (g)

(a) Tree of topologies

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Centralized node or HUR Computer

Computer

Computer Star

Computer

Computer

Computer

Bus

Ring

Computer Computer

Computer

Computer

Ring

HUB

HUB HUB Computer

Computer Computer

Computer

Computer

Computer

Star Bus (HUBs are in bus, Computers in each hub are in star)

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Main HUB

HUB

133

Computer Computer

Computer HUB Computer Computer HUB Computer

Computer Computer

Star Ring (HUBS are in ring with main HUB, Computers are each hub are in start) Computer

Computer

Computer

Fully Connected/Mesh Topology (Not really used in practice in networks) Fig. 2 Mac

Poll (Deterministic)

Contention (random access)

Aloha

CSMAS/CD (bus)

Nonpersistent

Serial poll Selective polling

Daisy chain

Humb polling

Selective polling

Token passing used in (bus and ring)

Fig. 3

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Group polling used in (Tree)

Persistent

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Primary station

No ?

? Yes Group 1

Group 02

? Yes ?

No Sub Group 21

Sub Group 12 Sub Group 11

Sub Group 22 No ?

? Yes This node has message to sent

Sub Group 11

Fig. 4

Capture token

Load down counter

Counter = 0 ? No

Data waiting ?

Send data

Fig. 5

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Release token

NETWORK AND INTERNET TECHNOLOGY T

T

ST

ST Segment-1 T

ST ST

ST

Segment-2 R

R

T

ST

ST Segment-3

T Co-axial cable T +

RR

R

ST Point to point link (1000 m max.)

RR

ST Segment-5

ST

ST

ST

Segment-4

ST

+

+

(a) Ethernet’s large scale configuration

Head end

Modem

Modem

Modem

Terminal

Terminal

Terminal

(b) Broad band Bus Fig. 6

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CPU

Memory

Interrupt

Channel attention

8 or 16 bit system bus Local communication controller

Host interface

LCC SIA

Serial interface adapter Transceiver cable

XCVR

Transceiver

Ethernet link

Fig. 7 A

B

C LAN-1 Bus

U

Bridge Bridge LAN-2 ring

LAN-2 ring

U A

A

Fig. 8

Interface for LAN-2 LAN-1

Interface LAN-1 for LAN-02

Packet buffers

Packet buffers

Control filter

Control filter

Full-duplex communication link/point-to-point-link It may be high-bandwidth common carrier-circuit, an optical link, or a private microwave link

Fig. 9

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IEEE 802 Frame Format IEE 802.3 Preamble

Start of frame delimiter

7 bytes

1 byte

Destination address

Source Address

Length of data field

Data

Pad

Checksum

2 or 6

2 or 6

2 bytes

0–1500

0–46

4 bytes

bytes

bytes

bytes

bytes

Preamble = 101010…….10 Start of frame delimiter = 10101011 Destination/Source address = 2 bytes or 6 bytes but for Ethernet they are each of 6 bytes (d) Table 3 Check error on Transmission. It is Effectively a 32 bit Hash code of data And same for all the Three standard. 802.4

Preamble Sum

Bytes

>= 1 4

802.5 Start of delimeter

Bytes same 1. 2. 3.

Start of frame End of delimeter

1

1 1 Access Frame control control delimeter address status 1

status contains two A = 0 and C = 0 A = 1 and C = 0 A = 1 and C = 1

1

Frame Destination 2. Destination and control address

Source Data

1 standard.

02 or 6

address

Pad

delimiter 02 or 6

Destination respectively needs 10

Source address Sum

Data Check

02 or 6

02 or 6 4 Unlimited

Check

0-8182

End of

Frame

1

1

bytes A and C to act as below destination not present. destination present but frame not accepted. destination present and frame accepted.

5. INTRODUCTION OF INTERNET Internet is a network of networks. It is called mother network. Internet has dominated the world of computer communication as a single most computer network. Internet is a backbone

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network, which connects several others networks. It is known as the mother of networks or the collection of the networks. Internet is the mostly used backbone network world over. Any other network can be connected with Internet for the world wide coverage with suitable interfaces. Each network connected to the Internet has a NOC (Network Operation Centre). Though NOC, users are connected to the Internet. Internet connects thousands of local, regional and national networks together to share information globally round the clock. Internet connects about two million computers and ten thousand networks over the world. More than five millions access the Internet from over more than hundred countries. Due to such wide connection of so many networks, flow of information is hardly disrupted for failure of one or two small networks. Alternative paths are available for transmission and reception of information like the flow of traffic on highways. Thus Internet is known as Information Superhighway or Cyberspace. Historically it was lunched by ARPA (Advanced Research Project Agency) of USA Defense Department in late 60s. Internet in Network is what is UNO (United Nation Organization) in governments. There is no single organization that owns the Internet. It is run by volunteers and controlled by professional forum. Three different organization that keeps on running Internet[26] are: Internet Corporation for assigned names and numbers (ICANN), Internet Engineering Task Force (IETF) and Internet Society (ISOC). ICANN is a nonprofit corporation responsible for allocating IP address space and management of domain name and root server system among others. IETF is an international community of network engineers, managers, researchers, operators and vendors responsible for evolving and creating technical standards through consensus. . There are so many definition of Internet that one may conclude that nobody knows the real definition. Internet is the largest computer network in world. However, to avoid the confusion over the definition of Internet, in 1995 the federal Network Council of USA defined the following definition: “Internet refers to the global information system that (i) is logically linked together by a global unique address space based on the Internet Protocol (IP) or its subsequent extensions/ follow-ons, (ii) is able to support communications using the Transmission Control Protocol/Internet Protocol (TCP/IP) suit or its subsequent extensions/follow-ons, and/or other IP-compatible protocols and (iii) provides, uses or makes accessible, either publicity or privately, high level services layered on the communications and related infrastructure described here in” (source: http.//www.fnc.govt/Internet-res.html) New convergence and network technologies are breaking through all barriers of time, space and distance to user in a new phase of information revolution. Today, any computer can be linked to any other computer(s) anywhere in the world via the global network, called the Internet, for real-time, off-line communication and exchange of information. Today about 2 million computers and 10000 networks all over the world use the Internet. More than five million users access the Internet in more than hundred countries. A conservative estimate of growth of Internet[5] user up to 2000 will be as below: Year 1988 1990 1992 1994 1995 2000

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Users in Million 0.5 2.4 8.7 26.1 49.2 195

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However, there is existing a wide gap in Internet uses over the different countries of world. A statistic[6] of year 1995 is as below: Internet hosts World America Europe Asia India China

Internet users(M)

9.5 m 6.5 m 2.2 m 413 k 708 2146

23.7 m 11.8 m 7.9 m 2290k 10 k 72 k

It may be pertaining to mention here that Internet till date, is mainly for data communication. Although active research and development activities are going on to use Internet for time sensitive services like voice and video. Later we shall discuss some progress of voice communication over Internet. World wide trend of volumes of data and voice traffic is portrayed in Fig. (6) [7]. This reflects the increase demand of Internet.

Volume

Data traffic is increasing exponentially

Telecom is increasing but at a slow rate Time

Fig. 6: Telephone and Data traffic volume over time.

5.1 PROTOCOL Protocol means a set of rules or a procedure that computers, nodes and terminals attached to a network will follow for the purpose of meaningful communication. The use of protocol assures compatibility in communication. Protocol provides a common standard for communication among different incompatible or compatible devices and systems. In many computer communication network (CCN) environment (see Fig. 5) there may be increasing variety of computers and allied devices which are not always compatible with each other. In such an environment when two parties desire to communicate with each other, it is necessary to have some rigorously defined set of rules, based on which the communication will take place reliably and correctly. The set of rules to be adhered to by communicating parties in order to establish orderly information exchange and to efficiently manage the network resources is known as communication protocol. A great deal of effort has gone into the formulation of these protocols in computer communication. The reason is that unlike voice network, the CCN operates without human intervention. In computer communication, the related functions (like call initiation, maintenance, implication, monitoring and termination etc.) of human operators of voice network

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are done by the machines like computers. The situation is further complicated due to the fact that the CCN has to carry out many more operations for a successful exchange of information, than the voice networks. This is due to some unique characteristics of data, which are quite different from those of voice. These characteristics of data may be termed as “BAD” where : B is for bursty, A is for asymmetric, D is for delicacy. Due to “BA” features of data, packetisation and intensive buffering and storage are required for economic and efficient exchange of data, which gives the birth of packet switching. The powerful error control techniques and strategies are required to handle “D” features of data. Single bit errors are also dangerous for data exchange. Mind that human perception is no longer there in data communication.

5.2 SWITCHING TECHNIQUES The main objective of compunication is resource/load sharing to provide considerable economies of use. As such an efficient means of switched communication to access the resources of the network reliably and at very high speed to meet thereby the interactive demands was badly felt. This need gave rise to radically new concepts of switching. So many different techniques were developed to tackle the different characteristics of different traffic. In voice communication irregular gaps between words are not acceptable. Real time and interactive services are required in voice communication. Voice is the time sensitive/time dependant traffic. The voice traffic uses a kind of switching known as circuit switching. The characteristics of data is known as BAD-IT, that is to say, Bursty, Asymmetric, Delicate and Insensitive to Time. Bursty feature of data and its independence to time justifies a kind of switching that is different to voice traffic. This gave birth to packet switching suitable for data communication. Moreover to provide multimedia service, integrated handling of information consisting of time-sensitive and timeinsensitive data types requires a kind of switching quite different to both packet and circuit switching. ATM (Asynchronous Transfer Mode) Cell switching is a compromise switching technique to handle integrated traffic of voice, video and data etc. While digitized voice may be encoded into 64 kbps for quality communication, video requires a much higher bit rate. Speech may be assumed a slightly bursty as it consists of talk spurts (40%) and of silence spurts (60%); video is basically analog and highly time dependant in nature. The different switching techniques may be classified as below(Fig. 7): Switching Techniques

Circuit switching/STM

Store-and-forward switching/ATM

Message switching

Packet switching Very short fixed length packet/ATM cell/fast packet switching

Fig. 7: Tree of switching techniques

In circuit switching, Transmission between a source and destination is done through a dedicated physical link for the entire duration of transmission. The entire link remains dedicated and no other potential and/or emergent users can use it even when the path happens to be idle.

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Only when the path is released, can others use it. A circuit-switching telephone circuit is only 30-40% efficient. Most of the time is spent on listening. Circuit switching is used in telephones. Synchronous transfer mode (STM) uses circuit switching. The common T-carrier for digitized voice uses STM. (Fig. 8) In order to avoid the irregular laps/delays in between words and/or bunches of words, synchronized time-division multiplexing is used. Each slot in the time-division multiplexing system is assigned to a voice call and thereby the access is guaranteed as long as the call lasts. All the multiplexed time slots form a frame. Each slot in the frame is synchronized through the frame bit, and by position. ATM technology, in general represents a class of switching techniques where synchronization is maintained by the header or label. ATM cell switching is a special class of ATM technology which provides multimedia service at a very high speed. In store and forward switching, transmission is done on demand basis rather than on fixed reservation basis. Thus this technique has an advantage over the circuit switching technique. Each communication link is fully utilized whenever it has any traffic to send, thereby increasing the link utilization. Moreover, the demand basis allocation decreases the delay in any communication network. However, unlike in circuit switching, in this technique, queuing delays at nodes are always present owing to two reasons: (a) data comes to a node from different sources; and (b) storage capacity at any node in any network is always limited to a practical consideration. A kind of store and forward switching known as message switching has existed for many decades in telegraphy. While message switching is primarily intended for non-real time manto-man communication, packet switching is meant for fast machine-to-machine communication, including terminal-to-machine communication. This technique is also employed for designing of computer/data network. In store and forward switching, switching is done by a message or packet (a chopped message) as the path is already there, whereas in circuit switching, switching is done by hardware. In message switching, the message (a meaningful information or a set of data) as a whole is sent to its nearest node from a source depending on the availability of free link or not, it is either stored or passed for next chance. A major problem in message switching is that if a message is long, an urgent message may have to be sacrificed for an ordinary one, as once a message is on transmission it cannot be stopped. In order to avoid this problem and to make it more practical, the concept of packet switching is introduced. Message switching has another problem of overflow at the buffer of nodes. As the whole message will go as a unit, the intermediate nodes must have sufficient buffer to store different size of messages. This does not make sure that some data may be lost due to insufficient buffer space, until and unless theoretically the buffer space in unlimited or infinite. Packet switching is a form of store and forward switching where a message is broken into a number of packets each of say 1024 bytes prior to transmission. However in packet switching a packet may be of variable length. Thereafter, the packets are sent like in message switching. For proper sequencing at the receiver the packets are allocated a sequence number and destination address. A packet switching network is expected to deliver its packets in a fraction of a second, whereas a message switching network is expected to deliver its message typically in a fraction of an hour. Moreover, whereas in packet switching mode, node deletes a message from the memory as soon as an acknowledgement of the correct receipt is received from the next node,

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the message switching systems usually files the message for possible retrieval in future. Due to this whereas, a message switching network usually bears a tree or a star topology, a packet switching network bears a mesh topology where no particular node dominates the structure. Frame bit

A

B

C

A

B

C

A

B

C

A

B

C

A

B

A

(a) STM

Flag

2-bytes header

Variable-length

Information field

Frame check sequence

Flag

(b) Conventional Variable Length Packets 5–byte header

48–information field (c) Fixed size packet/ATM cell

Fig. 8: Illustration of different switching /Packets

The packet switching technique incorporates both the advantages of circuit and message switching. In the packet switching there are two approaches–(a) virtual circuit and (b) datagram for transmitting data from a source to a destination. In virtual circuit approach, a logical connection in between the source and the destination is established prior to transmission of packets. The approach is similar to circuit switching, but the established path is not physically dedicated unlike in circuit switching; and the established logical path may also be shared and used by other potential users. In the datagram service, each packet gets a path (throughout a network in between the source and destination) based on current information available to the node. The basic difference between the virtual circuit and the datagram approach is that the node does not take a routing decision for each packet in virtual circuit approach as is done in datagram approach. The virtual circuit switching is associated with call set up time unlike data gram service. This is why for high volume of data virtual circuit switching is preferred to datagram switching, and for low volume of data. Data services is preferred to virtual circuit switching.

5.3 LAYERED PROTOCOL The protocols are quite complex. There are many protocols for CCN. But it is desirable that there should be a widely acceptable standard so that all and any types of machine can work under such protocol. However in order to reduce design complexity, most CCNs are based on layered architecture (Fig. 9). Each layer or level of protocol performs a well-defined function and each one is built upon its predecessor. The number, the name, the function and the content of layers differ from protocol to protocol and hence from network to network. But the ‘n’th layer on one machine can only communicate with ‘n’th layer of another machine in any CCN under a peer-to-peer process. The set of rules used to communicate in any n-layer network is known as n-layer protocol. The set of layers and protocols is called network architecture. Between each pair of adjacent layers, there is an interface, which describes the services to be provided to the upper layer by the lower layer OSI-ISO (open system interconnection of International Standard Organization), SNA (system network architecture of IBM), DNA (digital network architecture), ARPANET

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(Advanced Research Project Agency Network of the USA) and TCP/IP (transmission control protocol/Internet Protocol) are example of networking protocols. TCP/IP is the Internet networking protocol. All protocols are based on hierarchical or layered architecture. Layered architecture provides many advantages, viz., simplicity in system design, simplicity in operation and maintenance of the system, flexibility, standardization, etc. In layered architecture, the whole operation is logically distributed among different layers. The layers are independent of each other. The layered architecture provides a means of designing system in modular fashion. Each module is built upon its predecessor. Each layer takes inputs from an immediate higher layer at the transmitting end, and from the next lower layer at the receiving end. The layer performs a function or a group of functions on the inputs. The relation between function, of a layer and arguments and values is: Value = f(argument) or Output = f(Input). Inputs are often known as arguments and produce outputs (outputs are often called values). The functional unit of a layer that performs a function or group of functions is called entity. Interactions between the entities of adjacent layers are done by primitives through service access point (SAPs). Primitives are requests, indications, responses and confirmations.

5.3.1 OSI Protocol At present, the most talked layered protocol is a seven layer (Fig. 10) OSI (open System Interconnection) protocol developed by ISO (International Standard Organization-which is a D-class member of CCITT). However OSI is an ideal and excellent guide to the protocols of the CCN. It tries to bring the uniformity in complex and diverse situation thereby bringing economic benefit. It is “OPEN” to any machine for interconnection to any CCN. The layers are application, presentation, session, transport, network, data link and physical (Fig. 10). These layers perform the necessary information processing and communication functions for a coordinated, complete and meaningful communication. Lower layer performs more communication related function than information processing whereas higher layer does the reverse. Lower layers are technologybased and hardware oriented. Higher layers are more software based. The name and the function of each layer of OSI protocol are as below: Layer 1 is the physical layer. It concerns with the physical process of getting data from one end to another. Here the optical or the electrical signals on the physical medium are converted in the digital data (sequences of zeroes and ones) that can be handled by any processor. It thus controls the transfer of data bits on the physical medium (electrical/optical cable) and the associated hardware (like drivers and receivers). Layer 2 is the data link layer. This layer groups different bits like data bits, address bits, control bits as well as error controlling bits into a frame. The assemble and the transmission of frame are done by this layer. For frame formatting , it may use protocol like HDLC (High Level Data link Control) or SDLC (Synchronous Datalink Control) or character oriented protocol (used in ARPANET), etc. However error control may be treated as an additional task of this layer depending on network environment. For long haul network (long error prone), error control may be done at several different layers including Datalink layer. For LANS (relatively error free transmission), error control at this level may be eliminated so that the cost is reduced. Layer 3 is the network layer. It deals with the flow of data from source to destination over tandem links. At each node of the network, the network layer determines the next node to where the data will be sent for its ultimate destination. Routing, switching, storing and buffering are the jobs of this layer.

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Thus it is seen that the first three layers deal with the components of the path of data from source to destination. Layer 4 is the transport level. This layer is not related with relay or routing of data. It provides with the enhanced quality of the network service by the End-to-end control and the optimization of the use of the network resources. The transport layer enables message reassembling from packets and end-to-end acknowledgements. The transport layer is independent of communication media and technology. The remaining three levels are known as higher level protocols. Layer 5 is the session layer. The users initiate and terminate calls at this level. It is like logging in and logging out to computer system. The major responsibility of the layer is to ensure an uninterrupted session. If somehow the session is broken, last mutually recognized check point may be used for restarting without going for a beginning of another session. Layer 6 is the presentation layer. At this level information is encoded in the desired form (for example : EBCDIC may be converted to ASCII); and the encryption for data security is done. Layer 7 is the application layer which is under the control of users. Its operations are to define as per specific application (banking travel reservation, medical information). It deals with the management of files and databases. In performing their functions, each layer adds a header at the transmitting end. The headers are then removed by the corresponding layer at the receiving side. This process of adding and removing the headers by the layers is known as the peer process (Fig. 12). A data terminal equipment or device (DTE) covers all seven layers of the protocol. A communication node covers only the lower three layers of the protocol. Accordingly, different networks may use only a collection of layers of this protocol based on their own requirements. For example, IEEE 802 series local area networks (LANs) use only the lower two layers, namely data link layer and physical layer. Other layers are not required in a LAN environment as it is a private network in which the concept of the nodes and DTEs is fast vanishing. Here terminal stations are each like a node. The presence of a network layer is not required as LANs share a single medium, in contention mode, for the purpose of communication. BOX 6 1. A node under ISO-OSI reference is made of lower three layers whereas a complete host/terminal must have full 7 layers (following figure). Terminal

Terminal Multiplexer

Node/switching node/three layers

Host/full seven layers

Terminal/ full seven layers

Communication sub network Node/switching node/three layers

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2. A detailed look into operations of lower four layers: Data from upper layers at the source end

Data from upper layers at the receiving end

Transport/data is spilt into segments each with a segment number

Transport/data segment are recovered and grouped for delivery to upper layer. Any lost segment is taken into account at this stage/end to error control may also be done

Network/address of source and that of the destination are added with each segment to produce data pack

Network/addresses of source and destination are recovered and based on these routing and delivery decision are taken

Data link/data frames are produced for each data pack with necessary headers

Data link/data frames are recovered and decision taken accordingly/node to node error detection and control etc.

Physical/data is converted into binary form for physical transport

Physical/data is converted into binary form for physical transport

Equivalently

Transport : Segment

Segment : Transport

Network :: Data pack

Data pack : Network

Data link : Frames

Frames : Data link

Physical Bits

Bits : Physical

3. Detailed functions of higher three layers may be seen as follows Application Layer: In OSI reference the network applications may be (1) File Transfer, (2) E-mail, (3) Web access/ www access, (4) Network management and (5) Remote login. Computer applications may be Word Processor, Web Browser / www Browser, Data base and Spreadsheets. The computer applications may make use of network applications in the application layer. Presentation Layer: The layer performs the conversion, formatting and coding so as to send the data correctly and securely to the destination. The formatting and conversion may be typically: For Text: ASCII, EBCDIC, Character based For Graphics: JPEG, GIF, For Audio: MIDI, Mp-3 For Video: MPEG

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The encryption standard in relate to security may be: DES, AES, TDES, IDEA: Secret key techniques RSA: Public key technique PGP: Pretty Good Privacy that uses RSA but highly optimized and fast technique Session Layer: It sets up, manages and terminates the sessions between applications. The typical sessions are: • FTP: File Transfer Protocol used to transfer files • HTTP: Hyper Text Transfer Protocol used to transfer files between web server and client • SMTP: Simple Mail Transfer Protocol used to send/receive e-mail • SNMP: Simple Network Management Protocol: used to manage network and devices • NFS: Network File Service used to link file systems • RPC: Remote Procedure Call used to run remote applications • TELNET: used in remote login • DHCP: Dynamic Host Control Protocol: used to assign network addresses • DNS: Domain Name Services used to convert logical addresses into network addresses • BOOTP: Boot Protocol used to assign Ip address based on MAC address • WINS: that assigns network addresses • NNTP: Network News Transfer Protocol • FINGER: used to get user’s information • NTP: Network Time Protocol • SQL: Structured Query Language used to transmit database information over network Layer boundary PCI POU PCI POU

POU

(a) SDU’s, PCI’s and PDU’s.

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Layer n +1 Request

Confirm

Response

Indication

Layer n Peer to peer protocol

Fig. 9: General illustration of Layered protocol Peer to peer application protocol

Layer 7 application

Peer to peer presentation protocol

Layer 6 presentation

Layer 7 application Layer 6 presentation

Layer 5 session

Peer to peer session protocol

Layer 5 session

Layer 4 transport

Peer to peer transport protocol Node/switch

Node/switch

Layer 4 transport

Layer 3 network

Network interface protocol

Network interface protocol

Layer 3 network

Layer 2 data link

Link control protocol

Network interface protocol

Layer 2 data link

Layer 1 physical

Physical interface

Physical interface

Layer 1 physical

(a) ISO OSI Protocol Reference Model LAYER NO.

NAME

MAIN FUNCTION

Interface

7

Application

Users’ program

6/7

6

Presentation

Transformation and security

5/6

5

Session

Call establishment and synchronisation

4/5

4

Transport

End-to-end processor

3/4

3

Network

Rotting and switching

2/3

2

Data link

Framing and error control

1/2

1

Physical

Encoding isolation modulation

(b) Details of OSI ISO Fig. 10: OSI ISO protocol

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7

OSI Application

7

6

Presentation

5

Session

4

7

6

SNA Nau and presentation services

5

Data flow control

5

Transport

4

Transport control

4

3

Network

3

Path control

3

Routing

2

Data link

2

Data link control

2

Data link control

1

Physical

1

Hardware

1

Physical

6

DNA Network management and application Session control and end communication

///  Not Protocol Exactly

(a) OSI with SNA and DNA Application

FTP/SNMP/SMTP/Telnet/NFS

Presentation Session Transport

TCP/UDP

Network

IP/Routing/ICMP

Data link

ARP, RARP

Physical

Coax, twisted wire pair, fiber cable

(b) OSI with TCP/IP Fig. 11: Comparison of some important protocols

5.4 HDLC/SDLC FRAME OF DATA LINK LAYER A HDLC or SDLC packet or frame has the basic format as shown in Fig. (12). The frame has 6 bytes header fields and variable information field. The information field contains the actual data bits. If there are no informative data, the valid frame will be only six bytes. The flag fields are used for receiver synchronization. The star flag and the end flag are each one byte and made of 0160 or 01111110. When a receiver gets the start flag, it starts accepting frame. The end flag indicates the end of frame. The end flag is essential as the frame size is variable due to variable information field. The address field is one byte. The address field provides the address of either sender or receiver as the case may be which we shall see later. The address field is fixed one byte in the case of SDLC, but it may be of multiple bytes in the case of HDLC that is discussed later. The one byte control field defines different types of frames and their services. The two bytes FEC/FCS (Frame Error Check/Frame Check Sequence) field is used to control error in the packets. Start flag

Address

Control

Information field

FEC/FCS

End flag

1

1

1

Variable

2

1

Packet Frame

Fig. 12: HDLC/SDLC frame format (Note: frame and packet relation as used by others is shown. But we will use frame and packet interchangeably as logically they use to mean one and same thing)

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5.4.0 Error Control The data communication network or the computer communication network or the digital communication deals with the transmission and reception of data. Let us restrict ourselves to binary data. Binary data is a string of 0’s and 1’s only. Error is said to occur if 1 is changed to 0 and/or vice-versa. In some cases a single such change may cause a havoc. Let us take an example, say a data byte 10111101 represents the amount of credit in rupees thousand of any account. This data is required to be transferred from a source bank to a destination bank. During transmission if say just only one bit, LSB or MSB 1 is changed to 0, the received credit at destination will be 00111101. The received credit is less than original credit by Rs 128000 due to single bit error. As data network or computer network is mostly operated by machines, this error will be there in the credit. Such type of delicate consequence is less in voice communication due to human perception and involvement in voice communication.

5.4.1 Types and causes of error The channel and/or the system noise causes error. In network environment there are basically two types of noise: Random noise and Burst noise. The transmission line usually attributes random noise which is often known as Gaussian noise or additive white noise (AWN). The random noise causes random bit error in data or computer networks. The random error means that the actual bits which reverse during transmission are randomly distributed over message or word. The random error may be single bit error (SBR) or multiple bit error (MBR). When only a single bit is in error of a code or a word, we call it a case of SBR. In case of MBR, there may be two or more bits in error per code or word, but such bits are not the adjacent sequence of bits. Burst error is associated with a high probability of bit errors occurring among a sequence of adjacent bits. Errors are clustered in burst error. The burst error occurs mainly due to system’s component failure or sudden on and off switching of heavy electrical machines like motors etc. However burst errors in transmission links also occur due to impulse noise produced lightening etc. Another reason of burst errors is due to failure of radio transmission system caused by rapid fading. Thus for a completely reliable data communication we need to have strategy and techniques which can control both the random and burst errors in any network. This is one of the reasons that the CRC (discussed later) is so much used in network.

5.4.2 Parameters of error measurement The bit error of any link or system is measured by a probability known as bit error rate (BER) probability. The BER is usually denoted by a and mathematically defined as : α = Lim (n/N) for N tends to infinity where n is the number of bits which are in error out of total N transmitted bits. Thus if α = 10–6, it means that there will be only one bit in error per 106 transmitted bit on average. In analog communication signal-to-noise ratio (S/N) determines the quality of reception. In digital or data communication it is the bit energy to noise density ratio(Eb/No) that determines the reception quality. That is to say (Eb/No) controls a. For example, in FSK communication, α = 0.5 Exp(– 0.5Eb/No) However, the sensitivity of data normally determines the maximum tolerable BER for any application. In case of speech, we know from our experience that minimum S/N required

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for acceptable reception is 30 db. For data, it is the system and the application which determines the maximum allowable BER. For example, military and financial transactions normally require a smaller BER than that required for entertainment purposes. Based on allowable BER, the control parameter (Eb/No) may be found out. As an example, we can address the problem for an ideal AWN channel. The capacity (C) for such a channel is given by famous Shannon’s Limit Theory, C = B log2 (1+S/N), where B is the bandwidth of the channel. N = ηB (when h is the noise density i.e. noise per unit bandwidth) defines the noise over total bandwidth of the link. Ideally speaking it seems that the maximum channel capacity may be met by infinite bandwidth though it is nit the case. Readers can analyze the reason behind it. But: Cmax = Lim {B log2 (1+ S/ηB)} where B tends to infinity. = (S/η) Lim {log2 (1+ S/ηB)ηB/S where (Bη/S) tends to infinity = 1.44 (S/η) ...(1) ...(2) Again, Eb = S. Tb = S/C where Tb = bit duration = 1/C For a limiting case of equation (1), equation (2) can be equated assumed equal S. In that case, in equation (2), Cmax will correspond to Ebmin. Cmax = (1.44) . (Cmax . Ebmin/η) ...(3) or (Eb/η)min = 1/1.44 = -1.59 db Equation (3) corresponds to maximum channel rate or infinite bandwidth. This may be assumed to approach zero bit error probability. Thus ideally speaking when (Eb/η) >= – 1.59 db, a tends to zero. But in practice even when Eb/η equals to 5 to 10 db, BER is seen to be typically of the order of 10–4 . This is because a channel in practice is never an ideal AWN channel.

5.4.2.1 Probability Model of Error There exist many different probability models for random variables. Two discrete probability functions for examples are Binomial and Poisson. Similarly examples of continuous functions are Rayleigh and Gaussian. From digital/data/computer communication point of view, our interest lies in binomial distribution function. This function deals with integer-valued discrete random variable. From point of view of data communication, probability that there is ‘k’ errors in a code word of length m where ‘k’ lesser equals to m, is given by binomial frequency distribution function : P( k, m) = mCk . (α)k .(1 – α)m – k ...(4) m where α = BER and Ck = m !/(m – k) ! k ! The equation (4) can be proved indirectly. Consider 3 bits code word. As alpha is the bit error probability rate, assuming bits are independent with each other; the probability that all the three bits of the 3 bits code are in error is given by : P(3, 3) = (α)3 ...(5) 2 Similarly the probability that any two bits of the 3 bits code are in error is α .(1 – α). Number of ways the 2 bits error can occur in a 3 bit code is 3C2.

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Thus the total 2 bit error probability is : P(2, 3) = 3C2 . α2 .(1 – α) ...(6) 2 The probability of any single bit error is α.(1 – α) . Number of ways that a single bit error can occur in a 3 bit code is 3C1. Thus the total single bit error is : P(1, 3) = 3C1 .α.(1 – α)2 ...(7) 3 The probability that there is no error in a 3 bit code is (1 – α) that is , P(0, 3) = (1 – α)3 ...(8) The general form of the equations (5) to (8) is the equation (4). For equation (4), the mean and the variance (the square of the standard deviation/second central moment) of P(k, m) can be found as : Mean = m . α ...(9) Variance = m . α . (1 – α) ...(10) When P(k, m) is to mean probability of k bits in error out of transmitted m bits. The relative spread defined as: (Square root of variance)/mean = Square root of {(1 – α)/m . α}. For fixed α the relative spread decreases by square root of 1/m. The likely value of errors is near the mean as m is large.

5.4.2.2 Error Correction and Detection Codes When a message m is transmitted from source to destination, the message is corrupted by the noise during transmission. The received message Mr at the receiver may be represented as Mr = M + E where E represents the error in the message. The error correction code (ECC) in general is capable of getting back M out of Mr; whereas the error detection code(EDC) in general is capable of detecting error, that is, it can declare that Mr is not correctly received as per transmitted message. We put the word “in general” because EDC/ECC may be capable of detecting/correcting SBR only and/or double bit error only etc. Any EDC or ECC is generated by adding a few extra bits known as check bits to the message. If message is of n bits, any EDC or ECC for such message need c numbers of check bits and we call such code as (m, n) code where m = n + c. The check bits are redundant bits. Redundancy increases reliability. For example, say you are interested to send two different data X and Y to your friend staying far away from you. In addition to sending just two data, X and Y you can send also two more redundant data a and b where they are respectively, (X+Y) and (X – Y). On receiving X, Y, a and b your friend can find out using a and b , X and Y. If the values of X and Y do not match with the received x and y, we can easily identify the occurrence of error. The same idea is there in designing EDC and/or ECC. The issue of delicate feature of data is related to the “accuracy” or “reliability” of data communication. The accuracy of data communication is maintained by the basic two techniques: FEC (Forward Error Correction) technique and BEC (Backward Error Correction) technique[1]. The techniques use respectively error correction codes (ECC) and error detection codes (EDC). ECC and EDC are made of redundant check bits appended with the actual informative data bits. Redundant check bits are used for error correction and detection. If data size is n bits, any ECC or EDC may be of size m (m > n) bits where number of check bits in the code is (m – n); and the code is called (m, n) code. The use of FEC and that of the ECC and the EDC was reported by Claude E Shannon as early as 1948 [50]. Thereafter a number of codes were introduced:

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(a) R W Hamming introduced one-bit error correcting codes in 1950. (7,4) code and (13,8) code are the examples of one-bit ECC. (b) P Elias developed convolution codes in 1955. (c) In 1959 R C Bose and D K Chaudhuri proposed multiple error correcting codes. These are very powerful codes and known as generalized Hamming codes. A Hocquenghem independently designed the codes proposed by Bose and Chaudhuri. That is why these codes are known as BCH codes. (d) In 1960 I S Reed and G Solomon designed a powerful block codes particularly for burst errors. The codes are known as Reed Solomon codes. (e) In 1960 G D Fornery introduced the concept of concatenated codes. (f) In 1967 A J Viterbi introduced an important convolution code known as Viterbi code Turbo code (g) Turbo code, Low Density Parity code[51], combined Turbo Code[52] and Punctured Turbo Code[53] are other important codes. ECCs are used in FEC[39-43]. EDCs are used in BEC[44-46]. More the check bits in a code, more powerful the code becomes both for error correction and detection. But with more the check bits, more the bandwidth is required for the transmission of the code. For a (m, n) code, the required additional bandwidth will be [{(m – n) X Original Bandwidth}/n]. The trade off between increased bandwidth and capability of the code acts behind the selection of a code for a particular application. The trade-off is shown in Table(I): Table 1: A brief comparision of a few EDCs and ECCs Code Type

Code Name

Detection/Correction capability of the code

% of additional bandwidth requirement

EDC

(5,4)—one bit parity check

Basically one bit error detection—but in general odd numbers error detection

25%

EDC

CRC-32 (4 bytes check bits) as used in IEEE 802.3 LAN —maximum packet size 1526 bytes and minimum packet size 72 bytes.

Single bit error detection, also double bit error detection on a large scale, Most burst error detection etc

0.26% with maximum packet size.

ECC

(7,4) Hamming Code

One bit Error correction

75%

ECC

(13,8) Hamming Code

One bit Error Correction

62.5%

ECC

(23,12) Golay Code

Three bit Error Correction

92%

5.55% with minimum packet size.

ECC always have more check bits than EDC and hence requires more bandwidth. Code capability and complexity in system design are the other parameters for selection of a code for particular applications. Transmission errors are typically two types: Random and Burst. Error is called random, if the bits in error are randomly distributed over the code. Burst error occurs when bits in error are clustered together over the code. For a transmitted byte 01010101, the examples of random error and burst error may be as below (underlined bits are in error):

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01110111 ---- random error --- errors are distributed and in second and sixth bit locations, 01101101 -----burst error --- errors are clustered on fourth, fifth and sixth bit locations. CRC is the code that can detect both the burst and the random error, unlike Hamming code or parity code that can basically detect random errors only. Hamming code or parity code may be used for detection of burst errors; but that will increase the design complexity to a large extent. Two-dimensional Hamming Code may be used for detection of burst errors; but in this case not only the complexity increases but the block wise data transmission becomes imperative also. CRC code does not create any such problem. For example, CCITT_CRC with generator polynomial, x16 + x12 + x5 + 1 can detect all single bit errors, all double bit errors, all errors over odd number of bits, all burst errors of length less than or equal to 16, 99.97% of burst errors of length 17 and 99.998% of burst errors of length greater than or equal to 18. The system design of CRC is very simple. These are the reasons behind using CRC in most of the data communication and computer communication networks including ARPANET, Ethernet LAN, and Internet etc. Performance of CRC is quite high[47]. In FEC, a data is transmitted for one time only. If error occurs in the data, the same is corrected at the receiver. Naturally ECC is used in FEC. In BEC, if data is received with error, the receiver requests the transmitter for the retransmission of that data. Unlike in FEC, in BEC the receiver does not correct but only detect the presence of error in the data. In the BEC technique, the bits-in-error of the data are corrected by the means of retransmission. BEC thus uses EDC and needs a feedback path for requesting the transmitter to retransmit the data or packet in error. As FEC uses ECC, it requires more check bits and more bandwidth for a particular data size, than those are required in BEC that uses EDC. Accordingly in long haul communication, BEC is used rather than FEC. But with increase of channel error probability, throughput efficiency (informative data size in bits divided by actual number of bits transmitted for final correct reception of data) of BEC decreases, due to increased probability of more retransmission expected with increased error probability; whereas throughput remains constant in FEC. However BEC provides higher reliability than that of FEC (Table 2 compares FEC with BEC). Many studies[2-4] show that BEC techniques perform well in many forms of transmission errors, and offer better performance than FEC techniques for wide and practical ranges of signal to noise ratios. This is why in many real applications and in long haul communications BEC is used invariably. and

Table 2: Comparison of FEC with BEC Forward Error Correction

Backward Error Correction

1. Uses error Correction codes ...... (a) coding efficiency is less (b) requires more bandwidth

1. Uses error detection codes ...... (a) coding efficiency is better (b) requires less bandwidth

2. Correction is done through codingprocess ...... does not require any feedback path

2. Correction is done through copy ...... retransmission of erroneous-feedback path is essential

3. Throughput remains fairly constant

3. Throughput decreases with increased bit error rate

4. Mostly used in local/short distance distance communication/localized system and wireless communication

4. Mostly used in long haul/long communication/networks ...... LAN/MAN/WAN etc.

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The codes may be of different types like block codes and convolution codes. The block codes are designed based on message blocks. The convolution codes are based on a very large size of message and as such a large string of series of 1 and 0. The convolution codes are basically used in satellite communication when S/N is very small. The block codes are again of two types : Linear block codes and Cyclic block codes.

5.4.2.3 Linear block codes There are both EDC and ECC under the class of linear block code. We shall first consider EDC’s. The simplest EDC is parity code. The parity code may be systematic or non-systematic. For a (m,n) code, whenever n information bits occur in the first or last of m bits code, the code is called a systematic code. Therefore (n + 1, n) bit code which is a simple 1 bit parity code may be designed as a systematic code. 5.4.2.3.1 Simple Parity Code Assume (n + 1, n) code. Data is n bits in length. One check bit is used to restore either even parity or odd parity. By even parity we mean that the code should have even number of “YES” states. Odd parity means that the code should have odd number of “YES” states. Receiver accordingly should check either for even parity or odd parity. If receiver fails to check the same, error detection is declared. The operation is illustrated in Fig. (13). EVEN PARITY Information  Parity Encoder  Cdn–1 ---- d1d0 (transmitted bits) Bits (dn–1 ---- d1d0)

Link R

R

R

R

R

C dn–1 dn–2 ---- d1 d0  Parity Decoder  Output (corresponding received bits) Declaration of Error, if any

Fig. 13: Parity Use Illustration

C = dn –1 ⊕ dn – 2 - - - - - ⊕ d1 ⊕ d0 Cg = dn – 1R ⊕ dn – 2R - - - - - ⊕ d1R ⊕ d0R Detection of error = Cg ⊕ CR where 1 = error and 0 = no-error. Here diR (‘i’ = n – 1 - - - - - 0) stands for received data bits corresponding to transmitted data bits di . CR stands for received check bits whereas Cg stands for generated check bit based received data bit. Limitation of such a parity based EDC is that it cannot detect occurrence of even number of errors ( why? Logic of XOR operation)

5.4.2.4 A Non-systematic Parity A typical non-systematic parity EDC may be (n + 3, n) where say check bits C2 C1 C0 may be placed as , dn – 1 dn – 2 C2 dn – 3 - - - - - - - - - - C1 C0 d1 d0

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And the encoding algorithm may be as : C2 = dn – 1 dn – 2 dn – 3 C1 = dn – 1 dn – 2 dn – 3 - - - - - - - - d2 d1 C0 = dn – 1 dn – 2 dn – 3 - - - - - - - - - - d2 d1 d0 However due to more use of check bits(more redundancy), possibility of power of detection is increased (can you justify ?)

5.4.3 Hamming code ( 7,4) Hamming code is a single bit error correction code for nibble data. For a nibble data d3 d2 d1 d0, the encoding rule for check bits C2 C1 C0 is as follows : C0 = d0 d1 d3 C1 = d0 d2 d3 C2 = d1 d2 d3 At the receiver the generated check bits are found out as : C0g = d0R d1R d3R C1g = d0R d2R d3R

C2g = d1R d2R d3R Where diR( ‘i’ = 0 to 3) is the received bit corresponding to transmitted bit di . Thereafter at the receiver, a comparison word as below is found out : C2R C1R C0R C2g C1g C0g ________________

E2 E1 E0 (Comparison Error) Where Ci (‘i’ = 0 to 2) is the received bit corresponding to transmitted bit Ci . The decimal value of word , “E2 E1 E0’’ points the error location in the code which is as below : Decimal value of E2 E1 E0 R

0 - - - - - - - - - - - - - - - - - - - no error 1 - - - - - - - - - - - - - - - - - - - error in C0R

2 - - - - - - - - - - - - - - - - - - - error in C1R 3 - - - - - - - - - - - - - - - - - - - error in d0R

4 - - - - - - - - - - - - - - - - - - - error in C2R 5 - - - - - - - - - - - - - - - - - - - -error in d1R

6 - - - - - - - - - - - - - - - - - ---- error in d2R 7- - - - - - - - - - - - - - - - - ----- error in d3R

Once the error location is pointed out, the corresponding bit is corrected by complementary bit(?) . The complete coder/decoder circuit is shown in Fig. 14).

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INFORMATION TECHNOLOGY, NETWORK AND INTERNET 4 data bits (d3 d2, d1, d0)

d3 d2 d1 d0 c2 c1 c0

3 check bits (c2, c1, co)

(7, 4) parity bits generator

Transmitter end

Transmitted bits Link

Receiver end

(7, 4) parity bits generator

Each block a two inputs XOR gate

6 5 4 2

Bit comparator

3 to 8 decoder

Corrected data

Fig. 14: (7,4) Hamming Code Generator Circuit

5.4.4 Hamming code (13,8) The (13,8) Hamming code is an one bit ECC. For a data byte D7 D6 D5 D4 D3 D2 D1 D0, the five check bits are generated as : C0 = D0 D1 D3 D4 C1 = D0 D2 D3 D5 D6 C2 = D1 D2 D4 D5 D7 C3 = D0 D1 D2 D6 D7 C4 = D3 D4 D5 D6 D7 Thus for a byte 11100011 , C0 = 0 C1 = 1 C2 = 1 C3 = 0 C4 = 1

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Assume that during transmission one bit of data byte say D5 becomes corrupted. So the received data byte will be 11000011 . Based on received data , at the receiver the check bits can be generated as : C0g = 0 C1g = 0 C2g = 0 C3g = 0 C4g = 0 Now compare the received check bits with generated check bits. On comparison we get a word known as error syndrome. In this case of example the error syndrome is : 0 1 1 0 1 (C0 C1 C2 C3 C4) 0 0 0 0 0 (C0g C1g C2g C3g C4g) ---------------------------------------------0 1 1 0 1 (E0 E1 E2 E3 E4) The error syndrome may be searched on a error syndrome Table (3). The match position will indicate the error location. In the case of considered example, the matching rightly points that D5 bit is in error . Table 3: Error Syndrome Error location

E0

E1

E2

E3

E4

Data bits

0 1 2 3 4 5 6 7

1 1 0 1 1 0 0 0

1 0 1 1 0 1 1 0

0 1 1 0 1 1 0 1

1 1 1 0 0 0 1 1

0 0 0 1 1 1 1 1

Check bits

0 1 2 3 4

1 0 0 0 0

0 1 0 0 0

0 0 1 0 0

0 0 0 1 0

0 0 0 0 1

The IC chip 74LS636 from Texas Instruments is a (13,8) ECC chip. The chip may be used for both generation of check bits and correction of error. Thus single chip may be used both for transmission and reception. It is a 20-pin IC as shown in Fig. (15).

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C0 — C4

SEF (Error) DEF (Flags)

D0 — D7 ^ ^ S0S1 (Function selector)

Fig. 15

S0 and S1 are used as function selectors. Function table 4 is shown below : Table 4 S1

S0

SEF

DEF

Memory Cycle

Functions

Data I/O

0

0

0

0

write

Generate check bits

Input data

Output check bits

0

1

0

0

read

Read data and check bits

Input data

Input check bits

1

0

enabled

read

Correct data Output and generate corrected error syndrome data

Output error syndrome

1

1

enabled

read

Latch and flag errors

Latched check bits

Latch data

Check bits I/O

The error flags operation table 5 is shown below : Table 5 Number of Errors 8 bit data 5 bit check

SEF

DEF

Remarks

0

0

0

0

Not Applicable

1

0

1

0

Correct data bit

0

1

1

0

Correct check bit

1

1

1

1

Interrupt

2

0

1

1

Interrupt

0

2

1

1

Interrupt

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5.4.5 General aspects of Parity Code The parity codes are guided by the following major rules : (a) The minimum number of check bits , C required to correct single, double . . . . . upto e-tuple bit errors is given by : ...(11) 2C >= ( e0n+c + e1n+c + ………….. + een + c ) where e represents the combinational factor and n is the number of information bits, that is to say that code is (n + c, n). (b) The capability of any code depends on the minimum Hamming Distance of any code. The weight of any code word is the number of ‘1’ (or YES state ) in the code. Smallest non-zero weight of the code words of any code is known as the minimum Hamming distance, dmin of the code. For a (7,4) code, we can refer to table 6 and see that dmin = 3. For a code to detect upto e bit errors per word, it is required that dmin >= (e + 1). Table 6 D3

D2

D1

D0

C2

C1

C0

Weight

0

0

0

0

0

0

0

0

0

0

0

1

0

1

1

3

0

0

1

0

1

1

0

3

0

0

1

1

1

0

1

4

0

1

0

0

1

1

1

4

0

1

0

1

1

0

0

3

0

1

1

0

0

0

1

3

0

1

1

1

0

1

0

4

1

0

0

0

1

0

1

3

1

0

0

1

1

1

0

4

1

0

1

0

0

1

1

4

1

0

1

1

0

0

0

3

1

1

0

0

0

1

0

3

1

1

0

1

0

0

1

4

1

1

1

0

1

0

0

4

1

1

1

1

1

1

1

7

Thus we can see that Minimum non-zero weight is 3 as evident from the table. Similarly, for correct up to e bits errors per code, dmin >= (2e + 1) ; while for correct up to e1 bit errors and detect up to e2 (e2 > e1) bit errors per code, dmin >= (e1 + e2 + 1). For a code to detect e2 errors and correct e1 errors we need d >= (2e1 + e2 + 1) where e2 > e1 is not required, dmin = 7

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Correct Detect 3 0 2 2 1 4 0 6 However, in general, for any (m,n) code, the coding rate or coding efficiency is defined as R = (n/ m ) × 100 %. ...(12) Coding rate measures the increased % bandwidth required for transmission of a code. For a good code R should be high. However from rule 1 as stated previously one can verify that R increases with n. This is the reason that in CCN/DCN, the higher data size is used.

5.4.6 Cyclic Redundancy Code (CRC) CRC is also a block code but it is a non-linear block code. However, all EDC/ECC ‘s can be classified as : EDC/ECC

Block codes

Liner block codes (7, 4), (13, 8) etc.

Convolution codes

Non-liner block codes (CRC-12, CRC-16, CRC-CCITT)etc.

The EDC that provides much more confidence to the users for detection of error is known as CRC (Cyclic Redundancy Code). The CRC is an extremely powerful error detection code and easily implemented using simple hardware; thereby, providing a cost - effective solution. The code and its circuit are based on a polynomial known as generator polynomial, which is mutually agreeable and known to both the sender and the receiver. The important four versions of generator polynomial commonly used in industry are: CRE–12 = x12 + x11 + x3 + x2 + 1 (Data bit is 6 and check bit is 6) CRC–16 = x16 + x15 + x2 + 1 (x0) CCITT–CRC = x10 + x12 + x5 + 1 (CRC–16 and CCITT–CRC are Used in US and Europe respectively for 8 bit data) CRC–32 = x32 + x26 + x23 + x22 + x16 + x12 + x11 + x10 + x8 + x7 + x5 + x4 x2 + x + 1 (Local Area Networks under IEEE 802 uses this code. One of the hottest LANs, Ethernet uses this CRC 32). Check bits generating Instead of going into theory, we can justify that there is a quite simple circuit which can be used to generate check bits of CRC instantaneously based on serial data input. CRC-16 may be taken up, for example. The generating circuit for check bits is shown in Fig. (2). The location for feedback taps of exclusive OR gate can be simple determined. Subtract each power of x of

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generator polynomial from the number of location of shift registers (which equals number of check bits). The results of all such subtraction (in the example; 16 – 16 = 0, 16 – 15 = 1, 16–1 = 14, 16 – 0 = 16) are the tap locations. Before sending any data, the shift register is to be initialized to all zero. After transmission of data bits using shift, the content of the shift registers will be check bits. You now simply transmit these check bits. At the receiver, same circuit (fig16) will generate the check bits based on received data bits. The difference in any or more bit positions in between the generated check bits at the receiver and the received check bits at the receiver will detect an error. In addition to the simply circuit and lower cost, CRC offers another improvement over parity. For generation of check bits, we do not have to wait for whole of the message data. It is done serially and instantaneously. This is not true in case of systematic parity, the systematic parity has the effect of delaying the message more than CRC. The efficiency of any code is measured by a parameter known as code rate (Cr): Efficiency (Cµ) =

Message size Code size

No way inferior If 256 bytes of data (equals to 2K or 2048 bits) is sent under 32-bits CRC (a typical case used in Ethernet) the efficiency of CRC becomes equals to 98.5 percent. Lowest efficiency in Ethernet scheme is 92 percent. A parity bit per bytes is usually used for systematic parity code. Thus, the efficiency of this simple parity code is usually 89 percent. Thus, in terms of efficiency CRC is no way inferior to the simple parity. The above example also illustrates that in case of CRC while a total of 2080 bits are to be exchanged per 256 bytes in case of simple parity the figure is 2304 (= 2048 data + 256 parity bits) per 256 bytes. This is how transmission time is less in CRC. However, CRC is not free from limitation. 16 bits check can only have 216 = 65536 unique words. Therefore, there may be some extremely rare combination of errors to fool the users. However, the above noted four generator polynomial, G(x) can detect early : • All single bit errors (typically 100 percent error detection). • All double bit errors so long G(x) is a factor with at least three terms (typically 99.8 percent error detection). • Any odd number of errors so long G(x) contains a factor of (x + 1) (typically 99 percent error detection). • Any burst error whose length is less than the number of check bits and most of the larger burst errors.

XOR gate

XOR gate Shift register 1

2

3

4

5

6

7

8

9

10 11 12 13 14 15 16

XOR gate Message data input

Fig. 16 : CRC check bit after transmission of message data.

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Till today more attention was given on single-bit error correction code rather than multiple-bit error correction code. The reasons are many: If the bit error rate of the channel is 10-4, the probability of the single-bit error and the double-bit error per code word of size 7 bit are respectively 7 × 10–4 and 21 × 108. This means that multiple bit error is much less probable than the single bit error.

5.4.7 Basic BEC techniques BEC has three basic techniques. These are: Stop-and-Wait Automatic Repeat Request (S/W), Go-Back-N Automatic Repeat Request (GBN) and Selective Repeat Request (SRQ). In S/W ARQ technique, after transmitting a packet, (a packet is a code appended at both end with flags of start and end, source address and destination addresses and other control bytes), the transmitter waits for an acknowledgement from the receiver before transmitting the next packet. On receiving a packet, the receiver checks, using the error detection technique used in the process, for any error. If no error is found, the receiver sends an acknowledgement, known as positive acknowledgement (ACK), to the transmitter through the feedback path. On the other hand, if any error is detected, the receiver sends a negative acknowledgement (NAK) to the transmitter. On receiving an ACK for the already transmitted packet, the transmitter transmits the next packet. But on receiving a NAK for any transmitted packet, the transmitter retransmits the previously sent packet. In short, until and unless a packet is received correctly by the receiver and it is positively acknowledged, the transmitter will not transmit the next packet. However there remain several questions to such operations. What will happen if ACK or NAK is lost in the feedback path? The transmitter waits for a period known as time out period, which is greater than twice the propagation delay between the transmitter and the receiver, for the acknowledgement. If no acknowledgement is received within the time out period, the transmitter retransmits the previous packet. The receiver understands the received packet as retransmitted one by checking the sequence number of the packet and takes decision accordingly. What shall happen if ACK is changed to NAK or vice-versa during the transmission through the feedback path? The change of ACK to NAK is tackled by the same technique, as that is used in case of loss of acknowledgement. When NAK is changed to ACK, the receiver on checking sequence number only detects the change. By this time the previous transmitted packet for which NAK was changed to ACK is not available with the transmitter. This causes a serious problem. The performance of the techniques is measured by a parameter known as throughput efficiency (n). It is defined as number of the information bits correctly transmitted divided by the total number of bits transmitted for the purpose. If we assume (i) (m, n) code were used in the protocol. (ii) processing time at the transmitter and the receiver for ACK/NAK or packet is negligible, (iii) transmission time of ACK/NAK is negligible and (iv) feedback path is error free; ν(s/w) = n/{(m + RT)E} ...(13) where E = expected number of transmission for successful reception of a packet, R = rate of transmission, T = total round trip delay. When each packet has the same probability that it is received with error, E = 1/β ...(14) where β is the probability that a transmission for a given packet is the last transmission. If P and Pu are the probability that a packet is in error and the probability of the undetected packet error respectively,

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β = 1 – P – Pu = 1 – P, as Pu << P …(15) If tp is one way propagation time and tt is the transmit time of a packet, we have: T = 2tp and R = m/tt Using these and eqns. (14-15) in eqn. (13) we find: ν(s/w) = {n(1 – P)}/{m(1+2a)} …(16) where a = tp/tt. The throughput efficiency of S/W ARQ is poor. It is because the successful transmission of a packet involves at least two propagation delays in between the transmitter and the receiver. In order to improve throughput, GBN ARQ was developed. In GBN ARQ technique, the transmitter continuously transmits a block of N (N is often known as window size) packets without waiting for the acknowledgement for the individual packet; and keeps the packets in its memory or buffer. The receiver sends only the negative acknowledgement if so detected. The transmitter on receiving NAK for the first time, stops transmission and retransmits all the packets which were transmitted prior to stopping of transmission but starting from the packet for which NAK is received; and discards the packets transmitted prior to the packet in error from the memory. For example when 5th (assuming N > 5) packet of the block is the first negatively acknowledged packet when up to Nth packet has been transmitted, the transmitter will then discard first to fourth packets from its memory, and now will retransmit all the packets from 5th to Nth. Worst situation in GBN ARQ occurs, when the first packet of the block is negatively acknowledged, the case of when the whole block of N packets requires retransmission. Best situation occurs when none of the packet is negatively acknowledged, thereby successful transmission of N packets involves with minimum two propagation delay rather one packet being involved with two-propagation time as in S/W ARQ. This gives throughput advantage to the GBN ARQ over S/W ARQ. GBN ARQ may be two types: continuous and non continuous. In the continuous scheme, after transmission of a block of N packets, the transmitter does not have to wait for the acknowledgements of these packets before starting the transmission of the next block. In the non-continuous mode, before starting the transmission of the next block, the transmitter has to wait for the acknowledgements for the packets of the previous block. If the transmit time of a packet/acknowledgement is one unit, we have: N >= (1 + 2a) for the continuous scheme and N < (1 + 2a) for the non-continuous scheme. The throughput efficiency for GBN ARQ is given as: ν(gbn) = {n(1 – P)}/{m(1 + 2aP)} for continuous scheme, = (n/m){1 + NP/(1 – P)}–1 where N=1+T/(m/R)=1+2a, [Note that m/r is chosen so as to make N = 2,3,4…. of GBN technique. When T = m/R, N = 2 and transmitter goes back by two blocks.] ν(gbn) = {n . N(1 – P)}/{m(1+2a)(1 – P + NP)} for non-continuous scheme …(17) The through put of GBN ARQ technique is higher than that of the S/W ARQ but still the throughput is a function of propagation delay, a. Selective Repeat Request (SRQ) ARQ further improves the throughput. It operates like that of the GBN ARQ but retransmits only the packet for which negative acknowledgement is received. This means that theoretically infinite buffer is required at the transmitter. It has also two modes of operation, namely continuous and non-continuous.

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The throughput efficiency is given as: ν(srq) = {n(1 – P)}/m for continuous scheme ν(srq) = {n . N(1 – P)}/{m(1 + 2a)} for non-continuous scheme ...(18) The problems of the loss and/or the change of acknowledgements in GBN ARQ and in SRQ ARQ are tackled by the same techniques as in S/W ARQ. It is the trade off between buffer size and throughput that plays the role among the three basic BEC schemes, S/W ARQ, GBN. ARQ and SRQ ARQ. A comparison in terms of buffer size or memory requirements and throughput efficiency of the schemes is shown in Table (7). When the parameter “a” is zero i.e. tp<
Theoretically Minimum at transmitter

Memory Space Requirement at receiver

Throughput efficiency

Stop and Wait

1

2

Go-Back-N

N

0

Higher than the stop and wait

Selective Repeat Request

∞

0

Higher than both the stop and wait; and the go-back-n.

Low

Another way of looking into the throughput of ARQs: In order to compare three techniques of BEC strategy, it is assumed that p is the probability that a code word is received incorrectly at the destination. It is also assumed that the code can detect any type of errors (that is, all sorts of errors are detectable). Stop and Wait ARQ: One successful transmission of a word requires a total time Tt which is: Tt = tp + tt + tps + tp + ttk + tps where tp = propagation time of link between source and destination. tt = transmission time of code. tps = processing time at source/destination. ttk = transmission time of ACK/NAK. Usually, tps and ttk are small enough to be cancelled so that : Tt = tt + 2tp = tt (1 + 2a) where a = tp/tt, are design parameters of the system. We have here assumed that timeout = 2tp. In case of error, average number of attempt Nr required to transmit a code successfully is given by : Nr = 1 . (1 – p) + 2 . p (1 – p) + 3 . p2 (1 – p) + ........... = Summation of i pi –1 (1 – p) where i = 1 to infinity. = 1/(1 – p) as because, probability of i attempts is pi – 1 (1 – p) where pi – 1 represents probability of failures of (i – 1) attempts and (1 – p) probability of success.

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Therefore, average time required to successfully transmit the code is : Tr = Tt/(1 – p) = tt(1 + 2a)/(1 – p) Link utilization, U defined as transmission time of code divided by actual average time taken by the code for successful transmission under a scheme is : U = tt/Tr = (1 – p)/(1 + 2a). If this code is (m, n); then in case of transmission of uncoded system, the transmit time of the code, ttu could have been : Ttu = n/m . tt. Throughput efficiency (nt) of error control strategy is defined as : nt = ttu/Tr = n(1 – p)/m(1 + 2a) = (n/m) U Go-back-N-ARQ : Here two possibilities are there as discussed earlier : Case I when N > (1 + 2a) and Case II when N < (1 + 2a). Case I : In this case the average transmit time for successful transmission of a code is given by : Tr = tt + Σ i pi (1 – p) Tt = tt + {p/(1 – p)}.Tt = tt [1 + {p(1 + 2a)}/(1 – p)] = tt (1 + 2ap)/(1 – p). Link Utilization is therefore, U = (1 – p)/(1 + 2ap). Throughput efficiency is : nt = (n/m).{(1 – p)/(1 + 2ap)} = (n/m).U Case II : In this case, average number of attempts required per successful transmission of a code : Nr = 1/N [1.(1 – p) + (N + 1).p.(1 – p) + (2N + 1).p2.(1 – p) + ………..] = 1/N [1 + (N . p)/(1 – p)] Therefore, Tr = (Tt/N)[1 + N.p/(1 – p)] = {tt(1 + 2a)(1 – p + Np)}/{N.(1 – p)} and hence, U = N(1 – p)/(1 + 2a)(1 – p + Np) nt = (n/m).{N(1 – p)/(1 + 2a)(1 – p + Np)} = (n/m).U Selective Repeat As for Go-back-N-ARQ, for selective repeat technique there are also similar two cases. Case I : When N > (1 + 2a) : Here calculation of Tr is straightforward as below : Tr = Tt/(1 – p) But Tt = tt as propagation over large n when averaged get zero [that is, Tt = (Ntt + k.2tp)/N where k is number of error out of N. Value of k is very small compared to N]. So, Tr = tt/(1 – p) U = (1 – p) and nt = (n/m).(1 – p) = (n/m).U

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Case II : When N < ! (1 + 2a) : In this case, average number of transmission required for successful transmission of a code is : Nr = 1/N . Summation of {i . pi–1 (1 – p)} where i = 0 to infinity = 1/N(1 – p) and Tt = tt(1 + 2a) Thus, Tr = tt (1 + 2a)/N(1 – p) U = N(1 – p)/(1 + 2a) nt = (n/m).{N(1 – p)/(1 + 2a)} = (n/m).U Observations 1. When N < (1 + 2a), in all the cases it is seen that U and hence nt depend on N. Therefore the cases of N < (1 + 2a) are avoided in design. 2. When N = 1, as predicted go-back-N and selective repeat request become identical to stop and wait ARQ. 3. U and nt are practically measure same thing and directly related to each other through coding rate. For a fixed U, increased coding rate increases throughput (why?). Parameter N is known as window.

SOLVED PROBLEMS 1. For a communication system, a = 10–4 when Eb/N0 = 8.4 db. If data rate is 4800 bps, what is the required received signal level ? Assume effective noise temperature is equal to the room temperature. Solution. We know, N0 = kT where k = Boltzmann’s constant = 1.3803 × 10–23 J/oK T = temperature in oK = 290 oK as given. Thus : Where

Eb/N0 = S.Tb/N0 = (S/C)/kT Tb = bit duration = 1/C where C = bps.

In decibel notation, (Eb/N0)db = Sdbw – 10log10C – 10log10k – 10log10T or or

8.4 = Sdbw – 10log10 (4800) + 228.6dbw – 10log10(290) 8.4 = Sdbw – 36.81 + 228.6 – 24.62

or Sdbw = – 158.766. 2. A 4-bit message is transmitted via FSK and bit energy to noise density ratio (p) at the receiver is 12.32 db. (α =

1 2

e–p/2 )

(a) Compute the probability of single bit error rate. (b) If single parity is encoded, what is the probability of single bit error rate ? Assume equal transmitted power. Solution.

α= =

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e–(17.06)

10–4

since 12.32 db = 17.06

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(a) Without coding : P1 = 4C1.10–4.(1 – 10–4)3 = 4 × 10–4 (b) With coding : As equal transmitted power is to be maintained (Eb/N0) is to be changed. (Eb/N0)new = (S/Cnew)/N0 = S/N0 . 4/5 . 1/C [Cnew = 5/4C] = Eb/N0 . 4/5 ∴ αnew = 1/2 . e–(17.06/2 × 4/5) = 0.5 × 10–3 Hence P1 = 5C1 . 5 × 10–4 . (1 – 5 × 10–4)4 = 25 × 10–4. 3. In the problem (2) it is seen that due to coding P1 is increasing. Then how does coding offer benefits. Solution. Due to coding all single bit errors may be detected, which is not the case without coding. In problem (2) if we transmit 25 x 104 bits while in case of no-coding there will be (25/4) = 6 bits undetected one bit error ; in case of coding there will be no undetected error. 4. Calculate the probability of message or word error for problem (2) which can not be detected. Solution. (a) Without coding : Pw = 1 – (1– α)4 = 1 – (1 – 4 × 10–4)4 = 1.6 × 10–3. (b) With coding : Only single bit error can be detected and others cannot be detected. So, Pw = Σ[5Ci (α)newi (1 – αnew)5 –‘i’ ] where ‘i’ = 2…..5 = 10.25 × 10–8 = 2.5 × 10–6 This is the answer to the problem (3) also. 5. A brute code is repetition code. For example in triple-repetition code 0 is coded as 000 while 1 is coded as 111. (a) Find the probability of undetected word error in case of detection only (b) Assume majority rule(that is you are using a majority gate for detection) for correction. Find the probability of undetected error that would result. Solution. (a) For detection only, any word other than 000 or 111 is a detected error. Single and double errors like 001 or 011 (when 000 was transmitted) are detectable. But triple errors like 111(when 000 was transmitted) are undetectable. Hence, Pw = 3C3 . (α)3 . (1 – α)0 = (α)3 where α = BER . (b) Majority rule means assumption that at least two bits out of three received bits are correct. Thus 110 and 001 may be corrected as 111 and 000 respectively. This rule may correct words with single error but double or triple errors would result undetected error.

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Thus undetected error probability : Pw = 3C2 . (α)2 . (1 – α) + 3C3 . (α)3 . (1– α)0 = 3(α)2 (1 – α) + (α)3. 6. In general, a repetition code may be expressed as (2t1+ t2, 1) code. For a double repetition code t2 = 0 and t1 = 1; For a triple repetition code t2 = t1 = 1. However as for even numbered repetition, majority rule is not applicable; it is often taken as that generalized repetition code by (2t + 1, 1). For this code, (a) Plot word correction probability v/s t as well as (b) Plot code efficiency v/s t. Comment on the plots. Assume (i) α = 10 – 4 when t = 0; (ii) α = 0.5 exp (– p/2) . Solution. α = 10–4 that is p = 17 Therefore, (αt = 0.5) exp(– 17/2 . 1/2 t + 1) (a) (2t + 1, 1) can correct up to t-bit errors probability of which is : PC = Summation of [2t + 1Ci . (αt)i . (1 – αt)2t + 1 – i ] where ‘i’ = 0……t Now why ‘i’ = 0, as no error is also corrected. Based on this equation we can plot PC v/s t. (b) The code efficiency ---- nC = 100/(2t + 1) % Based on this equation we can plot nC v/s t. 7. In a data communication system, on average 8 hrs. is used in for full transmission. Find the average daily likely range of bit errors. Line is 2400 bps. α = 10–4 Solution. Average daily transmission of bits over syste = 2400 × 3600 × 8. = 6.912 × 107 Mean = 6.912 × 107 × 10-4 = 6.912 × 103 Variance = 6.912 × 107 × 10–4 (1 – 10–4) = 6.911 × 103 Likely range is Mean ± 2(Var)1/2 = 6.912 × 103 ± 2 × 83.13 = 7078 to 6746 Thus we can expect up to 7078 to 6746 erroneous bits out of transmitted 6.912 × 107 bits. 8. The BCH (Bose, Choudheri, Hocquenghem) codes are for MBR correction. One (m, n) BCH code can correct up to e-tupple bit error where : e = (m – n)/p where p is the integer being related to m as m = 2p – 1. The code has minimum distance as : (2e + 1) <= dmin <= (2e + 2) (a) Prove that Hamming code (7,4) is a special code of BCH code. (b) The Golay code is a (23,12) code ( although there is another Golay code known as Extended Golay code. Extended Golay code is (24,12) code with dmin = 8). It has dmin = 7.This is only code of length 23 which can correct up to 3-bit error per word. Compare BCH code with Golay code.

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Solution. (a) This can be proved if we can find (7,4) BCH code for e = 1 as well as if we find dmin = 3 for (7,4) BCH code. For BCH code : E = (m – n)/p Therefore, for this case e = 1, n = 4 and hence : 1 = (m – 4)/p or 1 = (2p – 1 – 4)/p as m = 2p – 1 The integer value of p = 3 only satisfies the above equation. Hence m = 7. This one bit BCH ECC is a (7,4) code. From dmin of BCH code we see in this case (e = 1) ; 3 <= dmin <= 3 Therefore dmin = 3 Thus it is proved that (7,4) Hamming code is a special code of BCH code. (b) In order to compare, we assume e = 3. The required BCH code is (31,16) and (63,45) Comparison

Code Efficiency/Rate

Code capability (e/m)

(31,16) BCH

16/31 × 100 = 50 %

3/31 × 100 = 10 %

(23,12) BCH

12/23 × 100 = 50 %

3/23 × 100 = 13 %

(63,45) BCH

45/63 × 100 = 50 %

3/63 × 100 = 5 %

Conclusion : (1) (23,12) Golay is more capable than (31,16) BCH (2) (63,45) BCH is more efficient than both (31,16) BCH and (23,12) Golay, but less capable. (3) There is trade-off of efficiency v/s capability in between BCH code. 9. Data is being sent over a 20 km. line at T1 rate of Japan System using BCH (1023,973) code. If p = 0.5, find throughput efficiency for (a) simple ARQ (b) go-back-N ARQ (N >= (1+2a)) and (c) selective ARQ (N >= (1+2a)). Assume that the velocity of light of the medium as (C0/10). Solution. Japan System T1 carrier relates to 1.544 Mbps. tp = propagation time = 20 km./(C0/10)

= (20 × 103)/(3 × 108/10) sec. = 0.66 msec.

tt = transmit time for a frame (code of 1023 bits) = 1023/(1.544 × 106) = 0.66 msec. a = tp/tt = 1.

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(a) Throughput efficiency for (m, n) code under ARQ is : nt = n.(1 – p)/m.(1+2a) = 973 × (1 – 0.5)/1023 × (1 + 2 × 1) = 0.158 =15.8 % (b) Throughput efficiency for (m, n) code under go-back-N (N >=1 + 2a) is : nt = n . (1 – p)/m.(1 + 2ap) = 973 × (1 – 0.5)/1023 × (1 + 2 × 1 × 0.5) = 23.77 % c) Throughput efficiency for (m, n) code under selective repeat request is : nt = n . (1 – p)/m = 973 × (1 – 0.5)/1023 = 47.55 % 10. Compare the different ARQ technique based on throughput efficiency and memory etc. and comment. Solution. Parameters

Simple ARQ

Go-back-N N >= (1 + 2a)

Go-back-N N <= (1 + 2a)

Maximum throughput

n/ m(1 + 2a)

n/ m

n.N/m(1 + 2a)

n/m

Nil

N

More than N

Much more than N. Basically it’s very large.

Efficiency Memory required

Selective ARQ

Comment : (1) Go-back-n with N < (1 + 2a) is not used as it depends on N. (2) Go-back-N is having maximum efficiency with selective repeat request but less memory. Thus it is used mostly. 11. The Reed-Solomon or R-S code is based on the group of bits known as symbol. It is a symbol correcting code. A (m, n) R-S code is defined as : n = number of information symbols m – n = number of check symbols where ; (1) Each coded symbol is of p-bit sequence that is p-bit symbol that is m = 2p – 1 (2) Code can correct up to e-symbols when e = (m – n)/2 (a) Find a (R-S) code when p = 8 and e = 4. b) Find the capability of code of (a) (c) If the code is coupled with interleaving techniques of burst error correction, find the capability of the code. Solution. (a) When p = 8, we get m = 28 – 1 = 255. As e = 4, we get e = (m – n)/2 or 4 = (255 – n)/2 or n = 247. Thus the code is (255, 247).

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(b) (i) Code Rate Efficiency = 247/255 = 96.86 % (ii) Code can correct up to 4 bit symbols that is, a burst of length = e . p = 4 × 8 = 32 bits. (iii) If there is at least one random error per code word symbol, it can correct 4(=e) bits errors out of 255 × 8 (=p) = 2040 bits. Thus the code is not efficient for random error correction. (c) Say we are using interleaving technique with (m1, n1) Hamming code such that : 2( m1 – n1) >= Summation of Cim1 where i = 0 to e Moreover there are q bits per column. Then interleaving technique as discussed can correct burst error of length L where L <= q . e Thus if the interleaving is merged with R – S code : Number of correctable symbols = q . e Number of correctable bits = q . e . p Thus if q = 10 in interleaving technique then for R – S (255,247) code : Number of correctable bits = 10 . 4 . 8 = 320 bits. 12. Data is being transmitted over a link at 9600 bps using (1023 , 973) BCH code. If a = 10–4 and velocity of propagation of signal in link is 2 × 108 m/sec, then find throughput efficiencies for the following cases : Case I. Data link = a twisted pair of cable of length 1 km. The error technique is ARQ. Case II. Data link = a satellite link of 25000 km. The error technique is go-back-N ARQ with window = 2. Case III. Data link = a satellite link of 50000 km. The error technique is selective repeat request with window = 7. Solution. As α = 10–4, then (α)′ = 1023/973 × 10–4 where α′ refers to BER with higher size of code with check bits Hence, p = 1 – (1 – α′)1023 = 0.1019 = 0.1 tt = 1023/9600 = 0.1065 sec. CASE I. tp = 103/(2 × 108) = 5 × 10–6 sec. a = (5 × 10–6)/0.1065 = 46.95 × 10–6 sec. nt = n(1 – p)/m(1 + 2a) = 973(1 – 0.1)/1023(1 + 46.95 × 10–6) = 0.8559 = 85.6 % CASE II. tp = (25000 x 103)/2 × 108 = 0.125 sec. a = 0.125/0.1065 = 1.173 Now, (1 + 2a) = 1 + 2 × 1.173 = 3.346 So, in this case when N < (1 + 2a) as given N = 2. Hence : nt = n/m . {N(1 – p)}/{(1 + 2a)(1 – p + Np)} = 973/1023 . {2(1 – 0.1)}/{(3.346)(1 – 0.1 + 2 × 0.1)} = 0.4651 = 46.51%

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tp = 50,000 × 103/2 × 108 = 0.25 sec. a = 0.25/0.1065 = 2.347 Therefore, (1 + 2a) = 5.69 This is the case when N > (1 + 2a) as N = 7. Hence, nt = n/m(1 – p) = 973/1023 × (1 – 0.1) = 0.856 = 85.6% Example 1. Binary data is being sent over a channel where BER = 0.1. If bits are independent, what is the probability that a transmitted nibble “1110” is received as “0111”. Solution. Probability = P10 × P11 × P11 × P01 = 0.1 × (1 – 0.1) × (1 – 0.1) × 0.1 = 0.1 × 0.9 × 0.9 × 0.1 where (1 – 0.1) represents the probability that the bit is not in error. = 8.1 × 10–3. Example 2. Data is being sent at 10 MBPS over a link whose propagation delay is 10 micro sec. If the probability that the frame in error is zero, find the size of frame that will give 50 % link utilization for IARQ technique. Solution. Link utilization = (1 – p)/(1 + 2a) = 1/(1 + 2a) × 100 % or 50 = 100/(1 + 2.100/N) or N = 200 bits [since a = tp/tt = 10 × 10–t/(N/10 × 106)] If Pe = 0.1, then 0.1 = 1 – (1 – 0.001)N or 0.999N = 0.9 CASE III.

or If Pe = 0.5, then 0

N = (ln 0.9)/(ln 0.999) = 105.3 .999N = 0.5

or N = (ln 0.5)/(ln 0.999) = 692.869 Example. Data is to be transmitted between a source and destination in the following cases : (a) 1000 meter twisted pair wire having a transmission rate of 1200 bps. (b) 10 km of co-axial cable at 1 MBPS (c) 72000 km of satellite link at 10 MBPS Assume velocity of signal is 2 × 108 m/sec. Determine in each case how many bits, source shall transmit before the first bit arrives at the destination. Why is it so. Solution. CASE A. Tp = 100/(2 × 108) = 5 × 10–7 sec. The first bit shall reach receiver after Tp time from its transmission instant. So by this time, number of bits transmitted is 5 × 10–7 × 1200 = 6 × 10–4 = 0. CASE B.

Tp = (10 × 103)/(2 × 108) = 5 × 10–5 Number of bits = 5 × 10–5 × 106 = 50 bits

CASE C.

Tp = (72000 × 103)/(2 × 108) = 36 × 10–2 Number of bits = 36 × 10–2 × 10 × 106 = 360000 bits.

This is due to Tp . Reasons,

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In the above problem, we have used relations derived on the following assumptions : Except propagation and data transfer time, all other such as acknowledgement signal transfer, processing time, modem turn-around time etc. are zero. But if acknowledgement is of 16 bits, modem turn-around time is 75msec., what shall be the link utilization in case of Half Duplex and Full Duplex. In case of Half Duplex, U = tt(1 – p)/{tt + 2(tp + tmodem turn-around + tack)} = 0.1065(1 – 0.5)/{ 0.1065 + 2(5 × 10–6 + 75 × 10–3 + 16/3600)} In case of Full Duplex : U = tt (1 – p)/{tt + 2(tp + tack)} as modem turn-around will be zero. BOX 6 Error Control in Networks A Short Questions 1. What do you mean by bit error? When a transmitted bit is received erroneously at the receiver, bit error is said to have caused. If the data is binary, bit error occurs if receiver detects a “1”, when transmitter has transmitted actually a “0” and vice-versa. 2. Who causes bit error? Different types of noises are responsible for bit error. They include Channel noise like Gaussian noise, system noise like thermal noise, fading, lighting, man-made noise produced by making On and Off of heavy electrical machines. 3. What are the different types of but error and who cause them? Different types of bit error are : random bit error and burst error. Burst error is also known as correlated error. Random error occurs when errors are distributed randomly over the message or word. Following is an example of a random error: Transmitted bytes : 11001110 Received bytes with random error at third and eight positions from left : 11101111. Random error can be again of different types: SBR (Single bit error), DEBR (Double bit error)… and MBR (Multiple Bit Error). SBR, DBR and NBR respectively mean one bit in error per word, two random buts are in error per word, and multiple random bits are in error per word. The earlier example of random error is an example of DBR. Burst error means a sequence of adjacent bits of the message or the word are in error. Error are clustered together in the message or the word. The following is an example of burst error : Transmitted bytes : 11001110 Received bytes with burst error of burst length 3 : 11010010 (4th, 5th, and 6th bits from left are in error). Random error is usually caused by channel noise like Gaussian white noise. Burst error is caused mainly by system’s component failure, on-off switching of heavy electrical machines like motors, impulse noise produced by lightening, failure of radio transmission system due to rapid fading. But causes of burst may also cause random error and vice-versa.

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4. Define BER (Bit Error Rate) or bit error probability? Bit Error Rate or probability is defined as : BER = Lim ( η / N) N→α

where η is the number of bit in error out of total transmitted N bits. 5. If BER = 10–3, how many bits shall be in error out of the total transmitted bits of 106? BER = 10–3 = 1/103, which in turn means that on average there shall be 1 bit in error out of 103 transmitted bits. Therefore for a total transmitted bits of 106, there shall be 103 bits in error. The generalized formula for calculation expected number of bits in error out of a given number of transmitted bits and for a given BER is : Bits in error = BER × Total Number of transmitted bits. 6. What is the controlling parameter of bit error? Does bit error rate depend on the data or the bit transmission rate? Eb/No which means that the ratio of energy per bit to noise density of the channel is the controlling parameter of bit error. As this ratio increases, bit error rate decreases. For given single power, S, we have Eb = S.Tb = S/C where Tb is the bit duration and C is the transmission bit rate. For a given signal power and given channel, as C increases, Eb/No decreases and this in turn increases bit error rate. We can also explain the same physically. As data transmission rate (C) increases, the spacing between consecutive data decreases, that is overlapping of spectrum of data becomes more. This causes more error. 7. If BER = 10–4 and bits are being transmitted at a rate of 1 MBPS, what shall be the average bit error per second ? Average bit error per second = BER × bit transmission rate = 10–4 × 1MBPS = 100 bits per second. 8. If the average packet size is 2048 bytes per packet and BER = 10–4, what is average bits in error per packet? Average bits in error per packet = BER × Average packet size in bits = 10–4 × (2048 × 8) = 1.638 per packet. 9. What are the different classes of probabilities associated with a received frame in terms of error? What is the probability that a frame is in error? There are three distinct probabilities : (1) frame received without any error (P1), (2) frame is received with detected/corrected one or more bit errors, but with no undetected bit errors (P2), (3) frame is received with one or more undetected bit errors (P3). Here P1 + P2 + P3 = 1 The total probability that a frame is in error = P2 + P3 = 1 – P1 10. What are the different probabilities of question (9) in terms of BER if no means are taken to detect/correct error? Assume that BER is constant and independent of bit position. Frame size is of N bits. Examine the result. P1 = (1 – BER)N ; P2 = 0; P3 = 1 – P1 We see that as frame size (N) increases, probability of receiving frame without any error (P1) decreases; thereby causing the probability of receiving frame with undetected error (P3) to increase.

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11. What is the probability, whether you use any coding or not, in terms of BER, that a frame is in error if the frame size is of N bits? Probability = 1 – (1 – BER)N . But BER of a coded system is slightly higher than that of the uncoded system. 12. Pe is the BER of binary data transmission, what shall be the BER of S-array data transmission? Comment on your answer. BER for S-array data transmission shall be approximately S times of Pe, i.e. SPe. Binary data transmission is less error prone. 13. What is the common unit of Eb/No? It is db. 14. What is appropriate probability model of bit error? Why? Give the model. The most appropriate prob+ability model of bit error from data/computer communication and networking point of view is binomial probability model. Data errors are discrete in nature. Two discrete probability functions are : binomial and Poisson. Discrete variable may be real as well integer in value. In bit error, error must be discrete and integer in value like 02 bits in error and not like ¹.4 bits in error. Binomial function deals with integer valued discrete function. Therefore it is appropriate model of bit error. Probability that there is i errors in a code or message of size m (i <= m) is given by binomial frequency function : P(i, m) = mCI αi (1 – α) m – i m where α = BER and CI = m !/(m – i) ! i ! 15. What is the relative spread of error under proposed model? Is it dependent on m? Relative spread is defined as Var/mean. In case of binomial model mean = mα and Var = mα (1 – α). Thus relative spread = 1 – α/mα Yes, it is dependent. From the equation of relative spread it is seen that for a given, spread decreases with increases of m. The likely value of errors is near mean as m is large. This is one of the reason to have large code size or word size. 16. How the error correction or error detection code is made of? Error correction or detection code is made of using the philosophy of “redundancy increases reliability”. Some redundant bits known as check bits are added with the given message bits to make a code. These check bits are used to detect or correct error. If the message is of n bits, a code is made of m (m > n) bits where number of check bits, c = m – n. The code is then called (m, n) code. 17. Classify different types of error correction/ error detection codes with example. Codes can be classify into basic two classes, viz, block codes, and convolution codes. Block codes again can be of two types: linear block code and non-linear block codes. Examples of linear codes are : Parity codes like Systematic and non-systematic parity codes. Hamming (7, 4) single bit error correction codes, Hamming (13, 8) single bit error correction code, Golay codes, BCH codes etc. Examples of non-linear codes are CRCs (Cyclic Redundancy Codes) like CRC-12, CRC-16, CRC-CCITT etc. 18. How the different error correction/detection codes are related to different types of bit errors? Linear block codes are usually used for correcting/detecting random bit errors. They can be used to correct/detect burst error, but in that case either the technique used (example : interleaving technique) shall be quite complex or the code be less efficient.

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Convolution code is used when making blocks out of a message is difficult, viz, satellite communication. 19. How to find the number of check bits of any linear block or parity code? The minimum number of check bits© required to correct dingle, double …. Upto e-tupple bit errors per code is given by :L e

2c > =

∑

n+c

Ci

i=0

Where n + cCi = (n + c) !/(n + c – i) ! i ! and n is the message size. Therefore if n = 4, to correct upto only single bit error per code minimum required check bits, c shall be 3. That is why one single bit error correction code is Hamming (7, 4) code, 7 is coming in adding message size (4) with number (3) of check bits. 20. Why do we always use minimum check bits? As check bits increase, bandwidth required to transfer message increases. The amount (IBW) by which the required band-width increases is directly proportional to the number of check bits: IBW = Required increases band-width – Original required band-width = [BW (n + c)/n] – BW. = BW (c/n). Where BW is the bandwidth required originally to transfer message of n bits without coding, and c is the check bits used in coding. 21. What is coding rate? For any code (m, n) , coding rate or coding efficiency is defined as: Coding rate = n/m = n/n + c where c is the check bit (i . E . m = n + c). Coding rate or coding efficiency is often specified in % and in that case it is (n/m) × 100. Coding rate or coding efficiency is related to the increases bandwidth required to transfer a message with coding that we discussed in question (20). Required increases bandwidth is inversely proportional to coding rate : Required increased bandwidth = BW (1/coding rate). 22. Is there any relation between coding rate and code capability? Yes, there is a relation between them. As check bits increases, the capability of the code increases. But with increases of check bits, the coding rate decreases which in turn requires increased bandwidth for transmission of the code… However, we can illustrate this feature with an example. Say we are using repetition code, (2t + 1, 1). In this case, probability of bit error shall be : 2t + 1

Pe =

∑

2t + 1

C i a i (1 − α) (2 t + 1 − i)

i=t+1

Using this, we can have following table: Code rate (1/2t + 1)

Probability of codeword in error (BER = 10–2)

1 1/3 1/5 1/7

10–2 3 × 10–4 10–6 4 × 10–7

1/9

10–8

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23.

24.

25.

26. 27.

28.

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As number of check bits increases, coding rate decreases but improvement in error performance increases. Why do we prefer to higher block size for coding? We have already defined coding rate or coding efficiency. This rate or efficiency can tend to 1 or 100% only when n tends to infinite (as n tends to infinite, n/(n + c) tends to 1). This is the reason that we prefer to higher size of message or packet or data block. But there is a trade-off. As we increases n, probability that a frame is in error increases. What is the relation among uncoded bit rate, code rate and coded bit rate? Such relation exists if we assume that codeword with m bits must be transmitted within the time required to transmit message with n bits. If Tb and Tc are respectively the bit duration in case uncoded and coded cases, it is required that : mTb = nTc : or fb/fc = n/m or fb/fc = coding rate, where : fb = 1/Tb, is known as uncoded bit rate, and fc = 1/Tc, is known as coded bit rate. What is throughput of a code? It is the probability of code acceptance. Probability of code acceptance is nothing but the probability that a code word is received with no error. If BER is the bit error rate when (m, n) code is being used, the throughput of the code is (1 – BER)m provided bit error is independent of bit position. How can you increase the throughput of a code? One solution is decreasing size of code i.e. decreasing m or n of any (m, n) code. Is the solution suggested in question (26) feasible? Not always. Decreasing m or n will cause false alarm (FA) as then it will increase the probability of one possible code word will be converted into another acceptable code word. FA is a case when code word is received with undetected error. What is coding gain? The coding gain is the reduction in the single to noise ratio or energy per bit to noise density ratio permitted by coding. Say, we achieve some reduced frame error probability using a (m, n) code. If the same reduced frame error probability were achieved, by increasing signal to noise ratio by 3 db in uncoded system, we can call the coding gain of (m, n) code as 3 db.

29. What are Hamming Codes? The Hamming Codes are linear block codes with following properties : Code size = 2P – 1 Message size = 2P – P – 1, and the number of check or parity bits = P where P > = 3. 30. Define minimum Hamming Distance. How dose it determine the capability of a parity code? If wi denotes the weight of the it h code word, minimum Hamming Distance, dmin is defined as: dmin = min(wi) of all possible i. The weight of a binary code word is the number of “1” present in the that code word. For example if a code word is 1110001, the weight of this code word is 4. Smallest non-zero weight of all weights of any code is the Hamming Distance of that code.

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For Hamming (7, 4) code, number of the code words = 24 = 16 and hence number of weights = 16. Out of these 16 weights, smallest non-zero weight is 3 which is the Hamming Distance of this code. Hamming Distance is related to the capability of a code in the following means: 1. For a code to detect upto e bits per code word, required minimum Hamming Distance > = (e + 1). 2. For a code to correct upto e errors per code word, required minimum Hamming Distance > = (2e + 1). 3. For a code to correct upto .1 errors and detect upto e2 (e2 > e1) per code word, required minimum Hamming Distance > = (e1 + e2 + 1). 4. For a code to correct upto e1 bit errors and detect upto e2 bit errors per code word, required minimum Hamming Distance > = (2e1 + e2 + 1). Dose coding increase BER? If so, why do we then use coding? Coding duly increases BER, If the transmitter power is S, this power is shared by n bits of message in case of uncoded system; and the same power is shared by n + c (where c is the number of check bits) bits in case of coded system in the same time interval as of the uncoded system. Naturally energy per bit decreases in case of coded system, and this causes BER to increase. Energy per bit equals to signal power divided by bit rate. We have seen earlier the relation between uncoded bit rate and coded bit rate which tells fb/ fc = coding rate. Energy per bit in case of uncoded system, Eb = S/fb and that in case of coded system, E = S/fc. Hence Ec/Eb = fb/fc = coding rate < 1. This means Ec < Eb. Therefore BER in case of coding is greater than that of the uncoded system. Even then we use coding because by coding we shall be able to detect/correct bit errors to some large extent. Detection/correction rate of bit errors is more than the amount of bit errors caused by increased amount of BER due to coding. So, we use coding. What is repetition code? In repetition code, bit to be transmitted is transmitted more than once. In triple repetition code “0” and “1” are coded respectively as “000” and “111”. At the receiver majority rule is applied to decide about the bit. In general repetition code can be expressed as : (2t1 + t2 , 1) where t2 = 0 and t1 = 1 for a double-repetition code; and t1 = t2 = 1 for a triple repetition code. However, double or even repetition code can not use majority rule for decoding or detection, and that is why repetition code is usually a odd repetition code. A generalized odd repetition code is represented as (2t + 1, 1) where t = 1, 2, 3, ……… What is shortened code? A shortened code is made by deleting any number of message bits from any block code (m, n). If e bits are deleted, the shortened code becomes (m – e, n – e) code. For example by deleting one bit of message, Hamming (7, 4) code could be converted to a (6, 3) code. In (6, 3) code, minimum distance is three, therefore it shall able to correct single bit error like (7, 4) code. Define the capability of (13, 8) Hamming code. (13, 8) Hamming code can correct upto one bit error and can detect upto two bits error per code word. How Hamming (7, 4) code be used to detect upto two bits error per code word. By adding an extra check bits i.e. by a code (8, 4), the same can be achieved. The extra check bit shall be used to provide parity over 7 bit codes of the (7, 4) code.

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36. What is BCH? What is the lower bound on the error correcting capabilities of BCH codes? BCH is Bose-Chaudhuri-Hoequenghem code. It is the generalized parity code. Lower bound (e) on error correcting capabilities of a BCH (m, n) code is given as: e > = (m – n)/log2 (m + 1) = (1 – r)n/log2(n + 1) where r is the coding rate. 37. In general how the error correcting capabilities of BCH code can be defined? Hence define the limits of the distance of the code. A BCH (m, n) code can correct upto e bit errors per code word where: e = (m – n)/p = (1 – r)n/p where p is an integer related to m such that m = 2p – 1 and n >= (m – pe). The code has minimum distance as : (2e + 1) <= distance <= (2e + 2). 38. What is Reed-Solomon code? Reed-Solomon codes are a class of non-binary BCH codes. A RS code which can correct upto e error can be defined as: Code size = 2p – 1 symbols. Message size = n symbols, Parity check size = m – n = 2e symbols and Minimum distance = 2e + 1 symbols. 39. What is Golay code? Give some examples of Golay codes. What is the advantage of Golay code over BCH code? Golay code is the special BCH code. Examples are : (23, 12), (24, 12) and (31, 16) codes. Implementation complexity is higher in BCH code than Golay code. 40. What is the specialty of Golay (23, 12) Code? This is only code of length 23 which can correct upto 3 bits errors per word. It dose so with moderate circuit. 41. For a (m, n) block code what is the upper bound that a code is in error? Is your result exact? What is the probability that the data is in error? For a (m, n) code there are altogether 2m possible received words out of which, possible code word are only 2n. Error in codeword occurs if any one of the other 2m – 2n codeword is received. Thus the error of code word (Pe) is bounded as: Pe <= (2m – 2n)/2m = 1 – 2 – (m – n) The result is not exact, it is a loose one. This is because, BER is not constant for coded and uncoded system. For a (m, n) code there are altogether 2m possible code words, out of which are 2n acceptable code words, each of which related to a valid data. Error in data occurs if one is converted to any one of the other 2n – 1 data. Thus data error is bounded by : Ped <= (2n – 1)/2m = 2 – (m – n). As c = m – n, check bits increases, Ped decreases but Pe increases, which shall be the case. 42. Why are we more interested in codes which can correct/detect upto only single or double bit error. If BER = 10–4, as per our error model of binomial probability the probability of single bit error, double bit error, three bit error…. become respectively of the order of 10–4, 10–8, 10–12 …… so on.

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43. What is Fire code? It is class of non-linear block code or cyclic code designed by Fire for burst or correlated error. The Fire codes are defined by parameters, M and L. The codes can correct any single burst length of length b or less and can simultaneously detect any burst of length d > b or less if L > = b + d – 1 and M > = b. For maximum error correction, d is made equal to b so that : b < = minimum [(L + 1)/2 , M] The required check bits are L + M, and L + 1 is usually made equal to M to maximize capability of correcting burst error. 44. Give a comparison between a Fire code and a BCH code. A (155, 127) BCH code which is having 28 check bits can correct upto four bits in any combination. A Fire code with L = 19 and M = 9 which is also having 28 check bits can correct any single burst of length mine. But it could not correct two errors if spaced by more than eight bits. Fire code is having extremely high coding rat. With 23 parity check bits, it can handle block of length upto 2032 bits. Its coding rate becomes about 99%. 45. What is CRC? CRC is Cyclic Redundancy Code. This code detect random errors as well as burst errors of certain length. Code uses a given polynomial generator for code generation and error detection. Check bits equal to number of bits of polynomial generator less 1. 46. Give a few versions of polynomial generators. Code Polynomial generator X12 + X11 + X3 + X2 + 1. X16 + X15 + X2 + 1. X16 + X12 +X5 + 1. X32 + X26 + X22 + X16 + X12 + X11 + X10 +X8 + X7 + X5 + X4 + X2 + X + 1.

CRC-12 CRC-16 CRC-CCITT CRC-32

47. Why generator polynomials are given and standardized? CRC technique generates code by multiplying the given message by the given generator polynomial using modulo-2 operation (XOR operation i.e. an addition ignoring carry). In CRC, error is detected by dividing the received CRC (which is transmitted CRC with error due to noises) code by the given generator polynomial using modulo-2 operation (XOR operation i.e. now subtraction ignoring borrow). If there is no reminder in the division, it is taken that there is no error; otherwise error is said to be detected. Now, it may so happen that error which has been added with transmitted CRC is a factor of generator polynomial in case of which, the process of division shall not be able to detect error. Standardized polynomials are used to cause such effect extremely rare. 48 Use a generator X3 +1 to find transmitted code for a given message 111001. Given message = 111001 = X5 + X4 + X3 + 1, Transmitted polynomial = Message Polynomial x Generator Polynomial = (X8 + X7 + X6 + X5 + X4 + X3 + X3 + 1) = (X8 + X7 + X6 + X5 + X4 + 1), where two X3 are removed because of XOR operation (addition without carry) which gives 0 if two identical bits are added (i.e. 1 + 1 = 0 or 0 + 0 = 0). Thus transmitted code is 111110001 corresponding to above derived transmitted polynomial.

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49. Why do we use modulo-2 or XOR operation in case of addition or subtraction in CRC? In fact, we not only use such operation in case of CRC only, but we use always a sort of XOR operation in all other binary codes. This is due to two reasons : (1) by XOR operation, the difference between addition and subtraction is missing, and what is actually being done by such operation is bit comparison, and this is by which decision about binary bit error is arrived at, and (2) by XOR operation, the received frame shall be represented conveniently as : Received frame = Transmitted frame + Error. Where error is a binary word of length equals to length of transmitted frame and is made with 1s at the bit location at which error has occurred and 0s’ at all other location. In other words, by just bit to bit comparison of the received and transmitted frame we could be able to know about the error. For example if transmitted frame is 11100001 and received frame is 11100011, we could find error is there at second bit position from right of the received frame. 50. What is error Syndrome? In a transmitted codeword, there is some check bits. These check bits are received at the receiver either correctly or with some errors. Similarly, receiver receives message bits. While using a code it is fundamental that transmitter and receiver mutually know about the coding being used. Accordingly, receiver based on received message bits and encoding algorithm being used shall generate check bits. These generated check bits are compared with received check bits, bit wise using XOR operation. By such comparison a binary word will come out which is known as error Syndrome. The error Syndrome points at which location error has taken place if any in the received code word. 51. Say, you are transmitting four symbols 0, 1, 2, and 3. What are the possible cases of symbol error? The possible cases are : 1. P00 = Probability that symbol 0 was sent and received without error, P01 = probability that symbol 0 was sent but received as symbol 1, and so on ………. P02 and P03. 2. Similarly P10 = probability that symbol 1 was sent but received as 0, …….. so on P11 etc. 3. Similarly we have P20 …… etc. for the case of transmission of symbol 2, 4. We have P30 ……. etc. in case of transmission of transmission of symbol 3. In general Pii for i = 0 to 3 represent probability that transmitted symbol, i is received correctly, and Pij (i is not equal j, simultaneously, but i and j both varies from 0 to 3) represents the probability that a transmitted i is received as j. 52. What is transmission matrix? It is a matrix of probabilities of reception of different bits correctly or with possible other forms due to error, while contaminated with noises due to transmission through a channel or otherwise. Referring to question (51), the transition matrix is given below: P00 P01 P02 P03 P10 P20

P11 P21

P12 P22

P13 P23

P30

P31

P32

P33

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53. Enumerate typical capabilities of a CRC code. Typically capabilities are detection of : (1) all single bit error, (2) all double bit errors as long polynomial generator has a factor with at least three terms, (3) any odd number of errors, as long as polynomial generator contains a factor of (x + 1), (4) any burst error of length less than number of check bits, and (5) most burst errors of larger length. 54. How do you represent CRC-12 equivalently as (m, n) code? It can be defined as (n +12, n) code. However, typical n for this CRC is 6 bit characters (ASCII characters). 55. Say a binary CRC code is presented as (m, n) code. State capabilities of this CRC code in terms of m and n. Capabilities can be stated as it can be capable of detecting : (1) all burst error of length m-n or less, (2) a fraction equals to 1-2–(m–n–1) of burst error of length equal to m – n + 1, (3) a fraction equals to 1-2 –(m–n) of burst errors of length greater than m – n +1, and all combinations of dmin minimum distance) – 1 or less random errors. 56. What code do we normally use in data/computer communication and networking Why? Normally CRC codes are used. The reasons are : (1) CRC can detect both random and burst errors of practical importance, (2) coding and decoding are quite simple from both software and hardware point of view, (3) cost of circuit for implementation is low and (4) CRC dose not take much time except for sending check sum or frame check sequence bits. 57. Dose checksum bits of CRC can locate bit error position? Dose it can provide any hint about volume of error in any frame? In the both the cases, answer is NO. By checksum or FCS we can only whether received frame is correct or not, and no more than this. 58. What is fingerprint? How many fingerprints a CRC can have? Fingerprint is a FCS. Total number of fingerprints equals to the number of different unique values that a FCS can have. If generator polynomial is of M bits, FCS shall be of M – 1 bits; and therefore for this generator polynomial, there can be 2M–1 fingerprints. For CRC –16, the number of fingerprints is 216 = 65536. 59. Due two data frames can produce same checksum or FCS or fingerprint for a given generator polynomial? Why? If “Yes” how can it be minimized? What is the probability that there can be a single fingerprint for more than one data? Yes, they can. This is because for data size of N, we can have different and unique 2N data, whereas for a CRC with M (N > M) bit checksum can have only 2M, where 2M > 2N, different and unique fingerprints. This can be minimized using longer generator polynomials, but then coding rate shall go on decreasing. The probability that there can be single unique fingerprint per individual code word is 2M/2N, and therefore the probability that there shall not be unique fingerprint for individual data or there can be same fingerprint for more than one data is 1 – (2/2N). 60. Why CRC is more efficient than even simple parity? It can be illustrated with a simple. Say, we are using simple one bit parity each for a byte. Therefore its coding efficiency is (8/9) X 100 = 90% and the percentage of extra bits is (1/8) X 100 = 12.5%. We can use only 32 check bits of a CRC-32 code over a data of size 2048 bytes. In this case coding efficiency is { (32 + 2048 X 8) / (2048 X 8) } 100 = 99 % and percentage of extra bits is {32 / (2048 X 8)} X 100 = 1.6%. In addition, capability of CRC32 is far more than simple parity.

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61. How a circuit is made to generate checksum or FCS, for, say CRC 16? Step 1. CRC 16 = X16 + X15 + X2 + X0. Take a shift register of size equals to the highest power of given generator polynomial. In this case, shift register shall be of 16 bits. Step 2. Subtract the powers of X of given generator polynomial from 16 (size of the shift register as find in step 1), and note the results as different positions. In this given case, the positions are 0 (which is due to 16 – 16), 1 (which is due to 16 – 15), 14 (which is due to 16 – 2), and 16 (which is due to 16 – 0). Step 3. Use XOR gates at positions found at Step 2. as shown in the following Fig. (1). Before transmitting data, Clear the shift register. After that transmit the transmit the data as shown. When all data bits are transmitted, what is left in the shift register is nothing but FCS. 1

14

16

Shift register 2

3

4

5

6

7

8

9

10 11 12 13 14 15 16

Checksum

Input data

62. What theory gave the birth of CRC? It is algebra theory. 63. What CRC is used in Local Area Networks of IEEE 802 series? What is the size of check bits or FCS (Frame Check Sequence)? CRC-32 code is used in LANs of IEEE series. Number of check bits or size of FCS of CRC-32 is 32 only. 64. What CRC is used in link level of SS7 signaling, LAP B and LAP D protocols? In all these cases CRC-CCITT code is used. 65. What is concatenated code? Concatenated code is a coding technique which uses more than one code. For example, for a given message block of n bits we can a error correcting code (m, n). Now the code of m bits so generated shall be used as a message block to generate a new code say, (M, m). Concatenated code improves the performance of coding. For example if two CRCs are operated on a message as illustrated above, detection of burst error of length may go upto the sum of lengths of individual CRCs. 66. What shall be the coding rate of the concatenated code? It shall be equal to the product of individual code rates. 67. What is maximum length code? For any positive integer P > 3, a maximum length code is a code that following: Code size m = 2P – 1, Message size = P, and Minimum distance = 2P – 1 These codes use generator polynomial (1 + Xn)/h(X) where h(X) is any primitive polynomial of degree P. 68. What is convolution code? Where is it used? In block code, a fixed message size is required. Coding is done over this message block. There are situations where message come in form of serial bits rather than block, viz,

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satellite communication. In such situations, convolution coding is used. Convolution coding operates serially on incoming message continuously. An encoder of a binary convolution code with rate 1/m, produces a coded output sequence of length m(L + N) bits corresponding to a N bit message sequence using a finite state machine that consists of 1 shift registers. Coding rate is then, N/m (L + N) bits/symbol. Usually N > > L, hence coding rate is 1/m bits/symbol. Give the advantage and disadvantage of convolution in general. Convolution code in general more effective and simpler than block coding. The code has advantage : (1) codes are applied to channel of fixed bandwidth, (2) in general, code has better performance than block code for equivalent circuit complexity, and (3) decoding algorithm in case of convolution coding can be adopted in compliance with data source statistics. The disadvantage of the convolution coding is that code design with bit error approaching zero fairly depend on a random coding argument rather than on a specific construct as in block codes. Name a few decoding algorithm used in convolution codes. Code tree algorithm, State and Trellis diagram, Viterbi algorithm are to name a few. What for convolution is used? Convolution coding in general used for error correction, where block codes are used in error detection, in general. State a practical concatenated code that makes use of both block code and convolution code. In fact, concatenated or hybrid code that is mostly used in practice, uses both block code and convolution code. The given message is first encoded using block code, the block coding thereby being known as inner code. The resulting block code then is encoded using convolution code, whereby convolution code is outer code. This combination provides an excellent error correction and detection capability. What are the basic two error control techniques? How are they related to error detection/correction coding? Basic two error control techniques are : Forward Error Correction (FEC) technique, and Backward Error Correction (BEC) technique. In FEC technique, error is corrected at the receiver; and therefore Error Correction Code is must for it. In BEC technique if error is detected at the receiver, receiver asks transmitter to retransmit the data: and this way the error correction is made with. Thus, in BEC error detection code is sufficient. What error control technique is used in practice in data/computer communication and networking? Why? We have seen CRC is the best code for networks due to a number of reasons. CRC being a error detection code, BEC is the usual technique of error control in networks. Beside, error correction code requires much more check bits than error detection code which tills the decision in favor of BEC. What are different techniques of BEC technique? Do they require some memory storage at the transmitter, exclusively for error control? Different techniques are : (1) Stop and Wait Automatic Repeat Request (ARQ), (2) GoBack-N Automatic Repeat Request (Go-Back-NARQ) and (3) Selective Repeat Request. ARQ dose not, as such, require any memory for error control. Go-Back-N ARQ requires about N × D (where D is the data/packet size in bytes) bytes of memory locations. Theoretically, Selective Repeat Request requires infinite memory location.

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76. If acknowledgement is lost in different ARQ techniques, how the error control operation shall be successful? Transmitter in ARQ techniques waits for acknowledgement from the receiver before taking subsequent appropriate action. But if the acknowledgement is not reached due to loss etc within a time, known as time out period, after transmission of a frame or a set of frames the transmitter assumes that the transmitted frame or frames have not reached the receiver correctly. The transmitted, then starts retransmission. 77. If the transmitter retransmits frames after a time out period in case of loss a positive acknowledgement (signifying that frame has been correctly received by the receiver) from the receiver, how the receiver shall identify the retransmitted frame as retransmitted frame rather than next frame? The receiver identifies retransmission frame from the next frame by the sequence number of the frames, which is attached by the transmitter as a header to each frame during transmission. 78. At what value of N, both selective repeat and Go-Back-N-ARQ shall reduce to ARQ? When N = 1. 79. What technique is mostly used for error control in networks? Why? Go-Back-N ARQ is mostly used. It requires moderate memory which is much less than that is required in selective repeat request. Besides its throughput efficiency and link utilization are better than those of IARQ, although they are less than those of selective repeat request. 80. Name a few data link protocol where Go-Back-N ARQ is used? SS7 signaling, LAP B, LAP D and X.25 are to name of few. 81. What is the value of N used in Go-Back-N ARQ technique used in ARPANET? It is 127. 82. Define link utilization as used in case of error control? Link utilization measures the efficiency of utilization of available link capacity. It may be defined as the ratio of transmission time required by a transmitter to transmit a frame to the time before the next frame could be transmitted. It can also be defined as a ratio of transmission time of a frame to the actual average time spent in successful transmission of the frame under a particular error control technique. 83. What is the throughput efficiency as defined in case of error control? Throughput efficiency of an error control technique is defined as the product of link utilization of the technique and coding rate of the code used in that error control technique. 84. Compare throughput efficiencies of different error control techniques. Throughput efficiency of the selective repeat request > that of the Go-Back-N ARQ > that of the ARQ. 85. Define parameter “a” commonly used in analysis of link utilization etc? What dose it signify? “a” is defined as the ratio of the propagation time in between source and destination to the transmission time of the frame. Significance is as below: 1. If “a” is less than 1, the round-trip delay between source and destination is mainly due to the transmission delay.

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2. If “a” is equal to 1, the round-trip delay is determined by both the propagation delay and transmission delay. 3. If “a” is grater than 1, the round-trip delay is mainly determined by the propagation delay. When is the maximum link utilization achieved? It is achieved when probability of the frame in error is zero. (A) If the maximum link utilization is attained when probability that the frame is in error is zero, it then natural that utilization shall be 100%. Dose it happen so in all cases of error control technique? NO, it dose not happen in all the cases. It happens in two cases : (1) Go-Back-N ARQ with continuous operation and (¹) selective repeat request with continuous operation. In all other cases, namely, ARQ, Go-Back-N ARQ with discontinuous operation and selective repeat request with discontinuous operation, the utilization is not 100% even in case of zero probability that a frame is in error. (B) What is problem with discontinuous operation? What is the maximum link utilization attainable in case of ARQ? Why is it so? The maximum attainable link utilization in case of ARQ is 1/(1+2a), (provided processing time, acknowledgement time and modem turn-around time etc negligible) which is less than 1 i.e. 100%. The maximum utilization is achieved when the probability that a frame in error is zero. Even then, as per IARQ technique transmitter has to wait for a positive acknowledgement from the receiver for each frame. This makes additional loss of twice propagation time per frame. This contributes to make utilization less than 100% What is piggybacking? This is a technique of sending acknowledgement with data. Receiver sends acknowledgement (positive or negative) by the sequence number of received frame. Instead of sending acknowledgement sequence number separately, receiver sends the same with its piggybacked frame which consists of data field, a field reserved for data sequence number, and a field reserved for acknowledgement sequence number along with others. By this, communication resource is utilized in a better way. What will happen to a piggybacked frame, if a receiver has only data to send and no acknowledgement? The receiver shall send piggybacked frame where the field of acknowledgement sequence number will be filled with previous number. This previous number when reaches at the other end will be simple ignored. What will happen if there is only acknowledgement, and no data to send? Receiver has to send a separate acknowledgement frame. Is piggybacking used in Go-Back-N-ARQ? Yes, it is used duly. In a Go-Back-N-ARQ, if a frame is received out of sequence, what is done? Frame is simple discarded. What is mode of sending acknowledgement sequence number in Go-Back-N-ARQ? If the sequence space is k bits, sequence number can range from 0 to 2k – 1. If k = 3, sequence number ranges from 0 to 7. Sequence numbering is done with modulo 2k operation. This means after sequence number 2k – 1, the next sequence number starts at 0 and proceeds towards 2k – 1.

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95. What is maximum window size in Go-Back-N-ARQ technique? If the sequence space is k bits, the maximum window size is governed by: Window size < = 2k – 1. For k = 3, therefore maximum window size is 7. 96. What can be maximum window size in case of selective repeat request? Maximum window size is half of the sequence space. If sequence space is k bits, maximum window size is 2 k – 1. If k = 3, this maximum is 4. 97. Can we use ARQ in satellite communication? The use of ARQ in long distance communication could be unacceptable. Long distance communication means long propagation delay, and that means a very high value of “a”. We have seen earlier chat the propagation delay or for that purpose the parameter “a” limits the link utilization to a maximum of 1/(1 + 2a) when even probability of frame in error is zero. With increase of “a”, the utilization decreases. Therefore, ARQ is not suitable for long distance communication like satellite communication. 98. If the channel is Gaussian, what shall be the limit of the ratio of energy per bit to noise density, in order to communicate at the full channel capacity under Shannon limit? Eb/No > = – 1.6 bd. 99. What is 74LS636 IC? What is 8004 IC? 74LS636 is a complete error detection and correction chip for (13, 8) Hamming Code. The chip is of Texas Instruments. 8004 IC is a dual CRC-32 SDLC generator and checker. 100. (A) Different types of Errors occur in frame as below : In case of synchronous communication: (1) invalid frame, (2) Abort, (3) Overrun and (4) FCS error; and in case of asynchronous communication : (1) framing error, (2) overrun error, and (3) parity error. By error control, what error of the above do we reduce ? (B) In data/computer communication, we have three important aspects: Security, Accuracy and Privacy, which are controlled by coding. By error coding, which of these we are controlling? (A) By error control, we control FCS error and parity error. (B) By error correction and/or detection coding, we control accuracy. 101.Write a critical note on Turbo Code Turbo Code-An Error Correcting Code for Next Generation The coming up knowledge age like the present age of information will vastly and mainly depend on how reliably, securely and effectively the talking computers (the computers connected in the networks) exchange data, information and knowledge among themselves. The data, information and knowledge the talking computers exchange among them are represented in the form of a string of binary logical bits, 1s and 0s. The reliable transport means the sender’s bits are transported accurately over the network and are correctly received by the receiver. The message may be incorrectly received due to bit errors. The bit error occurs when a transmitted 1 is received as 0 or a transmitted 0 is received as 1. The transformation of bit either from 1 to 0 or from 0 to 1 is caused by noise signals in the channel or in the system. The logical bits are transported over networks as physical signals, the simplest example of which is: logical 1 is represented by a positive pulse of 5 volts or more and logical 0 by negative pulse

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of – 5 volt or less. The noise signals when are contaminated with physical pulses, a positive pulse may convert to a negative pulse and vice versa. This is how bit errors are resulted in. The different methods are used to combat with the bit errors. The well-known two methods are Forward Error Control (FEC) and Backward Error Control (BEC). FEC uses error correction code. Error correction code is capable of correcting error. Thus the error is corrected at the forwarding devices, the receivers. FEC is better suited to the environment where the bit error rate is higher. The environment of wireless communication and network is such an example. For the environment of moderate to low bit error rate the example of which is the current wired based network, the technique of BEC is used. BEC makes use of error detection codes. Error detection codes can only detect errors but can’t correct errors. Thus the receivers can’t correct errors. Once the errors are detected, the receivers ask the transmitters for retransmission of message received but detected in error. This is how BEC corrects error. Both the error correction codes and the error detection codes are based on the philosophy of “redundancy increases reliability.” For example, if you post same letters twice to your friend, you become more confident that at least one of the two letters posted, should reach your friend even if another is lost and/or misplaced or damaged. In the example, one more letter (redundant) increases the probability of your friend getting the letter, that is, the reliability of reaching the letter to your friend increases. But the increased reliability is achieved at the higher cost of post, because the required postal stamps for two letters will be double that of the one letter. A simple example may further clarify the concept. Say you want to send two data, x and y to your friend. In addition to original data x and y, you may send two more redundant data as x + y and x – y. So your code now: x, y, x + y and x – y. You transmit your code to your friend. Say error occurs at y, and due to the error your friend received the code as: x, z, x + y and x – y. Your friend from received x + y and x – y can compute x and y. On comparing computed x with received x and computed y with received z, he can easily detect the occurrence of error. Similarly the error correction codes and the error detection codes use redundant bits, known as check bits. Examples of error detection source codes among others are parity code and CRC-32 code that use respectively 1 and 32 check bits. Examples of error correcting source codes among others are (7,3) Hamming code, Repetition code and BCH codes. The redundant check bits are used to correct or detect error but they result in the increased cost of transmission by consuming more bandwidth like the increased cost of postal mailing of redundant letters. The increased reliability of data transmission due to the application of the correction code or error detection Step 1 :

Data

Encoder 1 (Convolution encoder) Step 2 : Interleaver

1 0 1 0 0 1

Step 4 : (Parity 1) Step 3 : Encoder 2 (Convolution encoder

0 1

Function of Interleaver : An example Transmitter Side

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Puncture. Example : data bits followed (Parity 2) parity bits of encoder 1 and encoder 2

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Step 6 : Data received :

Step 7 :

Step 5 :

Data bits as received

Decoder 1 (Convolution decoder)

Parity 1 bits as received

Interleaver

De interleaver Decoder 2 (Convolution decoder)

189

De-puncture

Parity 1 as received

Receiver Side (Turbo Decoder) Fig. 1: Illustration of working principle of turbo codes

code has a trade off with increased consumption of bandwidth for transmission. The error correction code uses more check bits than that of the error detection code. Thus till date the application of the error detection codes has outnumbered the application of the error correction codes. But with the appealing flexibility and other benefits, the coming age of networking has widely titled towards the wireless networking. This has made necessary for a search of powerful error correction code. The famous Turbo Code is ahead in the race in this regard as FEC line code. Turbo code has out scored. As we had seen code is made of original data bits and a set of extra bits called check bits. Turbo code does follow the same concept but with a great innovation. Turbo code is a combination of two simple recursive convolution codes (Fig. 1) in parallel. Thus turbo code may be called PCCC (Parallel Concatenated Convolution Code). Every code has a rate. Rate is the number of the information bits divided by the number of the information bits plus that of the parity bits used in code. In the turbo code, each of the two encoders produce the number of parity bits equal to the number of the information bits. The total parity bits are twice the number of information bits. Thus the turbo code is usually a 1/3 rate coder. The input data is a block of data of n information bits. Actually turbo encoder starts with three copies of input information bits. One copy directly goes to the puncture unit. Second copy goes to first encoder. Third copy goes to the interleaver before it goes to encoder 2. One out of two encoders of the turbo code generator generates parity bits based on input bits. The real innovation in the turbo code is the use of an interleaver. The interleaver allows a permutation of the n information bits before they are input to the second encoder. The second encoder generates parity bits based on the interleaved bits of the original sequence. Even though the individual groups of the parity bits from two respective encoders produce each a weak code, yet the combination becomes a powerful code. Logically the combination produces increased redundant bits thereby increasing reliability. The increased redundancy can be achieved by just duplicating encoder, but that will not produce any increased redundancy in effect. Only the interleaver innovates increase redundancy. How? Let us take a very simple example. If x and y are the two data, we can have an encoder to produce two redundant check data, x + y and x –y. If we duplicate the encoder, we may get one more set of x + y and x – y. But that does not produce any unique redundant data. Had the original data were interleaved as y and x before inputting to the second encoder, the redundant data would have been y + x and y – x. So by interleaving three unique redundant data are produced, x + y, x – y and y – x (note that x + y and y + x are same and one, but x – y and y – x are not same and one so long they are not equal).

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Really, the interleaver makes the turbo code what it is today. The information bits and the two groups of the parity bits from two encoders of the turbo encoder are punctured before releasing to the link. A few examples of puncture are: (a) information bits followed by parity bits of encoder 1 and then parity bits of encoder 2; and (b) first bit of information followed by first bit of parity bits of encoder 1 followed by first bit of parity bits of encoder 2…. the pattern repeats for other bits. In the link the logical bits propagate as physical electromagnetic or optical signal as per the media of the link. The link noise contaminates with the physical signals of the bits resulting bit errors. The received signals are used in the turbo decoder at the receiver to recover the corrected bits. The decoder at the receiver performs the complimentary functions of the turbo encoder as shown in fig(1). In order to finally correct the errors, the two decoders exchange their individually assessed data repeatedly. After a number of exchanges or iterations, around 5 to 10, the decoders may reach an agreed decode data as a fully corrected data. The operation may be illustrated in a very simple way with our previous example of x and y respectively as 7 and 5: Turbo Encoder Turbo Decoder #Data bis = 7, 5 #Encoder 1 produces parity bits = 12, 2 #Interleaved data = 5, 7 #Encoder 2 produces parity bits = 12, –2 #Puncture data = 7, 5, 12, 2, 12, –2 which is released in link

#Say data received as 7, 6, 12, 2, 12, –2 (note that data 5 is in error and is received as 6) #Decoder 1 receives data set as = 7, 6, 12, 2 ; and say it decodes data as 7, 6 assuming wrongly received data 2 as error. Then it sends decoded data 7, 6 to decoder 2 #Decoder 2 receives data set as = 7, 6, 12, – 2 ; and say it decodes data as 7, 5. Then it sends decoded data top decoder 2 #Decoder 1 getting the data 7, 5 from decoder 2 verifies that this data set conforms to its received parity 12, 2. So it agrees to 7, 5 Agreed corrected data = 7, 5

(THIS IS ONLY A TOO SIMPLIFIED ILLUSTRATION. IN ACTUAL CASE, agreed corrected data is achieved after a number of iterations) The existing error correction codes namely Hamming Codes, BCH codes, Reed Solomon code and Convolution codes are widely used. They are very powerful codes too. So what is the great in Turbo code? How is it different from other codes? The problem basically lies with cost and complexity required to decode the data. As was pointed out earlier, the capacity of a code in terms of error correction and/or detection increases by increasing the check bits. As check bits increase, the code size increase that necessarily results in higher data rate. With data rate the bit error rate increase. So the problem and its solution are in trade off and counter challenging to each other as if in conformity to the basic natural law of the Newton that to every action there is an equal and opposite reaction. Thus obvious solution of the trade off problem is to have an infinitely long code word that was duly demonstrated by Shannon, the father of the information theory. But it is only theoretical answer. It was Claude Borrow and Alain Glavieux, two French Professors by their invention known as turbo code implemented practically the idea of Shannon. Two encoders of turbo encoder actually generate two unique codes (in our example considered earlier two codes x, y, x + y, x – y and x, y, x + y, y – x) each of 2n bits (if information is n bits, check bits are n bits) for a single but transmit not 2 × 2n = 4n bits but 3n

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bits. This is the beauty, and it is possible only with interleaver of the turbo code. This is how turbo code solves the cost and the complexity problem of existing codes as data or code size increases. Turbo codes are very powerful FEC. The code has viable application in high bit error rate links like wireless environments and/or in the power limited noisy environments like satellite links, deep space communication, space communication, military hand held satellite radio communication. The turbo code is believed to be effectively applied in Wireless LAN (Local Area Network), WLL (Wireless Local Loop), CDMA (Code Division Multiple Access), 3G/4G (Third Generation/Fourth Generation) and PCS (Personal Communication Services) that require codes to handle bit error rate ranging from as low as 10–5 to as high as 10–2. A bit error rate of 10–2 means on average out of 102 transmitted bits 1 bit will be in error. In the satellite and the deep space communication, the transmitted power is limited due to battery operation. This causes the system to have low signal to noise ratio resulting high bit error rate. The turbo code is suitable for these applications. High data rate is another source of bit error. As data rate increases bit error rate increases. Turbo code effectively maximizes the data rate. The data rate is limited by the famous Shannon’s theory. It is estimated that with turbo code the Shannon’s limit may be achieved. In fact it was Shannon who first gave the idea of the code to tackle errors that plagued all communication links. He also argued that with right code word it is possible to achieve the channel capacity - that is exactly what is being expected now from the turbo code. The multimedia communication is based on high data rate communication in order to provide guaranteed quality services. With turbo code, the high data rate multimedia communication will be easy to operate.

5.5 CONTROL FIELD HDLC or SDLC frames can be of three types, namely information frame (I-frame) that carries valid information in the information field, supervisory frame (S-frame) that is used to transfer supervisory signals, and unnumbered frame (U-frame) that is used for specific services and administration. When the first bit (LSB) of the control field is 0, it is an I-frame. When the first two bits of the control fields are 10 and 11, the frames are respectively S-frame and U-frame (Fig. 17). The three bits N(S) and N(R) field of the control filed respectively indicate the send and receive sequence of the frames. The reason and use of sequence numbers are discussed in the section of error control. The two bit codes of different frames are shown in Fig. (17). Flag

I-Frame

Address

Control

0

P/F N(S)

S-Frame

1 0

N(R) P/F

Code U-Frame

Information

0

N(R) P/F

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Flag

P/F

Pool/Final bit.

N(S)

Sequence number of frame sent.

P/F

Pool/Final bit.

N(R)

Sequence number of next frame expected.

Code

Code for supervisory or unnumbered frame

Code

(a)

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Only in I-and U- frames

Flag

Address

Control

Information

FCS

Flag

Flag

Address

Control

User information

FCS

Flag

I-frame

Flag

Address

Control

Flag

Address

Control

FCS

Flag

U-frame

FCS

Flag

S-frame

Management information

(b) Different HDLC frame types The four different types of S-frames as : Code 00 01 10 11

Frame type RR REJ RNR SREJ

Frame name Receive Ready Reject Receive Not Ready Selective Reject

(c) S Frames Code Command/Response

Meaning

SNRM

Set normal response mode

SNRME

Set normal response mode (extended)

SABM

Set asynchronous balanced mode

SAMBE

Set asynchronous balanced mode (extended)

UP

Unnumbered poll

UI

Unnumbered information

UA

Unnumbered acknowledgement

RD

Request disconnect

DISC

Disconnect

DM

Disconnect mode

RIM

Request information mode

SIM

Set initialization mode

Command/Response

Meaning

RSET

Reset

XID

Exchange ID

FRMR

Frame reject

(d) U-frame control command and response

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Code 00 11 11 11 00 00 00 10 00 11 11

001 011 100 110 000 110 010 000 100 001 101

10

001

Command

Response

SNRM SNRME SABM SABME UI

DM

DISC SIM UP RSET XID

193

UI UA RD RIM

XID FRMR

(e) U-frame control field in HDLC Fig. 17: HDLC control fields in details with illustration

5.6 OTHER PROTOCOLS In addition to OSI protocol, other established protocols are SNA (Systems Network Architecture) developed by IBM and DNA (Digital Network Architecture) developed by DEC. However there is no single protocol, which is to be used as superior to other in most applications. But the functional similarity among OSI, SNA and DNA is readily observed (Fig. 11). Two networks having different protocols may be connected through a protocol converter. Protocol converter is a complicated system. OSI/ISO protocol is open to any networking system. The initial ISDN (integrated services digital network) was also based on this protocol. If incompatible device and nodes follow the protocol, the incompatibilities among them would not cause any problem in communication. That is why protocol is known as 'open system' i.e. open to any system, whether it is a device or/and network. BOX 7 Determining packet size still may be a research topic ! Switching Techniques: Review and Modifications

Introduction In the non peer-to-peer networks, there is no direct path between every pair of computers and/ or terminals that may wish to communicate. This needs some form of switching within the network. There are different forms of switching [1] used in different applications (Fig. 1): 1. Circuit switching 2. Message switching 3. Packet switching 4. Hybrid switching 5. ATM/Fixed size Cell Switching

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Switching techniques

Circuit switching (Used in telephone networks)

Store-and-nforward switching

Message switching (Used in telegraph networks/e-mail)

Virtual circuit

Packet switching (Used in data networks)

Very fast/fixed size packet switching/ATM cell

Datagram

Fig. 1: Switching Techniques

1. Circuit switching The principle of circuit switching is to form a continuous physical “copper’’ or “wire” path in between the source and the destination by appropriate switching [2] at the intermediate switching centers; and to dedicate the path in between the source and the destination (a single set of users) for entire duration of the transmission without interruption. No other potential user can use the path until released and during this time even if intervals are available. Circuit switching is the technique on which public switched telephone networks (PSTN) work. This technique is appropriate for voice communication as it provides instantaneous (note that voice is time sensitive in nature) and two-way interactive link so important for man-to-man communication. However when circuit switching is used for data, there develop a number of problems: 1) the link utilization is low [3] and hence the system is uneconomical. This is due to the fact that data is very bursty. The data traffic often consists of short bursts of data followed by long intervals of no data (during which dedicated path is not being used). Typically two computers can use only 1% of the time allocated. But the users have to pay for the entire duration of the dedicated link period. A possible alternative might be to realize separate call for each burst of data; but it is likely to be inefficient due to relatively long call set-up time (typically in seconds for analog switching exchanges). 2) data transmission has a quite wide range of transmission rate-typically from hundred bits per second (communication in between a computer and a terminal) to million bits per second (communication in between two computers on o long haul network). Hence switching must provide the maximum transmission rate between all users simultaneously (i.e. exactly same data rate to all users - how is it possible?) to tackle peak demand as data can not be stored or delayed in circuit switching mode and as the two station are in direct communication in circuit switching.

2. Message Switching The message switching technique [4] overcomes most of the limitations of the circuit switching technique when used in data networks. In this technique, instead of switching the centers to establish the link as in circuit switching, the circuit (links) are made permanent and the message is switched around the network to reach the destination from the source i.e. the message is passed from node to node till it reaches the destination. An address is attached as header to

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the beginning of message. Based on the address and routing strategies, the message is forwarded from node to node when link facilities become available. During the intermediate period (when link is not free or when routing is being processed based on address etc.), the message is temporarily stored; and that is why the message switching is also known as “Store and Forward” switching. Message switching is in use for telegrams and E-mails. The technique overcomes the problems of the circuit switching as there is no call set-up delay and two users do not have to communication at the same speed, as they do not communicate directly [5]. The problems of message switching are due to the fact that the whole of the message under transmission is treated as one unit. For some applications message may be vary long (exam: a complete file, a whole of a database etc.) and for some other applications message may be quite short (exam: a database query, a file query etc.). The long message may monopoly a particular link preventing other messages some of which may be of more urgent in nature, to use the link. Moreover, the storage space of an intermediate node may be insufficient for a long message or for a whole set of message. These two problems may lead to slower response time to other users.

3. Packet Switching In the packet switching technique each message is broken into smaller but optimal pieces called “packet” [6] that provide an acceptable compromise in between response time and efficiency. By this, delay is minimized and problem of storage as in message switching is avoided. Thus the packet switching is nothing but a modern version of old concept of message switching. In packet switching a packet as the unit of data is transmitted rather than while message as a data unit as in message switching. Typically a packet is of length from 1000 to a few thousand bits. The packet size may be different for different public switched data network(PSDN). However when such different PSDNs are inter-connected; data packets may have to be combined or split as they pass from one network to another network. Each packet bears a header carrying the address of destination and a sequence number. However packet switching is an attempt to combine the advantage of both the message and the circuit switching (using two concepts of datagram and virtual circuit; and optimal packet length); and at the same time to minimize the problem of both. Packet switching is rather like a convention postal service in which letters are passed from one post office to another till they arrive their destination. Typically the delivery time on modern packet network is about tenth of a second. The problem in packet switching is the proper handling (sequencing, error, flow and congestion control etc.) of the stream of packets. However there are two approaches in packet switching; datagram and virtual circuit. In datagram [7] service, each packet (datagram) is treated independently. They are transmitted from node to node independently till they reach destination. Routing at intermediate nodes is determined based on network’s strategies and availability of link. Thus the packets of a particular message may reach the destination out of order of sequence. Thus it is the destination’s responsibility (host machine’s responsibility) to order the sequence as well as to see loss of or duplicate packets by providing error and flow control. Hence in datagram service error and flow control are done as in the layer-4 (transport layer) in terms of OSI/ISO protocol. In virtual technique, a logical circuit in between a source and a destination is established prior to data transfer; and the circuit is used to route the data. The logical circuit is not dedicated path as in circuit switching, and hence the virtual circuit may be shared by other potential users. The packet may be buffered and queued at each node. The difference of the virtual circuit from the datagram service is that there is no need of routing decision for each packet at

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the intermediate nodes. It is made once for all along the route and the logical circuit remains open till it is disconnected by the users. As all packets follow same path they arrive in order of sequence. Error control and flow control can be done via virtual circuit facility in intermediate nodes. The error and flow control are done at level-2 (OSI/ISD protocol) for each link and at level-3 end-to-end control. If two stations wish to exchange data over a long period of time the virtual circuit approach is preferable to the datagram approach. A comparative study of datagram and virtual circuit is given in table (1). Crucial Issue of Packet Size—A New Look In literatures [8-11], the packet switching is often compared with the message switching in terms of speed only. Based on such comparison, often a decision about the optimal size of packet is arrived at. Undoubtedly, speed is an important parameter of comparing switching techniques. But the actual trade off parameter of the packet switching is speed versus overhead bits. In this section, we propose a comparison of the switching techniques based on the trade off parameter: speed vs. overhead bits. Parameters of comparison A parameter with name power, η is introduced for the comparison of the switching techniques. The power, η, is defined as: η = Coding Efficiency × Speed ...(1) The power η of the switching technique is analogous to the gain-bandwidth product of an amplifier. The coding efficiency takes care of the effect of overhead bits; and thereby, the product of coding efficiency and speed, defined as the power of the switching technique shall measure the quantitative aspect of the switching technique in reference to the trade off parameter: speed versus overhead bits. Source

Destination

1

3

2

4

K+

Nodes are equidistant apart. C = link speed in BPS.

Fig. 2: A typical Network

In order to compare the packet switching with the message switching in terms of the power, h, we assume a network of Fig. (2). A message of M bits is assumed to be transferred from source to destination under the message switching and the packet switching. Under the message switching, the full message of M bits shall be transferred as one unit with negligible overhead bits. Under the packet switching, we assume that N packets each with h overhead bits are required to transfer the whole message of M bits. Hence, neglecting propagation delay, we have:

and

LM 1 OP N K. M/C Q R| U| L O 1 1 =M N M + N. h PQ × S| K(M/N + h) + (N − 1) (M/N + h) V| C T C W

ηmessage = 1 ×

…(2)

ηpacket

...(3)

where speed is measured in messages transferred per sec.

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From eqns. (2) and (3), we derive a relative power as : ηpacket ηr = ηmessage

...(4)

From eqn. (4), the packet-switching shall have maximum gain over the message switching when ηr is maximized with respect to N. Maximizing ηr with respect to N, we get optimal N (as N1) as:

M 2 (K − 1) =0 …(5) h If conventional comparison as done is literatures[8, 9] is done in terms of the speed only, the optimized N (as N2) that could have offered maximum gain to packet switching over message switching is obtained as: N13h + N12 {h(K – 1) + 2M} –

N2 =

M(K − 1) h

...(6)

Results of comparison For different sets of K, M and h, we make a comparison N1 and N2 and relative power in table (2). In table (2) we have :

FG M IJ × RS H M + N . h K T K(M/N F M IJ × RS =G H M + N . h K T K(M/N

ηr(based on N1) = and

ηr(based on N2)

1

2

UV W UV + h) W

...(7)

1

KM + h) + (N 1 − 1) (M/N 1 + h)

...(8)

2

KM + h) + (N 2 − 1) (M/N 2

The value of N1 was obtained by Computer Program based on Newton-Raphson’s method and using equation (5). We find that the optimal N (as N1) derived from the comparison based on η offers better result than that of N (as N2) derived from the comparison based on the speed only. It is concluded that the power defined in the paper may be more appropriate for comparing all switching techniques of store and forward types. Frame bit

A

B

C

A

B

C

A

B

C

A

B

C

A

B

A

(a) STM Flag

2-bytes header

Variable-length

Information field Frame check sequence

(b) Conventional Variable Length Packets 5–byte header

48–information field (c) Fixed size packet/ATM cell

Fig. 3: Illustration of different switching/Packets

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Table 1: Comparison of Virtual and datagram services Datagram Service

Virtual Circuit Service

1. There is no call set-up phase. Good for short period of exchange. Data delivery is quicker.

1. There is call set-up phase. Not good short period of exchange.

2. Communication processing in terms of routing decision based on address and relieving of station etc. is required for each packet at each intermediate node. Not good for extended period of exchange.

2. No routing decision etc. required. Good for extended period of exchange.

3. More flexible. If congestion develops at any node, incoming packets may be routed through other link to avoid congestion.

3. As there is predefined path it is more difficult to deal with congestion.

4. More reliable. If a node is lost, packet may find alternate route.

4. If a node within path fails, no alternate route is there.

5. Service quality is not so high Sequencing is a must at destination along with flow & error control.

5. Service quality is high as there is no need of sequencing at destination. Error & floe control are done at level 1 and level 3.

Table 2: Comparison of optimal N and ηr Set

N1

ηr based on N1

N2

K = 10, M = 1024 bytes, h = 4 bytes

32.64

6.16

48

5.97

K = 10, M = 1024 bytes, h = 8 bytes

22.64

5.16

33.94

4.93

K = 10, M = 1024 bytes, h = 16 bytes

15.55

4.09

24

3.84

K = 10, M = 2048 bytes, h = 4 bytes

46.74

7.046

K = 10, M = 4096 bytes, h = 4 bytes

66.66

7.76

K = 100, M = 1024 bytes, h = 4 bytes

95.81

26.04

159.19

23.43

K = 1000, M = 1024 bytes, h = 4 bytes

195.85

52.61

505.71

37.96

7.88 96

ηr based on N2

6.88 7.64

4. Hybrid switching Now-a days different hybrid forms of circuit and packet switching are possible as computer and communication technology are coming more and more closer together. Fast-connect circuit switching is an approach where call set-up time of telephone switching is expected to be milliseconds or less. However such systems will be expensive [8]. Time division switching is an interesting variant of packet switching. In this technique each node will scan input line in a predefined rotation and each packet is immediately outputted on a correct output line as soon as the header is read. Fixed size packet and strict synchronization are needed. No storage space is required at node.

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5. ATM Cell Unlike conventional packet switching, in ATM cell switching the ATM cell(a typical packet) uses a fixed length and very short cell (Fig. 3). Each cell is 53 bytes in length with 5 bytes headers and 48 bytes data field. Such a fixed and short cell was proposed to make switching technique applicable to integrated service of voice, video and data etc. The time dependent service like voice and video suffer from randomly varying delays and irregular gaps. Data has no such problem but may have a message of 64 kilobytes or more. By breaking the message into short cells and assigning priority to voice, video and other time dependent service over time independent service, the multimedia over a single channel is possible to realize. By allowing several service on a shared physical medium, ATM increase the efficiency of communication network. This make ATM a practical switching for any service. ATM is an extension of fast packet switching. ATM is virtual connection oriented technology. The connection is ensured and determined by 5 bytes of header. ATM is a best choice for an integrated network because of its ability 1) to handle variable bit rates upto tens of megabits per second, 2) to comply with real time services and 3) to handle scalable service; and all these with maximum flexibility. Short cell of ATM ensures integration possibility and high speed network.

Variant ATM—An Efficient Proposal It is seen that ATM differs from traditional packet switching in two respects. First, the ATM cell is fixed in size unlike the variable size of packets in packet switching. Secondly, the size of ATM is short and only 53 bytes; whereas packet size in conventional packet switching may be on average more than 512 bytes. These two departures of ATM cell from conventional packet switching, make ATM appropriate for voice and voice-like communication in addition to traditional applications of data communication[12-17]. By incorporating voice and voice-like services in this manner. ATM may likely to do some unjust to traditional data communication. We shall discuss a concept of multilevel ATM. Multilevel ATM may be an open transport technology for incorporation of existing and future services and application independent networks.

Coding Efficiency and Quality Factor Two parameters those shall be used in comparing ATM, with proposed multilevel ATM are coding efficiency and quality factor. Coding efficiency measures how less extra bits are required in a particular scheme for carrying a given message. It is defined as per conventional definition of coding efficiency used in error correction or detection codes. Quality factor measures the quality of service being provided. It is a factor of a number of things like effective utilization of bandwidth, intelligibility, acceptability (audibility in voice for example), latency and jittering etc. However, we shall assume a simple model for evaluating quality factor. If Q0 is the quality factor of ATM cell for data communication, the quality factor, Q of other packet schemes for data communication shall be measured as: Q = (Q0/48) * (Packet size of the scheme). The model is justified as the quality of data communication usually increases with packet size. The quality measurement can be done in many other ways, but here we have assumed a very simple model. Although the model is simple, yet it is not far from real environment. This is because the quality measurement in the present model is relative.

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Binary ATM In a concept of binary ATM (Fig. 4), there are two ATM cells: one for data and related services, and another for voice and voice-like services. The cells will have 3 extra bits and they shall define the service type as below: 000 111

for time dependent services; and for time independent services.

The simple majority logic will be used by nodes and/or stations to identify the service types based on received 3 extra bits. It is illustrated as below: Received extra bits

Decision

000

Time dependent service

001 010

Time dependent service Time dependent service

011 100

Time independent service Time dependent service

101 110

Time independent service Time independent service

111 Time independent service We can compare binary ATM with conventional ATM in terms of relative efficiency and relative quality factor. Relative coding efficiency of binary ATM divided by that of the conventional ATM and is given by: n=

(1 + (5/48)) [(43/16) ((1/ y1 ) + (1/ y2 ) + 1)]

Similarly relative quality factor is the quality factor of binary ATM divided by that of the conventional ATM. Using the same simple mode as discussed earlier, if Q0 be the quality factor of any service under conventional ATM, the quality factor of time dependent and time independent services under binary ATM shall respectively be:

LM 48Q OP N y Q 0

1

and

LM y . Q OP N 48 Q 2

0

and hence average quality factor of binary ATM shall be : Q = [0.5{ 48Q0/y1 + y2Q0/48} ] and, therefore, relative quality factor shall be : Q/Q0 = 0.5[ 48/y1 + y2/48] The mathematical analysis is given in appendix-I. For a different set of fixed y1 and y2 the variation of relative efficiency and quality factor was calculated as shown in table (3). In binary ATM the data sizes for data and relative services; and for voice and related services have been proposed as 64 and 32 respectively as optimal choice. A three or four levels ATM may be possible with 6 extra bits in header of the cell. The encoding algorithm for service identification at source may be as below:

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First 3 extra bits Next 3 extra bits Service Type 000 000 type–1 000 111 type–2 111 000 type–3 111 111 type–4 Decoding algorithm at the receiving nodes and/or stations shall be simply majority decision based on received two individual groups of 3 bits out of 6 extra bits. However, in general with n times use of 3 bits in groups as above we can accommodate 2n numbers of ATM levels. And in each case majority logic will be used in the decoding. ATM and conventional packets can be compared with a single quanta of fixed frequency and quanta of any frequency respectively of quanta physics. We can have a multi quanta of a few frequency as a compromise i.e. we can suggest multi cell ATM. While suggesting the points to be remembered are: (1) cell size shall not be very high in order to avoid large latency which is unwarranted in time dependent services and (2) levels in multilevel ATM shall not be too high in order that the node processing becomes simpler.

Full Bytes Cell To make proposed ATM cell of full bytes in length, the extra bits may be used for error coding and security purposes. The error control will be an issue to reckon with mobile ATM particularly for data packets. In case of proposed binary ATM cell, three bits are used for service identification. To make the cell of full bytes in length, another five bits are required to be added. These five bits may be used for error control and security/coding purposes. The conventional ATM cannot be taken as an end but a means for achieving better transport technology for application independent network. Fixed (48 + 5) cell ATM may not be suitable for some of the future services - both time dependent and time independent. Therefore, multilevel ATM with proposed service identification encoding and decoding algorithms may be thought of an alternative and needs experimentation. Table 3: Efficiency and Quality factor of Binary ATM for different sets of y1 and y2 y1

y2

8 16 32 48

88 80 64 48

Relative Efficiency 80.7% 91.86% 98.03% 99%

48 bytes For CBR For VBR

Relative quality

5 bytes

Y1 <= 48 but fixed y > = 48 but fixed

3.92 2.3 1.4 1 Traditional ATM

5 bytes 5 bytes

3 bits 3 bits

CBR = Constant Bit Rate VBR = Variable Bit Rate Fig. 4: Illustration of Multilevel ATM

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Performance of Switching Techniques Generally, the comparison of techniques based on performance cannot be done because the performance depends on a number of factors, like number of nodes, total traffic, processing speed of nodes, packet size of the network, etc. However, a few observations can be made as: (a) For time sensitive service, till now there is no substitute to circuit switching on the ground of quality (b) for light and/or intermittent load under-interactive traffic, circuit switching is the most efficient (c) for heavy amount of data and efficient link utilization, packet switching is most suitable; (d) store-and-forwarding switching is not applicable to interactive real-time traffic; (e) for fast rate, packet switching may be applicable to interactive and real-time data transfer. (f) For multimedia services, ATM and proposed multilevel ATM is more appropriate based on QoS desired Performance of Proposed ATM in comparison with Conventional ATM In the networks, the techniques and systems are compared usually in terms of throughput and delay. In the above, the comparison was made in terms of quality factor and coding efficiency. In the present section we propose to compare in terms of the delay. Such a comparison depends on a number of factors, like number of nodes, total traffic, processing speed of nodes, packet size of the network, etc. However, we assume same simple network as in Fig. (2). We shall compare average transmission delay, and assume other components of the delay, namely propagation delay, node processing delay etc are negligible. We assume that: Tt = transmission delay of a conventional ATM cell over a link Tt’ = transmission delay of a proposed binary ATM packet (with six bytes header) over a link N = total number of packets constitute the message (the packets are both voice and data packets) Then the average transmission delay under the conventional ATM transport is given by: Dc = [K . Tt + ( N – 1) . Tt]/ N whereas the average delay of voice packets and that of the data packets under binary ATM transport are respectively: Dv = [K . Tt’ + (V – 1) Tt’}/N Dd = [K Tt’ + (N – 1) Tt’]/N

and

where V is the number of voice packets out of the total N packets. We perform numerical evaluation assuming each link capacity of 48 Kbps that gives Tt = 8.8 msec and Tt’ = 9 msec. The result is given in table (4) with corresponding curve at Fig. (5). From the results so obtained we find that : 1. so long the number of voice packets are less compared to the number of data packets, the advantage of sending voice packets at far less average delay is possible with proposed binary mode of transport

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2. when the number of voice packets are insignificant compared to data packets, the gain in terms of less average delay for voice packets is insignificant 3. when all the packets are data packets, the increase in average delay in the proposed binary ATM is very insignificant. Thus an overall conclusion of the suitability of the proposed binary ATM over conventional ATM is apparently established, although the same type of calculation over more generalized network needs to be done for achieving at a conclusive judgment. The advantages achieved at in the binary ATM as mentioned above are at the cost of the 1 byte higher header in the binary ATM. But the increased 1 byte header size in the binary ATM does not significantly reduce the coding efficiency.

Conclusion Behind the choice of 53 bytes size ATM cell, the motivation was to define a packet standard for all services. The same motivation worked behind the proposal of binary or multilevel ATM with the added consideration of reducing the delay for time sensitive packets. Our analysis with a simple network confirms in meeting the objective. No significant disadvantage is found in respect of binary ATM. Thus the binary ATM is the new challenge that needs some more experiments before final acceptance. The work critically analyzes the different switching techniques in terms of applications, link utilization and effectiveness. In addition to these common parameters of comparison, we have introduced a parameter of power making use of overhead bits and speed to compare the techniques. The power based comparison as usual but with clearer terms shows that the size of the packet is the critical to derive maximum gain in applying packet switching for data transport. We have also proposed a multilevel ATM cell switching and showed with analytical derivation that based on quality requirement and coding efficiency, variant ATM could be tried out. Further research is required to support the ideas. We also note that the implementation of binary ATM on the stated logic fails only if two or more bits of the proposed extra header bits for service identification are in error. The probability of two bits in error is: 3C α(1 – α)2 1 where α is the bit error rate. For common maximum bit error rate of 10–3, the probability is not so significant. Table 4: Numerical result in msec V

Dc

Dv

Dd

10

9.5920

1.7100

9.8100

20

9.5920

2.6100

9.8100

30

9.5920

3.5100

9.8100

40

9.5920

4.4100

9.8100

50

9.5920

5.3100

9.8100

60

9.5920

6.2100

9.8100

70

9.5920

7.1100

9.8100

80

9.5920

8.0100

9.8100

90

9.5920

8.9100

9.8100

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Average delay in ms

10 Average delay for conventional case

8 6

Average delay for voice packet

4 2 0 1

16

11

16

21

26

31

36

41

46

51

56

61

66

71

76

81

86

No. of voice packets out of 100 total packets

Fig. 5: Delay Comparison

APPENDIX-I We assume a message of p bytes. The message contains 50% VBR services and 50% CBR services.

Under conventional ATM The number of cell required = p/48 (or next higher integer). The coding efficiency (A) is therefore:

p 1 = ( p/48) * 5 + p 1 + (5/48) Under proposed Binary ATM The number of cells required for CBR Services = p/2y1 (or next higher integer ) The number of cells required for VBR Services = p/2y2 ( or next higher integer ) where y1 = information field bytes in proposed binary ATM cell for CBR services y2 = information field bytes in proposed binary ATM cell for VBR services. The coding efficiency (B) =

p ( p/2 y1 + p/ y2 ) (5 + 3/8) + p

Relative efficiency = (B/A) =

[1 + (5/48)] [(43/16) (1/ y1 + 1/ y2 ) + 1]

REFERENCES 1. 2. 3. 4.

Andrew S. Tanenbaum, Computer Networks, Prentice Hall of India. 1988. William L. Schweber, Data Communication, McGraw Hill International, 1988. Torub & Schilling, Principles of Communication Systems, McGraw Hill Pub. Co. 1986. James Martin, Computer Networks and Distributed Processing, Prentice Hall International, 1981.

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5. 6. 7. 8. 9. 10. 11. 12. 13. 14. 15. 16. 17.

205

Lyun A. Denoia, Data Communication, CBS Publishers, 1989. A. Bruce Carbon, Communication Systems, McGraw Hill Pub. Co. 1986. Stalling, Data & Computer Communication, Mac Million Pub. Co. Taub and Schilling, principle of Communication System, Tata McGraw Hill, 1991, Ch. 16. U. D. Black, Data Communication and Distributed Networks, Prentice Hall, 1993, Ch. 7. William Stallings, Data / Computer Communication, Addision Wesley, Ch. 12. V. Ahuja, Design and Analysis of Computer Communication Networks, McGraw Hill, 1982, Ch. 3. William Pugh and Gerlad Bayer, Broadband Access : Comparing alternatives, IEEE Communication Magazine, Aug, 1995, pp.34-46. J.P. Coudruse, General Principles of ATM, L ‘e’cho des Pechrches, English issue, 1992, pp. 5-18. John D. Hunter and William W. Ellington, ISDN: A Customer’s Perspective, IEEE Communication Magazine, Jan, 1996, pp. 20-22. Roger Levy, High Speed and Flexible Switching with ATM. Express Computer, Special issue networking, Sept., 1995, pp. 7-16, India. M. De Prycker et al., BISDN and the OSI Protocol Reference Model, IEEE Network, March, 1993, pp. 10-15. Borko Furht et al., Design issues for interactive television system, IEEE computer, May, 1995, pp. 25-38.

Application layer protocol UDP

TCP Gateway protocol

Address mapping protocols ARP, DCHP

IP and ICMP

Data link layer Physical link layer

(a) The TCP/IP (Internet) Model HTTP

FTP

SMPT

HTTP

TCP

ICMP

FTP

SMPT

TCP

IP Data link layer of IEEE 802.3 (Ethernet) Electro mechanics of IEEE 802.3 (Ethernet)

(b) Layering relationship between protocols

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TCP

Transport

IP

Network

Physical network/ ethernet etc.

Datalink and physical

(c) TCP/IP and OSI relation (TCP/IP may be either 3 layers, when physical network is taken as one layer or 4 layers, when the data link layers and the physical layer of the physical network are taken as two layers, protocol but they are same and one. In analogy with the OSI reference, thus Internet node is made of IP and lower layers whereas the Internet full host and station are made of TCP, IP and other lower layers) Fig. 18: TCP/IP in different angles

5.7 TCP/IP PROTOCOLS TCP/IP (Fig. 18) is actually made of two protocols - TCP and IP. That is why TCP/IP is often known as the Internet protocol suit. TCP and IP are the two protocols in the suite. Although, TCP/IP is a four-layered protocol. Layer 1 is the network interface services layer. It corresponds to layers 1 and 2 of the OSI/ISO protocol. It provides services relating to MAC (medium access controls), device drivers, physical medium, physical attachment and physical signals. In the layer, datagrams are packaged into frames. Layer 2 is the internet protocol (IP) layer. This layer corresponds to layer 3 of the OSI/ISO protocol. It provides routing of the datagrams as units of data. Logically, they are packets. The service is connectionless. Thus switching is a datagram service rather than virtual circuit switching. IP provides the basic service of getting the datagram to their destinations. It provides this service in best effort protocol. IP receives the TCP packets from the upper layer TCP, and then forms it own packet known as IP packets. Each IP packet is associated with several IP headers. Data pack 1

Data pack 2

Data pack 3

(a) Say a three bytes original message is fragmented into three data pack each of one byte ↓ One Original Data Pack is sent to TCP layer from higher layers TCP headers ↓ TCP headers TCP Packet (= Data Pack + TCP headers added by TCP layer) is formed by the TCP layer and then it is sent to IP layer.

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TCP headers

207

IP Headers

IP Packet (= TCP packet + IP headers added by IP layer) is formed by the IP layer and then it is released to the Physical Layer. (b) Illustration of IP packet formation. Three such IP packets are done to send the whole message under the illustration. 0 3,4 7,8 15,16 31 (bit) Version (4)

Length(4)

Type of service (8)

Total length (16)

Identification (16) Lifetime (8)

Flags (3)

Protocol (8)

Fragment offset (13)

Header checksum (16)

IP source address (32) IP destination address (32) Options (variable)

……

…….

Padding (variable)

(c) IPv4 Headers Fig. 19: TCP/IP pack

5.7.1 IP Header Description The 4-bit version field is used to indicate IP version that is being used in the IP datagram. The current version is IPv4 (IP version 4). IPv6 (IP version 6) is emerging as a standard for next generation IP datagrams. The 4-bit length field, which is also known as Internet Header Length (IHL) contains information about the number of 32-bit words in the header of the IP datagram. The minimum number of headers in any IP datagram is five 32-bits words or 20 bytes, and this happens when there is no option and padding fields. In this case, the IHL field contains 0101. As the IHL field can have maximum value 1111 (=15), the maximum number of headers in any IP datagram will be fifteen 32-bit words or 60 bytes. As the number of headers will be in multiple of 32-bit words or 4 bytes words, the headers that an IP datagram can carry will be one of the following sizes: 20, 24, 28, 32, 36, 40, 44, 48, 52, 56, 60. The 8-bit type of service or TOS field indicates if any special type of service the IP datagram should receive. The indication of special service offers priority or precedence given to the particular packet or packets over the other packets, and enables the routers to choose appropriate path for the transport of the packet or packets. In past, the field was largely not used. The TOS field splits into two sub-fields: 5-bits D/T/R/C/R (low Delay/high Throughput/ high Reliability/low Cost/Reserved) sub-field and 3-bit precedence sub-field (Fig. 20). The precedence is absolute and not relative. An IP datagram for a regular packet with no special and no priority will have D/T/R/C/ sub-field set as 0000 and precedence sub-field set as 000 Consider a multimedia packet which may need a higher throughput and highest level precedence. For this multimedia packet, IP datagram will set D/T/C/R sub field as 0100 and precedence sub field as 111. The TOS field is used by routers to choose routing path. If a regular datagram that needs a normal service reaches a router, the router selects a normal

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route as per outing strategy. But when say a datagram requiring minimum cost reaches a router, the router will select a path with minimum number of hops. Similarly for a datagram with higher reliability, the router will select a more reliable (lower BER) path. Likewise for a datagram with higher throughput, a path with higher link capacity needs to be selected. The details of the TOS field are available with RFC-2474, RFC-2475 and RFC-2430. A few examples of use of TOS field for the different services are given in the table (VIII). Precedence bit

Precedence bit

Precedence bit

Delay bit

Throughput bit

Reliability bit

Cost bit

Reserved bit

0

1

2

3

4

5

6

7

Fig. 20: Bits of TOS field

Table 8: TOS field for different services Service

Delay

Throughput

Reliability

Cost

bit

bit

bit

bit

Remarks

FTP data

0

1

0

0

Service requires high throughput only

FTP control

1

0

0

0

Service requires low delay only

SNMP

0

0

1

0

Service requires high reliability

SMTP data

0

1

0

0

Service requires high throughput

SMTP Command

1

0

0

0

Service requires low delay

DNS TCP query

0

0

0

0

Service is normal type

DNS UDP query

1

0

0

0

Service requires low delay

ICMP query/error

0

0

0

0

Normal service

Remote Login/Telnet

1

0

0

0

Service requires low delay

DNS zone transfer

0

1

0

0

Service requires high throughput

TFTP

1

0

0

0

Service requires low delay

BOOTP

0

0

0

0

Normal service

The 16-bit total length (TL) indicates the total number of bytes that the datagram carries including the headers. Thus the maximum number of bytes that a datagram carries is 1111111111111111 or 65,535 (= 216–1) bytes. The IHL counts only the header fields in units of 32-bits or 4-bytes words. But the total length counts the entire packets in unit of bytes. The

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entire IP datagram is a variable one and so also header fields. By IHL, the receiver identifies the number of header bytes in the packet. The receiver by subtracting the header bytes from the total bytes identifies the beginning and the end of data, which is actually the TCP pack. The following table (9) illustrates this for maximum data (TCP pack) sizes under valid sizes of IHL. As per RFC-791, the IP end stations must be capable of handling 576-byte datagram. Table 9: Valid maximum sizes of data(payload) in IP datagram 4-bits IHL field

Valid IHL in bytes (multiple of 4)

Maximum data or TCP pack size = (65535 – IHL) bytes

16-bits TL field

0101

20 (minimum)

65,515 (maximum)

1111111111111111

0110

24

65,511

1111111111111111

0111

28

65,507

1111111111111111

1000

32

65,503

1111111111111111

1001

36

65,499

1111111111111111

1010

40

65,495

1111111111111111

1011

44

65,491

1111111111111111

1100

48

65,487

1111111111111111

1101

52

65,483

1111111111111111

1110

56

65,479

1111111111111111

1111

60 (maximum)

65,475 (minimum)

1111111111111111

BOX 8

SOLVED PROBLEMS 1. As per RFC 791, the minimum TL size is 576 bytes. For this what will be size of possible payloads? Answer is given in table (1): Table 1 4-bits IHL field

Valid IHL in bytes (multiple of 4)

Maximum data or TCP pack size = (576 – IHL) bytes

0101

20 (minimum)

556 (maximum)

0110

24

552

0111

28

548

1000

32

544

1001

36

540

1010

40

536

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1011

44

532

1100

48

528

1101

52

524

1110

56

520

1111

60 (maximum)

516 (minimum)

2. Let us define the header coding efficiency of IP datagram as [payload/(payload + header)]. Find coding efficiency for maximum and minimum possible/permitted payload. Answer: The maximum permissible payload is 65515 bytes when the headers are 20 bytes. In this case the coding efficiency = 65515/ (65515 +20) = 0.9996 The minimum permissible payload is 516 bytes when the headers are 60 bytes. In this case the coding efficiency is = 516/ (516 + 60) = 0.8958 ≅ 0.9 3. What possible inference can you draw from the results of question (2) above. Possibly, the RFC aimed to keep the coding efficiency >0.9 4. List possible maximum and minimum size of the data (payload) in the IP datagram when TL field is made of all 15, 14, 13, 12, 11 and 10 left most bits are 1s. What does happen when 9 left most bits are all 1s? Answer is given in the Table 2 TL field

Bytes in whole IP datagram

0111111111111111

32767

0011111111111111

16383

0001111111111111

8191

0000111111111111

4095

0000011111111111

2047

000001111111111

1023

Maximum and minimum payload respectively correspond to 20 bytes and 60 bytes headers

Payload in bytes

Maximum

32767– 20 = 32747

Minimum

32767 – 60 = 32707

Maximum

16383 – 20 = 16363

Minimum

16383 – 60 = 16323

Maximum

8191 – 20 = 8171

Minimum

8191 – 60 = 8131

Maximum

4095 – 20 = 4075

Minimum

4095 – 60 = 4035

Maximum

2047 – 20 = 2027

Minimum

2047 – 60 = 1987

Maximum

1023 – 20 = 1003

Minimum

1023 – 60 = 963

When left most bits are all 1, the total bytes in the IP datagram will be = 511. But as per RFC 791, the minimum size is 576 bytes. Hence it will correspond to an invalid IP datagram

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The identification field is a 16-bit header. Internet connects the different types of networks. The different networks will have the different packet size. For example Ethernet packets may be in the rage of 64 to 1518 bytes, a MAN may have maximum packet size of 1500 bytes, an FDDI packet can be of 4472 bytes and a token ring packet may be typically 17,800 bytes. When a packet of, say 1500 bytes arrives at a node or router and the same is required to be transferred or routed to a network say having maximum packet size of 512 bytes, it is required that the arriving packet of 1500 bytes is to be fragmented into several smaller packets each having a size of 512 bytes or less. Each part or fragmented packet of a particular original must be given some identification so that the receiver can recombine all the related fragments in packets. Same identification to all fragments of a packet ensures that the wrong fragments are not combined to reconstruct the original packet in the receiver or otherwise speaking the reassembling of the correct fragments is done to reconstruct the original packet at the receiver. The identification field assigns a single group number to all the fragments of a packet being fragmented. The 3-bits flags are used to control and convey information in regard to fragmentation. The use of the flags bits is shown in the table (10).The left most bit is reserved. The middle bit known as DF (Don’t Fragment) provides a flexibility to the sender. If the sender likes that the packet must not be fragmented even if it does not reach the destination, the sender can construct IP datagram with DF bit set to 1. When DF bit is set to zero, the fragmentation is allowed. When a node or routers breaks a packet into several fragments, MF (More fragment) bit of all the fragments except the last one is set to 1. The MF bit is set to 0 in the last fragment. By this receiver is made to known about the last fragment of a packet. Table 10: Illustration of fragmentation Bit position

Flag name

Function

Bit 0 or the left most bit

Reserved

Set to 0 on transmit, and ignored on receive

Bit 1 or middle bit

DF (Don’t Fragment)

Set to 0 while allowed to fragment and to 1 when not allowed to fragment

Bit 2 or right most bit

M (More fragment)

Set to 1 for all fragments but set to 0 for the last fragment

The 13-bits fragment offset indicates the how to insert fragments in the receiver’s buffer in order to reconstruct the original datagram. The field is measured in unit of 8 bytes. But why? There are only 213 offset values but the datagram can be as big as 216 bytes. So each fragment must be multiple of 8 bytes long as 216/213 = 8. The requirement applies only to the data field (or TCP pack) of the original packet as because the header fields of original IP by its right will also be the header fields of each fragment, and one can not expect the header fields be divisible by 8 always. So a fragment’s data field will always be multiple of 8 bytes long, but the total length of fragment (total length of the fragment = original packet header length + data length of fragment) will be multiple of 4 bytes long. However the total length of fragment may be made multiple of 8-bytes long by padding for the purpose of storage at the receiver. The multiple of 8-bytes long is only applicable to data field and not to the whole datagram. In the worst case, the data field may be just 8-bytes long. So for the permissible

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header lengths, the worst case fragmented packet sizes would be 28, 32, 36, 40, 44, 48, 52, 56, 60, 64 and 68 bytes long. But IP does not run over the networks that have MTU (Maximum Transmission Unit) smaller than 68 bytes (see it is the highest fragment size in worst case). Thus for IP datagram with 20-bytes IHL, the worst case refers to fragments having data field multiple of (68-20 =) 48 bytes long. 48-bytes are multiple of 8-bytes. Consider an original IP packet with the 20-bytes minimum IHL and the maximum TL as (65535-20 = ) 65515-bytes long. In worst case such an original IP packet may be fragmented into (65515/48=) 1364 fragments with remaining 43 bytes left over as 1365th fragment. The 1365th fragment will be made 68 bytes long by padding. Problem: Assume that an original IP datagram with TL of 1500 bytes and IHL of 20 bytes requires to be transported over a network that supports MTU of 512 bytes (Fig. 21). Propose a fragmentation scheme for router R.

IP with 1500 bytes

Router

Network with MTU 512 bytes

Fig. 21

The original IP packet with 1500 bytes has a data field of 1500 – 20 = 1480 bytes. Network with MTU of 512 bytes will carry fragments each with data field of maximum 512 – 20 = 492 bytes. This is because each fragment must have 20 bytes header field as in original IP packet. But 492 is not divisible by 8. Nearest lower number of 492 that is divisible by 8 is 488. Thus each of the fragment will have a data field of 488, and number of fragments will be 1480/ 488 = 3 with remainder of 16 bytes that will need one more fragment. Thus total of four fragments will be needed with three fragments each of size 488 + 20 = 508 bytes and fourth fragment of 16 + 20= 36 bytes. However each fragment will now be made multiple of 8-bytes long by padding of 4 bytes in which case each of the three larger fragments become 508 + 4 = 512 bytes and last fragment becomes 40-bytes long. Now the fragments will have the fragmentation characteristics as shown in Fig. (22). OS stands for fragment Off Set field. The first fragment bears OS field as 0, the second fragment has OS field as 64 (this means total 64 numbers each of 8-bytes fields are there in the previous fragment), the third fragment bears OS field as 128 (which means previous fragments has total 128 numbers each 8-bytes fields) and so on. Such a numbering of offset field gives advantage in storage at receiver. Fragments move in the networks just like datagram. So the fragments may reach at the receiver out of sequence. If offset number was given like serial sequence number, then looking into the fragment number reassembling might have been done, but storing of the fragments at the receiver would have not been so easy. Consider the arrival of the fragments at the receiver as in Fig. 23). We find: Fragment with off set field 128 arrives FIRST Fragment with off set field 0 arrives SECOND Fragment with off set field 192 arrives THIRD Fragment with off set field 64 arrives LAST.

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The receiver based on the offset value places the fragments in the buffer location as in Fig. (23). The offset value refers to the first location of buffer for the storage. By the process, the fragments are arranged or stored in order in the buffer. This is called pigeon holding. Fragment 1 ID = xx M=1 OS = 0 TL = 512

Fragment 2 ID = xx M=1 OS = 64 TL= 512

Fragment 3 ID = xx M=1 OS = 128 TL = 512

Fragment 4 ID = xx M=0 OS = 192 TL = 40

Fig. 22

Fragment 3 reaches first with OS as 128 and M as 1

Fragment 1 reaches second with OS as 0 and M as 1

Fragment 4 reaches third with OS as 192 and M as 0

Fragment 3 reaches last with OS as 64 and M as 1

Say each location is of 8 bytes in the receivers buffer Location 0 to 63 Location 64 to 127 Location 128 to 191 Location 192 onwards

Fig. 23

The 8-bits time to live (TTL) field is used to ensure that an IP packet does not persist in the Internet forever. If packets persist in the network forever, the congestion is bound to occur. So TTL in its own right provides a mechanism to avoid congestion in the Internet. The maximum TTL field is 255. Early days it was taken as 255 seconds. Presently the TTL field is taken a hop counter. Thus an IP packet can move the maximum 255 hops in the Internet. The transmitting station sets the TTL field of the datagrams. When the datagram passes through routers or nodes, each router or node decreases the TTL field by default value set by the router or network administrator; the minimum value of default being one. If a router or node receives a datagram with TTL field as 0 and if the datagram not then reaches its final destination, the datagram is discarded by that router or node. The 8-bits protocol field defines the protocol in which the data is encapsulated. A list of a few permissible protocol is given in the table (11). For example when the IP datagram carries a protocol field with 6, it means the data is a TCP pack. Table 11: Protocol as defined in the protocol version field in the IP header Protocol

Protocol Version Field in the IP header

ICMP (Internet Control Message Protocol)

1

IGMP (Internet Gateway Message Protocol)

2

GGP (Gateway-to-Gateway Protocol)

3

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IP in IP (encapsulation)

4

TCP (Transmission Control Protocol)

6

EGP (Exterior Gateway Protocol)

8

IGP (any private Interior Gateway Protocol)

9

UDP (Users’ Datagram Protocol)

17

ISO TP4 ( ISO Transport Protocol class 4)

29

IPv6 (Internet Protocol version 6)

41

IPv6-Routing Header

43

IPv6- Fragmented Header

44

IDRP (Inter-Domain Routing Protocol)

45

RSVP (Reservation Protocol)

46

GRE (General Routing Encapsulation)

47

MHRP (Mobile Host Routing Protocol)

48

ESP (Encapsulation Security Payload for IPv6)

50

AH (Authentication Header for IPv6)

51

SWIPE (IP with Encryption)

53

NARP (NBMA Address Resolution Protocol)

54

MOBILE (Mobile IP)

55

ICMP for IPv6

58

No Next Header for IPv6

59

Destination Options for IPv6

60

DGP (Dissimilar Gateway Protocol)

81

IGRP (Interior Gateway Routing Protocol)

88

OSPF (Open Shortest Path First)

89

PIM (Protocol Independent Multicast)

103

IPX in IP

111

Unassigned

130-254

Reserved

255

(RFC-1700 or its successor provides a complete list of protocol field) The header check sum field is 16-bits. The field is to check error on the headers fields of the IP datagram. The check sum is calculated using 1’s complement addition rule, which was discussed at length previously at the section of error control. The checksum is computed by taking 16-bits words from header fields. The checksum field ensures checking of error only at link level error on header fields. TCP as we will see later will perform end to end error control, which will take care of error on data field. That is why at the IP level error checking at the data

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field is not done. The IP header checksum is computed at each router/node or on hop-to-hop basis since each router/node changes the TTL field as well during fragmentation the IP headers (like identification, M flag, off set fields etc) are changed. IP header has two address fields: source IP address and destination IP address. These are used to route the datagrams. IP uses 4 modes of addressing for this purpose. These are known as A- class, B-class, C-class and D-class. The difference in classes is due to how many bits are used for network identification and how many bits are used for host identification. The different address formats under different classes are illustrated in Fig. (24). The IP addresses are 4-bytes and they are usually mentioned as four sets of dot-separated octets. IP address first-octet ranges are shown in table (12). For A class, most significant bit of first octet is always 0. Therefore next seven bits are used for network identification, and this means that only 27 = 128 networks can be of A-class. The hosts may be 224 . For class-B address, first two bits of first octet must be 10, and this makes first octet to be 128 to 191. 127 is kept reserved for special purposes. Total networks and hosts under class B are respectively only 214 and 216. For class-C address, first three bits of first octet is 110. This means the first octet will be from 192 to 255. Class-C address has only 221 networks and only 28 hosts. The minimum hosts under different addressing mode are: Class A = 27 × 224 = 231 Class B = 214 × 216 = 230 Class C = 221 × 28 = 229 The address scheme under class-A to class-C is for unicast communication. The other addressing schemes are for multicast communication. Class-D and Class-E belong to this scheme. In class-D, all the four octets are used to identify the group of nodes designated to receive a multicast. Class-D addresses do not specify the network. Class-D addresses are in the ranges of 224.0.0.0 to 239.255.255. Class-E addresses range from 240 to 255 in the first octet, and are used for experimentation. 1

2

3

Class A

0

Class B

1

0

Netid

Class C

1

1

0

Class D

1

1

1

0

Class E

1

1

1

1

4

8

Netid

16

24

Hosted Hosted Hosted

Netid Multicast address 0

Reserved for future use

(a) Structure of IPv4 Addressing (netid = network identification, hostid = host identification) Different classes

Starting address

Last address

Class-A

0.0.0.0

127.255.255.255

Class-B

128.0.0.0

191.255.255.255

Class-C

192.0.0.0

223.255.255.255

Class-D

224.0.0.0

239.255.255.255

Class-E

240.0.0.0

247.255.255.255

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(b) Address space under different classes. (Bold space is for netid and rest is for hostid; except in case for class-E. In case of class-E, the whole of the address space id undefined) Class

Range

A B C D E

1.0.0.0 to 126.254.254.254 128.0.0.0 to 191.254.254.254 192.0.0.0 to 223.254.254.254 224.0.0.0 to 239.0.0.0 240.0.0.0 to 255.0.0.0 127.0.0.0 to 127.254.254.254 255.255.255.255

Number of Networks 126 16,000 2,000,000 NA NA NA NA

Number of Hosts per Network 16,000,000 64,000 265 Multicast Test Loop back Broadcast

(c): Details in different addresses Fig. 24: IPv4 addressing scheme

Table 12: IP Addressing range under different classes

Class A Class B Class C

Range of first octet in IP address

Maximum number of network that can be addressed

Maximum number of hosts or nodes per network that can be addressed

Maximum number of total nodes or hosts that can be addressed under the addressing class

1-126 128-191 192-223

126 16384 2097152

16777214 65534 254

2113928964 1073709056 532676608

Maximum number of hosts or nodes per network is in Class A and it is about 256 times of that of the Class B and about 216 times of that of the Class C. Maximum number of hosts or nodes per network in Class B is about 256 times of the that of Class C. Maximum number of total hosts or nodes in Class A is about two folds of that of Class B and four folds of Class C; and that of Class B is about two folds of that of the Class C. The logical conclusion is that Class A addressing is for Larger networks, Class B for Medium networks and Class C for Small networks. So if your network is large, you can apply for Class A address, and for medium network, Class B and for small network, Class C address. There is a trade off among the different address classes. The trade off is: the number of network versus the number of hosts or node per network. Class A has least number of network address but highest number of host/node address per network, class C has highest number of network address but least number of host/node address per network. Class B falls in between these two extremes. The special addressing schemes are as below : 1. The address 0.0.0.0 is used to mean this host on this network. It actually refers to Internet itself.

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2. 255.255.255.255 is used as a broadcast packet for all networks. Network ID with any but host IDs with all 1’s or 255 is used for directed broadcast to the network. For example 10.255.255.255 is used for broadcast message to the class A network with ID 10. Broadcast addresses are destination addresses. 3. Packets with first octet as 127 are used for network testing. Address with network ID as 127 and host ID as any is used for internal host loop back address. Thus loop back addresses are 127.0.0.0 to 127.254.254.254 4. An entire network is specified by providing only the network identification and with 0s in all other octets such as: 124.0.0.0 for a class A network, 129.155.0.0 for a class B network and 200.127.110 for a class C network. 5. A specific host on this network is specified by network ID as 0s and with required host ID. This is used as a source address. Question. The different classes of addresses may have been identified by first two bits of the 32 bits as below: 00 for class A 01 for class B 10 for class C 11 for class D Compare the advantages and disadvantages of the proposed scheme of class identification with that of the IPv4 addressing.

Unregistered or Private Address Space For having global Internet connectivity, the hosts and networks on which hosts are connected must have distinct IP addresses. Till 1999, the IP addresses were assigned by IANA (Internet Assigned Numbers Authority) and now the job is being carried by ICANN (Internet Corporation for Assigned Names and Numbers). If any network is not connected to the Internet, the network even then can use some address space kept reserved for the purpose in RFC 1597. This address space is known as unregistered or private address space, which is listed in the table 13. When the network needs to connect to the Internet, the network can continue with the unregistered address. There are systems like NAT (Network Address Translator) gateway that will translate private unregistered addresses to public registered addresses. Table 13: Private unregistered address space Address class

Private unregistered address space

Number of networks

Class A

10.x.x.x to 10.x.x.x.

1

Class B

172.16.x.x to 172.31.x.x

16

Class C

192.168.0.x to 192.168.255.x

256

Subnet Mask Subnet mask is a special addressing scheme. It is used for two purposes: to show the class of addressing in use, and to divide a network into different sub networks to control traffic. For first purpose, a subnet mask determines which part is for network ID and which part is for hosts ID. A subnet mask for A class network is: 11111111.00000000.00000000.00000000

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(=255.0.0.0). All 1s indicate Network ID and all 0s indicate host ID. This mask is full mask or default mask or implicit mask. However for the purpose of masking it is not always that full or default mask be used. The Host ID part that is in the hand of the organization may be used for masking also (This is illustrated in the example of Fig. 25). For second purposes, subnet mask is used to divide the network within network administrator. For example: the entire third octet of class-B may be designated for subnet ID that would be 11111111.11111111.11111111.00000000 (=255.255.255.0). Another example: of subnet ID that under class - B could be: 11111111.11111111.11110000.00000000 (=255.255.240.0). In this example 4 left hand side bits of the third octet are used for subnet networks ID, whereas other 4 bits plus last octet is kept for host ID. The possible class full subnetting in class A, B and C network is listed in table (14). Table 14: Possible classful subnetting Possible Classful Subnetting breakdown under Class “A” addressing Mask size

Number of

Number of Hosts

Decimal Mask in Bits Network

8 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30

16,000,000 4,000,000 2,000,000 1,000,000 512,000 250,000 128,000 64,000 32,000 16,000 8,000 4,000 2,000 1,000 510 254 126 62 30 14 6 2

1 4 6 14 30 62 126 254 510 1,000 2,000 4,000 8,000 16,000 32,000 64,000 128,000 256,000 512,000 1,000,000 2,000,000 4,000,000

255.0.0.0 255.192.0.0 255.224.0.0 255.240.0.0 255.248.0.0 255.252.0.0 255.254.0.0 255.255.0.0 255.255.128.0 255.255.192.0 255.255.224.0 255.255.240.0 255.255.248.0 255.255.252.0 255.255.254.0 255.255.255.0 255.255.255.128 255.255.255.192 255.255.255.224 255.255.255.240 255.255.255.248 255.255.255.252

Possible Classful Subnetting Breakdown under class “B” addressing Mask size

Number of

Number of Hosts

16 17 18

64,000 32,000 16,000

1 2 4

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Decimal Mask in Bits Network 255.255.0.0 255.255.128.0 255.255.192.0

NETWORK AND INTERNET TECHNOLOGY

19 20 21 22 23 24 25 26 27 28 29 30

8,000 4,000 2,000 1,000 510 254 126 62 30 14 6 2

6 14 30 62 126 254 510 1,000 2,000 4,000 8,000 16,000

219

255.255.224.0 255.255.240.0 255.255.248.0 255.255.252.0 255.255.254.0 255.255.255.0 255.255.255.128 255.255.255.192 255.255.255.224 255.255.255.240 255.255.255.248 255.255.255.252

Possible Classful Subnetting breakdown under class “C” addressing Mask size 24 25 26 27 28 29 30

Number of 254 126 62 30 14 6 2

Number of Hosts 1 2 4 6 14 30 62

Decimal Mask in Bits Network 255.255.255.0 255.255.255.128 255.255.255.192 255.255.255.224 255.255.255.240 255.255.255.248 255.255.255.252

The use and operation of mask in IP addressing and routing could be further explained with an example network topology of Fig. (25). In the example, three Ethernets (I, II, and III) are connected using routers. There are two hosts on Ethernet I, and one each on Ethernets, I and II. Assume the IP addresses of the networks, Ethernet–I = 139.39.1.0/24 Ethernet–II = 139..39.2.0/24 Ethernet–III = 139.39.3.0/24 Thus all the interfaces and hosts connected to Ethernet–I will have IP address with first 24 bits as 139.39.1; similarly all the interfaces and hosts connected to Ethernet-II and EthernetIII will have IP address with first 24 bits as 139.39.2 and 139.39.3 respectively. 24 refers to mask. The mask of network is 24 numbers of 1 and followed by 0: 11111111 11111111 11111111 00000000. In the figures, addresses of the hosts and the router interfaces as shown are: Router-I interface of Ethernet-I = 139.39.1.1 Host-A = 139.39.1.2 Host-B = 139.39.1.3 Router-I interface of Ethernet-II = 139.39.2.3 Host-C = 139.39.2.1 Router-2 interface of Ethernet-II = 139.39.2 .2 Router-2 interface of Ethernet-III = 139.39.3.1 Host-D = 139.39.3.2

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The issue, now, is how a host will send the packets to other hosts. Hosts connected in a same Ethernet has the same 24 leading bits in address, whereas the hosts connected in different Ethernet has different 24 leading bits in address. When a node has a packet to send to a destination, it first checks whether the destination is in same network or not. This is done by mask, as illustrated bellow: Case-I. A to B communication: (Source) Host A's address: Host A's mask (24):

10001011 11111111

00100111 11111111

00000001 11111111

00000010 00000000

AND operation(X): 10001011 (Destination) Host B's address: 10001011

00100111 00100111

00000001 00000001

00000000 00000011

Host A's mask (24): AND operation(Y):

11111111 00100111

11111111 0000000

00000000 00000000

11111111 10001011

As X and Y are same, A shall understand that destination is on its network. Case-II. A to C communication: (Source) Host A's address: Host A's mask (24):

10001011 11111111

00100111 11111111

00000001 11111111

00000010 00000000

AND operation(X): 10001011 (Destination) Host C's address: 10001011

00100111 00100111

00000001 00000010

00000000 00000010

Host A's mask (24): AND operation(Y):

11111111 00100111

11111111 00000010

00000000 00000000

11111111 10001011

X and Y are not same. So A will understand that C is on different network. Therefore, A will send data to router. 139.39.1.2 Ethernet-1 139.39.1.0/24 139.39.1.1

Host-A

Host-B

Router-I

139.39.1.3

139.39.2.3 Ethernet-II 139.39.3.0/24

139.39.3.1

Router-2

139.39.3.2

Ethernet-II 139.39.2.0/24

Host-C

Host-D

Fig. 25: example network for subnet masking

The subnet masking actually eases out the routing problems of the network to the extent that within the organization the routing is internal and not on the Internet routing.

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Classless Addressing and Routing In Ipv4, the address classes are A, B and C. Any organization may opt for any one of these classes based on the size. Say an organization is having 2000 hosts under it’s a local network. Thus organization can not opt for class C address, as because under one network address of class C, maximum 28 = 128 hosts can be addressed. The optimal option is then for class b when under a network address of class B, maximum 216 = 64 k hosts can be addressed. Here lies the problem. If the organization is opting for a network address of class, the organization will not be able to use full host address space; and the 64 k–2000 host address space will remain unutilized. To solve the problem, RFC 1519 proposed the concept of classless addressing. Unlike the fixed numbers of bits in the network address as in class a, b or c; the class less addressing proposes to have any bits in the network addressing. Thus a classless address for a network may be in decimal dotted of the form a.b.c.d/n where n indicates the number of leading bits of 32 bits address that constitutes the network address bits. The address space must always be in the block of 2n. Thus in the present example for the 2000 hosts, a block of 211 = 2048 address space is required. The 11 bits will identify the hosts. So the 21 bits (as 32-11 = 21) may be used for network addressing of the organization in the form of a.b.c.d/21. Again the organization may use 11 bits for subnetting. The organization’s network address in this example may be: 110010000.00010111.00011000.00000000 (bold and underlines bits for network address and remaining 11 bits for hosts) → 200.23.24.0/21 The classless addressing has unique application in ISP (Internet Service Providers). Say one ISP has four class C addresses as 192.16.88.0, 192.16.89.0, 192.16.90.0 and 192.16.91.0. Instead of using four addresses for the ISP, one address ISP can use for Internet routing. In the present example, the address 192.16.88.0 can be used as the address and other addresses can be thought of a group of this address and can be recognized for Internet routing by the subnet mask 255.255.252.0. How? When a packet with the address of 192.16.90.xxxxxxxx reaches, the AND operation between 255.255.252.0 and 192.16.90.xxxxxxxx will result to 192.16.88.0 that is the address of ISP. This makes the Internet routing simpler. The question remains how the mask was selected? The two things to be considered for such cases are:: (1) as we pointed out earlier, the address block must be in the poser of 2 and (2) the staring address of the group must be evenly divided by the block size (why? This is because then only the block will be of in the power of 2). We assumed block size of 1024 (10 bits). Thus, the network address bit is 22, and the mask is 11111111.11111111.11111100.0000000000 (=255.255.252.0). BOX 9

QUESTIONS 1. What shall be binding rules for using subnetting/subnet mask? The following conditions must necessarily be meet with: (1) Each and every host/ node must have unique IP address, (2) Each and every network must have unique classful IP address, (3) Each and every network segment must use unique IP subnet address space, (4) All hosts/ nodes on a network segment must use same subnet mask, and (5) All hosts/ nodes on the same IP original network must use same subnet mask 2. Give a pictorial illustration of benefit of classless Inter Domain Routing (CIDR) compared to conventional routing. The Internet users and hosts are growing exponentially. To cope with such growing demand, the Internet routers are supposed to enhance at the same space but that was not practically feasible. Therefore to solve the problem of routing, the CIDR was proposed.

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The following Fig. (1) illustrates the benefit of CIDR in application of routers. In the class full routing the routers is supposed to have routing information/ mesh of users of an Internet Service Provider (Fig. 1a), where as with CIDR routing the class full addresses of all users of the service provider are integrated into one aggregate class less address as 195.65.0.0/16. the notation/16 or prefix /16 denotes the number of the mask bits. 195.65.70.0

195.65.70.0

195.65.70.1 195.65.70.254

Internet service provide

195.65.70.255

195.65.70.1 195.65.70.254

Internet global router

195.65.70.255 This block must be in routing table

(a) Routing with class full address 195.65.70.0 195.65.70.1 195.65.70.254

Internet service provide

195.65.0.0/16

Internet global router

A single integrated address for routing of all hosts under service provider

195.65.70.255

(b) Routing with class full address with explicit mask of 16 bits (notation/16) i.e mask is 255.255.0.0 Fig. 1: Illustration of CIDR

3. (a) How is an IP address broken into several subnets/segments? (b) In CIDR, the prefix can be of any length unlike fixed 8, 16 or 24 in classful address namely for class A, B or C respectively. What is its use? The following table (1) illustrates how an IP address is broken into number of segments. Table 1 The given class B address is 150.150.0.0

Network Segment as class C

Mask

150.150.0.0

255.255.255.0

Segment-0

150.150.1.0

255.255.255.0

Segment-1

150.150.2.0

255.255.255.0

Segment-2

150.150.3.0

255.255.255.0

Segment-3

150.150.4.0

255.255.255.0

Segment-4

150.150.5.0

255.255.255.0

Segment-5

150.150.6.0

255.255.255.0

Segment-6

150.150.7.0

255.255.255.0

Segment-7

Note that the all eight subnets are identified by a single address 150.150.0.0; and the segments/subnets are identified by third octet in the address notation.

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(b) The prefix can be of any length. For example it may be/22. In above table (1), we have converted a class B address to several class C segments. But what is then its use? Say an organization has about 1000 hosts. So a single class C IP address is insufficient. On the other hand a single class B IP address is supposed to provide IP addresses to 216 = 32 K hosts. Thus allocation of class B address to the organization will lead to poor utilization of address space. In such a situation, with CIDR the organization may be assigned four class C segments with prefix of/22. Thus not only routing simplification, but also the increased address space utilization is the major application of CIDR scheme. In this case subnet mask will be 255.255.252.0. Say the registered address is 193.193.0. x for the organization. Now all the four segments are identified with one address as 193.193.0, but segments are identified as 193.193.0.0; 193.193.1.0; 193.193.2.0 and 193.193.3.0. 4. Consider a INTERNET GLOBAL router as in Fig. (2). It receives one aggregate address as 195.65.0.0/16 from one neighboring service provider and another aggregate address 195.65.25.0/24 from another neighboring. Now two packets one with destination address 195.65.25.38 and another with 196.65.1.35 reach the router. How will the router route the packets? 195.65.0.0/16

195.65.25.0/24

Internet global router

Fig. 2: For question (5)

CIDR uses the principle of longest match for routing. The packet with destination address 195.65.25.38 will be routed as per routing entry of /24 as its match is longest. The packet with destination address 195.65.1.35 will be routed as per routing of /16. 5. Enumerate the rules for special addressing. The special addresses basically uses the following rules: • Internal host loop back address for a host refers to Net Id= 127, and host Id. (Example address = 127.45.78.90 is a loop back address for a host number 45.78.90) • This host on this network or Internet itself refers to Net Id = all 0s and Host Id = all 0s. (Address = 0.0.0.0) • A particular node on this network refers to Net Id= all 0s and Host Id. This is for a source address. (Example address = 0.0.0.34 for a class C network with host address as 34) • Broadcasting to all nodes on this network (Internet) refers to Net Id = all 1s and Host Id = all 1s. This is only for destination address.( Address= 255.255.255.255) • Broadcast to hosts of a particular network refers to Net Id and Host id = all 1s. This is for destination address only. (Example address = 86.255.255.255 for a class A address with Net Id = 86) • Broadcast to a particular subnetwork refers to Net Id, Subnetwork Number and Host Id = all 1s. This is for destination address only. (Example address= 150.150.5.255 all hosts of segment number 5 of class B network with address 150.150) • Broadcast to all subnets and their hosts refers to Net Id, Segment number = all 1s and Host Id = all 1s. This is for destination address. (Example address= 150.150.255.255) 6. Write a flowchart that will determine the class of a given IP address.

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Option fields of IP The option field in the IP header is optional. The options field is made of option type field followed by option length field and then the option data bytes as shown in the Fig. (26). Option type byte (1 byte)

Length byte (1 byte)

Option data field

Copied flag

Option classes

Option number

Fig. 26: Option field format

The option type is the first byte of option header and it has three fields: 1 bit copied flag, 2-bits class field and 5-bits number field. When the copied flag bit is “0”, the option field is not copied in the fragments of the IP datagram. But when the copied flag is set to “1”, the option field is copied in all the fragments of the IP datagram. The different option classes are as under: 00 → Control 01 → Reserved 10 → Debugging 11 → Reserved whereas the numbers are assigned as shown in table (XV): Table 15 Number

Descriptions

Bytes that follow

0

End of option list

Followed by 0 bytes, Total 1 byte

1

No operation

Followed by 0 byte, Total 1 byte

2

Security

Followed by 10 bytes, Total =11 bytes

3

Loose source and record routing (LSRR)

Followed by variable bytes, MAXIMUM total = 255 bytes

4

Internet Timestamp

Followed by variable bytes, Maximum total = 255 bytes

7

Record Route (RR)

Followed by variable bytes, Maximum total = 255 bytes

8

Stream Identification

Followed by 3 bytes. Total = 4 bytes

9

Strict Source and Record

Followed by variable bytes,

Routing(SSRR)

Maximum total = 255 bytes

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The end of option list is of type 0 as its option type byte has a value 0 (00000000). It belongs to class “00” This option field is used as the end of all options but not the end of each option. However it does not necessarily coincide with the end of all IP headers as per IP header length field. This is due to the fact of requiring padding to make IP headers a multiple of 32 bits words. The end of option field may be copied, introduced or deleted on fragmentation. The no operation option field is used between options for the purpose of alignment for 32 bits words. The field has a type equals to 1 (00000001) and belongs to control class “00”. The field may be copied, introduced or deleted on fragmentation. The Security option field looks as shown in Fig. (27). The option belongs to control class. The field has a type 130 as its option type field carries a value of 130 (10000010). The length field carries a value 11 (00001011) as its has total 11 bytes. The other fields are as below: 10000010

00001011

SSS .... SSS (2 bytes)

Option type byte Length byte (1 byte) (1 byte)

CCC .... CCC (2 bytes)

HHH .... HHH (2 bytes)

TCC .... TCC (2 bytes)

Actual option data bytes

Fig. 27: Security option field of type 130.

SSS Field: This is 2-bytes security field that specifies different levels of security as illusrated in table (16) for a few. Table 16: Different codes for SSS field Code

Description

Code

Description

00000000 00000000

Unclassified

00110101 11100010

Reserved

11110001 00110101

Confidential

01001101 01111000

Reserved

10101111 00010011

Restricted

00010011 01011110

Reserved

01101011 11000101

Top secret

11000100 11010110

Reserved

10111100 01001101

MMMM

10011010 11110001

Reserved

01111000 10011010

EFTO

00100100 10111101

Reserved

01011110 00100110

PROG

10001001 10101111

Reserved

11010111 10001000

Secret

11100010 01101011

Reserved

CCC Field: This is also a 2-bytes field. It is called compartment field. The field contains all 0s when data is not compartmented. The other codes are with Defense Intelligence Agency. HHH: This a 2-bytes field known as Handling Restrictions field This code of the field is available from defense agency. TCC Field: The field is known as Transmission Control Code and is of 3-bytes in length. It is used to segregate data and to define control. It is also defense controlled. The field must be copied on fragmentation. The Loose Source and Record Routing (LSRR) has the options field as shown in Fig. (28). The LSRR allows the source of an IP datagram to define routing information to be used by

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routers/gateways to forward the datagram to its destination. This is also used to record the routing information. Thus the options field provides the alternative routes. The routing is specified by recording the IP addresses of the routers and gateways in the data area of the options field. The routers and the gateways may use the information provided in the route data to route datagram and may not use their own routing tables for the same. In the Fig. (28), it is seen that its copied flag is 1 and it belongs to control class with type number 2. Alternatively it may be called type 131 option field as the type bears a value 131 (10000011). The length header indicates the number of bytes in the options field. The pointer field indicates the bytes after which the beginning of the next IP address to be processed into the route data. As the IP address is 4 bytes, the pointer minimum value is 4. The pointer value is basically increases by 4 at each router/gateway. The route data is made of a number of IP addresses of specified routers/gateways. The pointer value can be the maximum number of bytes in the route data. When the pointer is greater than the length of the route data, the router/gateways will consider their routing information for further transporting the datagram. This is because at this instant the routing information provided in the route data is exhausted. The options field is called loose because the routers and gateways are permitted to use any routing information either the routing information specified in the route data or routers/gateways’ own routing information. 10000011 (1 byte)

Length (1 byte)

Pointer (1 byte)

Route data (variable bytes)

Option type header

Fig. 28: LSRR options field

The Internet Timestamp options field has the format as shown in Fig. (29). The option type header belongs to copied flag as “0”, debugging class (10) with number 4 (00100). The option field is, therefore, type 68 as the type header has a value of 68 (01000100).The 1 byte length header specifies the number of bytes in the options field but a maximum value of 40. The 1-byte pointer header indicates the byte at which the time stamp is to be recorded. The minimum value of pointer is 5 as the first time stamp record must start 5 bytes after the pointer (1 byte for overflow + flags, 4 bytes for Internet address).. The 4-bits over flow header is used to hold the number of IP addresses (that may be routers, intermediate nodes, gateways etc) that cannot record the timestamp for want of space. The 4 bits lags are used to define different modes of recording timestamps, and these are: 0 for storing only the time stamps as 4 bytes word 1 for IP address followed by time stamp 3 for adding time stamps if only the gateway IP address is found in a specified table. In the mode, the IP addresses that can record the time stamps are prior specified in a table. Time stamp is a 4 bytes or 32 bits word that records value for the number of milliseconds since midnight under universal time frame. If non standard time record is made, any time can be recorded but in that case highest order bit of the time stamp is set to 1. The Internet address refers to the IP address of the source that has started the option. The source must reserve the sufficient bytes for records of time stamps. The size of the option does not change as the datagram traverses over the Internet. If the space is exhausted, an ICMP parameter Problem may be sent to the source.

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01000100 (1 byte)

Length (1 byte)

Pointer (1 byte)

Overflow (4bits)

227

Flags (4bits)

Internet Address (4 bytes) Time Stamps (4 bytes) ……… ……… Fig. 29: Internet Time stamps

Layer 3 or the transmission control protocol layer corresponds to layer 4 of the OSI/ISO protocol. It is transport protocol. The layer serves in connection oriented mode and guarantees the delivery of data. TCP provides the service of breaking messages into datagrams at the source end, retransmitting any datagram, which has been lost or acknowledged negatively, and sequencing the datagrams, etc. Service in the layer is guaranteed by acknowledgement of receipt of data from the receiving side. If the acknowledgement is negative, the data is retransmitted, and if the acknowledgement is not received with in a time period known as time out period, retransmission of the packet is done. This layer also performs multiplexing and demultiplexing functions if required. As we mentioned earlier, the TCP layer makes TCP pack using the data received from the application or higher layers. Each TCP pack is made of TCP headers and fragmented data (Fig. 30). A TCP pack has a number of header fields as shown in Fig. (31). Original data or payload data Data is converted into several fragments Data fragment 1

Data fragment 2

Data fragment 3

Data fragments are converted into TCP packs TCP header

Data fragment 1

TCP pack 1 TCP header

Data fragment 2

TCP pack 2 TCP header

Data fragment 3

TCP PACK and TCP SEGMENT ate same and one

TCP pack 3 TCP header

Data fragment 4

TCP pack 4

Fig. 30: Conversion of original data into TCP packs

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Source Port (16 bits)

Destination Port (16 bits) Sequence number (32 bits) Acknowledgement number (32 bits)

Data offset (4 bits)

Reserved (6 bits)

U R G

A C K

P S H

R S T

S Y N

F I N

Checksum (16 bits) (TCP Header + Data)

Window (16 bits)

Urgent Pointer (16 bits)

Options (Variable)

Padding (Variable) Data (Variable) Fig. 31: TCP headers

5.7.2 Description of TCP headers Like IP, the TCP header is of minimum 20-bytes in length. The TCP header has the following fields: The 16-bits source port number of sending device. The port refers to a virtual circuit between two end-stations communicating parties. The port is also called socket or session. The concept of ports or sockets implies that more than one process can communicate over a session at any time between end-stations or nodes. TCP ports are documented in RFC 1700. A few examples are given in table (17). The 16-bits destination port refers to the port of the receiving device for communication of an application process. Table 17: TCP ports Port Number

Function

1

Multiplexing

5

RJE (Remote Job Entry) applications

9

Transmission discard

15

Status of the network

20

FTP data21FTP commands

22

SSH(Secure Shell)

23

TELNET applications

25

SMTP e-mail applications

37

Time transactions

43

Who is Protocol

53

DNS server applications

79

Finger protocol / Find active users’ protocol

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80

HTTP (Hyper Text Transfer Protocol)

93

Device Controls

102

SAP (Service Access Point)

103

Standardized e-mail services

104

Standardized e-mail exchanges

110

Post Office Protocol

119

Network News Transfer Protocol

139

NetBIOS applications/WINS

443

HTTPS secure WWW server

512

Berkeley commands

513

Login

543

Klogin (Kerberos Login)

544

Kshell (Kerberos Shell)

750

Kerberos Server

751

Kpasswd (Kerberos Password)

2105

eklogin (encrypted Kerberos login)

2049

NFS (Network File System)

229

The 32-bits sequence number( 0 to 4,294,967,295) field is used to assign a sequence number to each TCP pack. It is actually assigned to the first byte of the TCP message. The transmitter assigns the sequence number. The receiving on reading the sequence number of the packs ensures about whether all packs are received. Using sequence number, the packs received out of order are also placed back in order. Using sequence number the receiver also identifies the duplicate copy of the packs, if any received and accordingly takes the corrective measure. While IP sends the datagrams these may be routed through deferent links. This in turn may deliver the datagrams out of sequence at the receiver end. The datagrams are put in order by using sequence numbers assigned to them during transmission. This job is performed by the TCP layer. The sequence number and the acknowledgement number together ensure the reliable transport of TCP. TCP is reliable transport, whereas IP is unreliable transport. The 32-bits acknowledgement field is used to send the acknowledgement of the pack received correctly. The receiver performs this function. The receiver checks the sequence number of a received pack. If the pack is received without error, the receiver send an acknowledgement number using the sequence number of the pack to inform the sender that the pack has been received correctly. The 4-bits offset field is used to indicate the number of header fields in units of 32-bits words. The offset field is also known as header length or HLEN. If there is no padding and option fields in the header, the offset field will be 0101. This means there are five 32-bits words or 5 × 4 = 20 bytes header. This information, the receiver uses to determine the start of the data field so that data will be derived from the received pack.

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The flags, window and urgent pointer fields are used to connect and manage the TCP connection. The SYN (Synchronization), ACK (Acknowledgement) and FIN (Finish) flags are each a 1-bit flag. These are used to establish a TCP connection. TCP is a reliable and connection oriented protocol, the TCP session needs a virtual connection. This connection is established by a process called TCP handshake (Fig. 32). To establish a connection, the initiator or sender sends a TCP pack with the SYN flag set (i.e. SYN=1) and ACK flag set to 0. The calling program sends a TCP message through a port that has been allocated to it, like 21 for a FTP connection. The initial connection request TCP pack chooses a random number as the sequence number. Actually the connection requesting message is a message of one byte. Thus the random number provides the number to the calling sequence. This sequence number is called ISS (Initial Send Sequence number) for the receiver. If the random number is 200, this value is assigned to calling sequence. If and when the receiver receives this pack correctly, it responds by sending a pack to the sender with both the SYN and ACK flags set (i.e. SYN=1 and ACK=1). The receiver choose a sequence number by its random generator, and response message contains this sequence number. If the chosen number is 300 (IRS Initial Receive Sequence number for the sender), the reply message contains 300 in the sequence number field. The acknowledge number field contains a value 201 (as the received sequence number is 200). If the sender receives correctly the pack from the receiver, the sender sends another pack with ACK flag set (i.e. ACK=1), sequence number field set to 201, acknowledge number field set to 301, and the connection is then established. After the connection is established, the communication parties can perform transmission in full-duplex mode. The connection can be terminated by any station, either sender or receiver by sending a pack with FIN flag set (i.e. FIN=1). For example, after transmitting all data, the sender can send a pack with FIN flag set. When this pack is acknowledged by the receiver, the connection is terminated. SYN = 1 SYN = 1 and ACK = 1 Sender

ACK = 1

Receiver

Connection

(a)TCP handshake for establishing connection FIN = 1, say with sequence number = 100 ACK = 1 with say sequence number = 200 and acknowledgment number 101 Sender A

FIN = 1 with acknowledgment number = 101 and sequence number = 201

Receiver B

ACK = 1 with say sequence number = 202 and sequence number 101

The sender, A terminates by first two communications, the receiver terminates by last two communications; As if four way termination (b) TCP connection termination Fig. 32: TCP connection establishment / termination

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Initial Sequence Number(ISN) generation During connection request phase, the initial sequence number should be so chosen that the previous connections (sockets) are not confused with new connection requests (sockets). This typically happens when a host application crashes and before the other side times out, reestablishes the crashed connection quickly by recognizing that the connection request is for old socket. The ISN selection is made by a 32 bit random generator created during connection request phase. The number is generated by a 32 bit clock. The clock has a ISN cycle of 4.55 hours ( clock is incremented approximately every 4 ms), thereby providing unique ISN number within a period of 4.55 hours. BOX 10

QUESTIONS 1. As the sequence numbers, ISS and IRS are used for synchronization, where is then the need of handshaking for connection phase? The sequence numbers are local and not global. Thus the handshaking for establishing the connection is required by the process of the setting of the flags 2. How is it ensured that the sequence number is not duplicated? The host has to wait a maximum segment lifetime, known as MSL before re transmitting segments after the connection. 3. Sometimes the sequence number is addressed as the number allotted to each bytes of TCP message. Sometimes it refers to the first byte of TCP message. Again sometimes it refers to TCP message as a whole. What is the correct position? In practice each TCP pack is sent with a sequence number. But theoretically each byte is assigned a sequence number. When the sequence number is assigned to the first byte in the segment, that sequence number becomes the segment sequence number(SSN). That segment is transmitted with SSN. 4. How is the TCP reliable transport? The TCP is reliable transport as because: (a) the sequence number and the acknowledgement number together take care of the loss identification and recovery, and (b) advertisement of window by which the receiver announces the number of bytes may be accepted by it. To further illustrate the reliable transport of TCP, We shall look into the flow control mechanism used in TCP in the next section. The 16-bits window flag indicates the number of bytes that the sender can transmit without receiving acknowledgements from the receiver. This field is basically used by the receiver to inform the sender about the availability of the buffer in the receiver. Based on the information the sender control the flow. The flow control is actually implemented by the window field. The mechanism of control is discussed in details under the window flow control. The urgent pointer field is used to alert the receiver about coming of some urgent data and indicates to the end of the urgent data within the sequence of the transmission of packs. The value in the urgent pointer is valid provided the URG (URGENT) flag is set (i.e. URG=1). The urgent pointer value defines the end of urgent data and the start of normal data in bytes. For example if urgent pointer is 0000000010000000, the receiver will understand that next 128 bytes data will be urgent.

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The Reserved bits (6 bits) are for the future use. All the flags, namely URG, ACK, SYN, FIN, RST and PSH together are known as control field of the TCP. We have already discussed the important applications of URG, SYN, ACK and FIN flags. The PSH flag is set when the receiver is told to pass the data immediately to the application. Otherwise the receiver will buffer the segments until sufficient storage of data is made. RST flag is used to abort the TCP connection. When this flag is set, the receiver is advised to terminate the connection for some abnormal condition. We earlier discussed about the purpose of the options field in IP data gram. Nearly for that kind of purpose the options header is used in TCP pack. The options header can be maximum of 40 bytes, and always be a multiple of 4 bytes (32 bits words). To make it, a multiple of 4 bytes, padding may be used, if required. A few important options of TCP pack are: (a) MSS (Maximum Segment Size) option: this is used to indicate the maximum size of the segment that the sender can accept. This option is used during connection establishment phase, whereby the sender specifies the MSS. This is 16 bits option field. So the largest block that the MSS can specifies is 216 – 1 = 65,535 bytes. So, what is the largest data size that an IP/TCP pack can carry? A TCP pack or segment has minimum 20 bytes header and so is the case for IP header. Thus the largest block size of data that a TCP segment or IP datagram can carry = (65,535 – 40) bytes = 65,495 bytes. The declaration of MSS during connection set up time is made to overrule the TCP default MSS. The TCP default MSS is 536 bytes. (b) WSO (Window Scale Option): TCP pack has a header of 16 bits window. Hence the maximum permissible advertised window size = 216 – 1 = 65,535 bytes. With WSO, the larger advertised window size may be used for which upward scaling upto 214 is permissible. Thus, with WSO, the scaled advertised window size may go upto 214 × 216 – 1 = 1.073,725,440 bytes. (c) Times stamp option: This is used for round trip delay calculation and high speed connection. The round trip calculation is used for calculating time out period so useful for flow control. After discussion on sliding window protocol in the next section, we shall discuss about finding the time out period. 5.7.2.1 Sliding Window Protocol The window protocol has been discussed in connection with ARQ protocols of error control. The matter has further been discussed in chapter 3. 5.7.2.2 Time out Period The concept is used to determine the maximum size of packet.

SOLVED PROBLEMS 1. Compare the coding/bandwidth efficiency of IP datagram with the TCP default MSS with that of the largest MSS possible. Comment on the results. Solution. The TCP default MSS = 536 bytes. The minimum IP + TCP headers = 40 bytes. Thus the coding efficiency for the default MSS = {536/(536 + 40)} = 0.930. The largest block of data in MSS size = 65,495 bytes. Thus the coding efficiency in this case = {65495/(65495 + 40)} = 0.999

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Comment : We find that the coding efficiency is much higher in largest possible MSS. This may be one of the reasons of having the option included. 2. Is local IP authorized to fragment the data block under defined MSS? Solution. No. Actually once a MSS is defined by a sending process, the MTU is defined ( MTU = Maximum Transfer Unit in bytes = MSS + 40 bytes). With MTU size the data is transferred without fragmentation by the local IP. There are two variations of layer 3. These are UDP (user datagram protocol) and ICMP (Internet control message protocol). UDP is used when no sequence number is used when no sequence number is required, and a simple example of this situation would be when a message can put into one data gram. ICMP is simpler than UDP. In ICMP, the message is put into one datagram only. Besides, the message is for the network, and hence no addressing is required. ICMP is intended for TCP/IP software only. TCP divides the data into manageable packets that are easier to deal with and thereby provides a guaranteed error free data communication. Figs. (30 & 31) illustrate this. TCP breaks a NT service pack into several packets. If an error occurs, a small packet needs retransmission rather than whole pack. TCP uses a header (Fig. 31) to establish connection, to ensure successful transmission of a packet and completion of transmission of whole message. The source and the destination port fields keep the trace of which packet belongs to which application (table 16 shows application-level Internet Protocols and their assigned port number) so that running of e-mail, on-line chart and multiple browses can be allowed at the same time. Sequence number is used to reassemble the packets at the receiver. Receiver on getting a packets performs check-sum operation to detect error. If packet is received error free, a positive acknowledgement is sent. If packet is received erroneously a negative acknowledgement is sent. Transmitter either getting a negative acknowledgement within a time known as time out period, retransmits the earlier packet. This is ARQ (Automatic Repeat Request Protocol) [812] for error control. Flags, window and urgent fields are used to manage TCP connection. Flags are used to connect and terminate connection. Window field tells transmitter how fast to send packets. When receiver buffer is full, the window is set to zero. This is the means of regulating throughput by the transmitter by the receiver. Urgent field is used to send urgent information. When a receiver gets a packet with urgent field, it processes the packet with priority and acts accordingly like aborting a transmission. RFC 1240 refers to the UDP datagram. UDP datagram are used to carry the time sensitive data like voice and video. UDP works in connection less mode. UDP has very less overhead bits and overhead fields are very simple. The format of the UDP pack is shown in Fig. (33). Source Port (16-bits)

Destination Port (16-bits)

Length (16-bits) Checksum

(16-bits) Data

(Variable) Fig. 33: The format of UDP pack

5.7.3 UDP headers The 16-bits source port field gives the address of the process at the sender, whereas the process means the individual process that is in communication with the same process at the receiver.

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The 16-bits destination port refers to the address of the port at the receiver that is in communication with the same process at the sender. The 16-bits length field refers to the length of the whole of UDP pack in bytes. How does it then separates out the data field from the pack? This is simple as because the total of the header fields is fixed, and it is 8-bytes only. The 16-bits checksum field is used to check error in the frame. The checksum measure is taken over the whole of the frame. Problem. A UDP frame is given in Fig. (34). Calculate the check sum that will be filled in at the transmitter. Now if the same frame is received by the receiver correctly how the receiver will know about it? However if the frame is received as in Fig. (35), how the receiver will detect error? 0000111100001111

0101010101010101

0000000000001100

Checksum (16-bits)

0011001100110011

0011001100110011 Fig. 34

0000111100001111

0101010101010101

0000000000001100

Checksum (16-bits)

0011001100110011

1111001100110011 Fig. 35

The checksum will be calculated at the transmitter by adding 16-bits words in the UDP frames using 1’s complement addition rule as below: 0000111100001111 0101010101010101 -----------------------0110010001100100 0110010001100100 0000000000001100 -------------------------0110010001110000 0110010001110000 0011001100110011 ------------------------1001011110100011 1001011110100011 0011001100110011 ------------------------→ Final sum 1100101011010010→

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The checksum will be 1’s complement of the final sum so obtained. Thus in this case checksum field will be: 0011010100101101. The receiver calculates the sum as is done at the transmitter end, but it does so over the received bits. If there is no error in the received bits, the receiver will get the final sum as exactly it was obtained at the transmitter end. And in the present example, the receiver will get the final sum as 1100101011010010. With this final sum, the receiver will add the received checksum field as: below in the present case: 1100101011010010 0011010100101101 ------------------------1111111111111111 Now the receiver finds that there is all 1’s in the sum, and then it will conclude that the frame is received correctly. When the receiver gets the UDP frame as in Fig. (35), the calculated check sum will be different one from that of the transmitter. Hence when receiver’s calculated checksum will be added with the received transmitter checksum, the result will not be all 1’s. Hence the receiver will detect the error. The Commonly used port for UDP applications are as in table (18). Table 18: Commonly Used UDP Ports Port Number

Service

49 53 67 68 69 137 138 123 161 1645 1646

TACACS authentication server DNS (Domain Name Server) BOOTP server BOOTP client TFTP NetBIOS name server NetBIOS datagram service NTP (Network Time Protocol) SNMP (Simple Network Management Protocol) RADIUS authentication server RADIUS accounting server

2049

NFS (Network File System)

Layer 4 is the application and service layer; and this corresponds to the remaining higher layer of the OSI/ISO protocol. The services of the layer include FTP (file transfer protocol), remote login, computer mail and program to program communications using socket programming interface and RPC (remote procedure calls). FTP allows a user, a computer or a terminal to get (or to send) files from/to another computer. FTP is a tool of the Internet. Remote login is based on TELNET (network terminal protocol) which is another tool of the Internet. It allows any user to log in any computer in the network. Computer mail allows users to send mail to any other computer in the network, and one classic example of this is the E-mail. Each of the layers, while performing its function, adds a header to the message/datagram at the transmitting side ; and these headers are removed by the corresponding layer at the receiving side. Layer to layer function is a peer process.

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TCP/IP is the internetworking protocol. The Internet is an interconnected network of wide varieties of networks. Therefore the question may arise: How does the TCP/IP actually interconnect incompatible networks and provide compatible services? Interconnection among different networks is done by several methods: 1. By repeaters which cover only the physical layer, an example of which would be the connection of Ethernet to Ethernet. 2. By bridges which cover physical and data link layers, corresponding to the OSI/ISO protocol, an example of which would be Ethernet to token bus connection. 3. By routers which cover physical, data link and network layers (complete node coverage) corresponding to the OSI/ISO protocol, an example of which could be any IEEE802 series LAN to any X.25 WAN connection as shown in Fig. 36 Corresponding to TCP/IP, routers can cover up to the IP layer. 4. By gateways which are used to connection networks of different protocols, an example of which would be a connection between the OSI/ISO network and a DNA network. Repeaters, bridges and routers are used when interconnecting network are of the same protocol. Gateways, repeaters, bridge and routes are all made of computing resources. In fact, they are all computers. The complexity of the systems increases as one moves from repeaters to gateways. Bridges and routers can be defined as simple gateways. With the help of the Fig.36 we may illustrate the operation of TCP/IP in internetworking. Consider a situation that A wants to transfer some data to station B. TCP of station A sends datagrams with the necessary information of source and destination address to IP of station A. The IP layer attaches a global header to the datagrams. The header includes the internet global address. The global address has two parts-network identifier and station identifier. Till date, IP version 4 is being used wherever 32-bit addressing is used. IP version 6 is under consideration. In IP version 6, the addressing size is proposed to be higher. IP of station A now finds that the datagrams have a destination of another subnetwork, and therefore searches the routing table to route the datagrams to the corresponding router. In Fig. (37), we have shown only one router but in practice there may be more than one router to connect different subnetworks. In our example, IP of station A sends the datagrams to router (1) through lower layers. On receiving them, the IP of router (1) sends the datagrams to router (2). But to send to router (2), it has to utilize the X.25 network. Therefore ties among them would not cause any problem in communication. So we see that this ‘open system’ is really open. LAN-1

B

LAN-1

B

B G

Network LAN-2

B

LAN-4

B = bridge

G = gateway

(a)

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Higher layer

Higher layer

Network

Network

LLC

LLC

Data link

MAC

MAC

MAC

Physical

Physical

Physical

Data link

Bridge

LLC = Logical Link Control MAC = Medium Access Control Higher layer

Higher layer

Network

Network

LLC

LLC

Data link

MAC

MAC

MAC

Physical

Physical

Physical

Data link

Gateway

(b) Source

Destination Bridge

Packet from higher layer

MAC

802.3 Packet

Physical

802.3 Packet

Packet to higher layer

802 Packet

802.4 Packet

802.4 Packet

802.3 Packet 802.4 Packet

802.4 Packet

802.4 Packet 802.3 Packet

CSMA/CD LAN

TOKEN BUS LAN

Network A to internet

Network A

Network B to internet Buffer

Internet to network A

Internet to network B A full gateway

A full gateway

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Network A

Network A to internet

Network B to internet

Internet to network A

Internet to network B

Network B

Two halfways T

T T

802.5 LAN T T

G

G X.25 LAN

T

802.4 LAN

G T

T

T

T

G

T T

HG T

X.75 HG

B IMP T

IMP

T 802.3 LAN

X.25 T

HOST

G = Gateway HG = Half gateway T = Terminal / Node / Host.

B = Bridge IMP = Interface message Processor. (d) Fig. 36

BOX 11

A Write Up on Internetworking Internetworking Communication between any two entities may be direct or indirect. The point-to-point link or the multipoint link provides direct communication link; a switched network provides indirect

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communication facility. The case of more indirect communication arises when two entities which do not share the same network but may be connected through two or more networks desire to exchange information; and for this sort of indirect communication internetworking is required. The internetworking effectively creates a single very large loosely coupled network, which is often known as Internet. The internet in terms of internetting LAN (Local Area Network) and WAN (Wide Area Network) may be basically of three types (Fig. 1). Two or more LANs are internetted. Such Network is termed as extended LAN. The LANs in any extended LAN may be homogeneous or heterogeneous. LANs and WANs are internetted. Two or more only WANs are internetted. Such Internetted network is known as catenet.

Objectives The objectives of internetworking are many. The maximum number of stations attached to a single isolated network may be limited due to technical, legal or performance grounds. Internetworking effectively overcomes this limitation of individual network. Internetworking provides the ability to a user to share the resource of other networks. The geographic coverage of any isolated network is limited due to technical, economic or legal reasons. Internetworking overcomes such limitations of isolated networks. Internetworking may provide global coverage. Now naturally one can ask: why don’t we then design a single global network? The first answer to such a question will be that a global network is not viable due to administrative and political factors, which differ from country to country. The second answer is a technical one. A single network is not technically sound so far as operation, maintenance, reliability and flexibility are concerned. Moreover, until we have a standardized communication and computer systems and protocols (which is widely optimistic due to existing abundant networks of different types and the worldwide very high competitiveness of vendors in order to meet the interest of customers), any step to implement any global network will be an exercise in futility. The concept of global network for all purposes (Integrated Services Digital Network) is based on typical internetworking.

Gateways and their Classification As internetworking is to connect together usually the many different networks which are having different technologies, protocol and standards, it is necessary to use a system (a set of hardware and software) which will be able to remove the incompatibility of multiple networks in an internet. Such as a system is known as gateway in general. There are basically two factors that determine the key issues in designing a gateway, thereby providing as many as four types of gateways namely protocol translator. Internet protocol X.75 standard and Bridge. The two deciding factors are interfacing level and the nature of transmission services. The interfacing level may be of two types-station (DTE–Data Terminating Equipment) level and node (DCE-Data Circuit Terminating Equipment) Level (Fig. 1). As summing that the networks are packet networks, the nature of service may be again of two types-datagram service (connectionless service) which provides end-to-end service and virtual circuit service

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(connection oriented) which provides network by network connection. Hence 02 × 02 = 4 types of gateway systems exit (Fig. 2). The gateway are also known as data relay. Relays may be bilateral (connecting only two networks) or multi-lateral (connecting more than two networks). The simple gateway is a bridge used in internetworking LANs at the data-link layer of OSI (Open System Interconnection) model. The repeater may also be used in implementing extended LAN. But the repeaters at the physical link layer of OSI model copy the individual bits of a LAN and amplify and then transfer them to other LANs. Bridge only accept the frames from one LAN if these are to be transferred to other LANs; and operate in a store and forward mode. The bridge interconnects the similar networks, basically LANs. The Internet Protocol (IP) operates the above network layer. This was originally developed by the defense department of the US. Protocol converters operate at higher levels. Gateway are used in internetworking dissimilar networks; and operate at higher levels of OSI model in store and forward mode. While internetworking different networks, the networks may be widely separated. Then the question may arise: who will have the right to own and operate and maintain the gateways? For this each network may have a half-gateway. Similarly half-bridge is used in extended LAN.

LAN 802.3

G1

LAN 802.3

G1

G2

X.25 WAN

G2 Extended LAN

LAN 802.3

One type of catenet

LAN 802.5

G2

G3 Station level interfacing

Snanet wan

Dec net wan

• = Network Node. O = Network Station (DTE) G1 = Bridge G2 = Gateways (X.75) Standard Typically G3 = Protocol Translator or Internet Protocol Except A, All Other Are Of Node Level Internetwork Interfacing. Fig. 1: Internet in terms of Internetting LAN and WAN

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Service

241

End-to-end (usually used in LANs0)

Network-by-network (Usually used in WANs)

Station Level

Internet Protocol (IP)

Protocol Translator

Node Level

Bridge (mostly used in extended LAN)

Gateway (Example : X.75 Standard gateway functions for X.25 network)

Interfacing

← Connectionless →

← Connection

Gateway

oriented gateway

Fig. 2: Gateway System

5.8 ADDRESS RESOLUTION PROTOCOL (ARP) IP addresses often known as protocol addresses are unique logical addresses each assigned to a physical machine or system. IP layer releases the IP packet with IP logical addresses to the physical layer for the purpose of transport. The physical layer treats the whole of IP packet as a data pack and encapsulates this pack into the transport packet commensurate to the physical network. For example, if the transport network is an Ethernet LAN, the IP packet as a data pack is encapsulated in the Ethernet packet. Hence making the Ethernet packet, the destination address required in the Ethernet packet must be the physical address of the destination node. The physical address is the address provided by the NIC of the node. And it is only the physical address that transport network can understand for transporting the packet to the desired destination. So before making the Ethernet packet, the physical destination address is required to be known. Actually the software in a node determines the next node or hop based on the IP addresses in the IP packet. To transfer the IP packet to next node or hop, the corresponding physical address is required to be known. The address resolution refers to finding the physical address for a given IP address. Problem. The Internet task forces assign IP addresses and the IEEE committees develop link MAC (Medium Access Control) addresses of LAN. Why so many addresses? We could have just one address for networking. Solution. Note that different addresses are to serve different purposes. IP addresses for TCP/IP internetworking and MAC addresses are for linking devices in a network. Of course technically it is right that one address could have served the purpose. But in this complex world, that just did not happen. The address resolution is always local to the network. A computer can resolve the physical address of another computer from its IP address provided that computer is attached to the same network. A computer can not resolve the address of a computer attached to another network. For example consider the network in Fig. (37). In the Fig. (37), the physical networks, namely network 1, network 2 and network 3 are connected by routers R1, R2 and R3 . The Computer A can resolve the physical address of the computer B once its IP or protocol address is known as they are on the same physical network. But the computer A will not be able to resolve the physical addresses of the computers C, D , E and F as they are on the different physical networks. Now if the computer would like to send data to say F, the software of

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computer A will find that the computer F is not on its network, and hence will resolve the physical address of default router. See there are two routers, R1 and R3 on the network1. But assume that R1 is the default router. So the computer A will send IP packet meant for the computer F to the R1. The R1 will determine that the packets must go through R2, and accordingly resolve the physical address of R2, and send the packets to R2. The router R2 then will resolve the physical address of the computer F, and send the packets to the computer F. A

C

E

R2

R1

Network 1 B

Network 2 Network 3

D

D

R3

Fig. 37: Address resolution is local

Three techniques (Fig. 38) are employed for address resolution. The techniques are: Table look up, Closed form computation and Message passing/exchange. Address resolution techniques

Table look up (Used in WAN)

Closed form computation (Used in configurable networks)

Message passing/exchange (Used in LAN)

Fig. 38: address resolution techniques

In the table look up technique, a table known as ARP table or binding table is used for the address resolution. The table contains an array of data, each array is having protocol address and the corresponding hardware address. The table (18) is a typical ARP table. For each separate physical network, separate ARP table or binding table is constructed. Hence, the IP address entries in the table will bear the same network ID for all nodes. The table (19) is an ARP table for a class C network with network ID 230.120.92. As separate ARP table is required for separate physical network, the network ID (prefix of IP address) being the same for all nodes may be omitted from the ARP table in order to save memory space. The main advantage of the table look up is the simplicity in operation. Once an IP address is known, a search of the table will resolve the physical hardware address of the corresponding node or host or computer. When a network has less than a dozen hosts or nodes, a sequential search will suffice. For networks having higher number of hosts or nodes or computers, hashing or direct index search would be better solutions. The table look up technique is basically used in the WAN.

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Table 19: ARP table for a class C network having network ID 230.120.92 IP or Protocol Address

Hardware Address (Ethernet Address in this example)

230.120.92.2

00-80-C4-45-E3-87

230.120.92.3

00-80-E6-45-F4-4E

230.120.92.4

00-80-B3-89-56-7E

230.120.92.5

00-78-E8-45-12-E4

The closed form computation technique of address resolution is meant for the networks that use the technologies of configurable addressing. In the configurable addressing, the physical addresses are not static. In the closed form computation, a computational or mathematical or relation function exists between the IP address and the corresponding hardware address. For example, suppose a class B network with address 155.45.x.x is a configurable network. As and when computers are added to this network, the physical addresses to the computers may be assigned such that the 1’s complement of host’s ID in IP address becomes the corresponding physical address. Thus the host with ID in IP address 155.45.0.1 will have the physical address FF.FE or 11111111.11111110 (the host’s ID in the IP address 155.45.0.1 is 0.1 which actually means 00000000.00000001. The 1’s complement of this ID is 11111111.1111110). Thus the mathematical rule for resolving address is to get 1’s complement of host’s ID in the IP address as the physical or hardware address of the host.

QUESTIONS 1.

2.

In a closed form computation technique, the conversion formula is physical address = IP address ^ 00EF Find the physical address for a host with IP as 203.67.56.39 In a closed form computation technique, it is desired that host ID will be the physical address of host. Find the conversion formula.

The address resolution protocol (ARP) is basically used in the message passing technique of address resolution. The specifications of the ARP are documented in the RFC 826. As pointed out earlier, the message passing technique of address resolving is mostly used in LAN. The nodes in the LAN may work with the look up tables also. In that case the look up tables are known as the ARP cache. To resolve any address, the nodes first search their respective ARP cache. When the ARP cache fails to supply the necessary resolution, the message passing is evoked. The node then sends a broadcasted ARP request to all other stations of its network to find the link address of the target or destination IP address. Any station that recognizes the target IP address sends a reply to the inquiring node. The reply contains the physical address of the target IP address. The technique is illustrated in the Fig. (39). In the Fig. (39) , the node “A” sends an ARP request and the node “D” sends the ARP reply.

A

B

Inquiring Node 130.24.45.90 00-89-FE-10-79-06

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D

Destination or Target Node 130.24.78.45 00-45-D4-A0-89-78

E

F

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Broadcasted ARP request from inquiring node, A ARP request

Source Hardware Address (00-89-FE-10-79-06)

Source Protocol Address (130.24.45.90)

Target Hardware Address (FF-FF-FFFF-FF-FF)

Target/Destination Protocol Address (130.24.78.45)

A → All (broadcasted/ FF-FF-FF-FF-FF-FF) Here is the request. What is the physical address of this target protocol address ? ARP reply from the node, D that has recognized the target or destination protocol or IP address ARP request

Sending Hardware Address (00-45-D4-A0-89-78)

Sending Protocol Address (130.24.78.45)

Target/ Destination Hardware Address (00-FE-10-

Target/Destination Protocol Address (130.24.45.90)

79-06)

89-78.

D → A Here is the reply. Well the physical address of your target /destination is 00-45-D4-A0Fig. 39: Illustration of ARP request/reply

The ARP message format is shown in Fig. (40). The destination address and the source address refer to respectively the Ethernet (if the network is Ethernet) destination and source MAC address. Ethertype field is very important. This field actually identifies what the type of the data in the frame. The Ether type assignments for ARP and RARP (Reverse ARP discussed latter) are as below: Assignment

Decimal number

Hex number

ARP

2054

0806

RARP

32821

8035

The hardware address space is 2 bytes. This is set to a value of 1 for Ethernet. The protocol address field is 2-bytes. For IP this field is set to a value of 2048. The hardware length field defines the length of the MAC address in bytes, which is typically 6 bytes. The protocol length field defines the length of the protocol address in bytes, which for IP is 4 bytes. The opcode field contains a value of 1 for ARP request and a value of 2 for ARP reply. The next four fields will be what we have described in the previous Fig. (39) in illustrating the ARP request/ reply.

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Hardware Type Hardware Length

245

Protocol Type Protocol Length

Operation Request 1, Reply 2

Sender hardware address Sender Protocol address Target hardware address (It is not filled in a request) Target protocol address

(a) ARP Packet Destination address (6 bytes) Source address (6 bytes) Ether type (2 bytes) This ethernet data packet

Hardware address space (2 bytes) Protocol address space (2 bytes) Hardware address length (1 byte) Protocol address length (1 byte) Op code ( 2 bytes) Sending hardware address (6 bytes)

This actually ARP DATA field as shown in Fig. (39)

Sending protocol address ( 4 bytes) Target/destination hardware address (6 bytes) Target/destination protocol address (4 bytes)

(b) ARP Message Format C B

F LAN 1 Router A

E LAN 2 D

(c) Proxy ARP Fig. 40: ARP Illustration

5.8.1 Proxy ARP The ARP techniques illustrated in section (5.8) are based on address mapping procedure. Proxy ARP is a flexible technique in resolving addresses. The proxy ARP works as in Fig. (40 C). In the figure, it is seen that two local networks are connected by a router. The router hides the

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networks connected in it, from each other. If the host A likes to send data to the host F, the host A will issue an ARP request to get the physical address of the host F. The technique does not allow the ARP request to reach the host F. The router intercepts the ARP request, and sends a reply of ARP request with the router’s own physical address. The host A on receiving the reply, processes it and sends the data. Thus the router in the example acts as a proxy ARP server. This technique will work only the network has the installation of the proxy ARP.

5.9 REVERSE ADDRESS RESOLUTION PROTOCOL (RARP) RARP is just reverse of ARP. The RARP protocol is used to find the protocol or IP address of a station whose link or physical address is known. The RARP is not so commonly used as ARP as because most stations or computers known their IP address as they have hard disk to store the same. The RARP is required for the stations having no hard disk like terminals or diskless workstations. . The RARP works like ARP, but to run a RARP service a RARP server is a must in the network. The RARP server maintains a mapping table like a look up table that was discussed earlier. In the ARP all nodes are at par. But in the RAR, the service is client–server oriented. The specifications for RARP are defined in RFC 903. The inquiring node sends a broadcast RARP request. The RARP server recognizes the request and searches its data base to find the protocol address for the given physical address of the inquiring node. The server then sends a RARP reply to the inquiring node. The operation is illustrated in the Fig. (41). When the node A is sending RARP request, the destination hardware address will be the hardware address of the RARP server, but the destination protocol address will be broadcast which is net ID plus all 1s in the field of host ID. In the example the network being a class C IP network, the last byte is thus made 255. The message format as in ARP is used in the RARP except that the opcode now changes. The op code is 3 for RARP request and 4 for RARP reply.

A

B

C

00-Ad-EE-67-45-89 00-67-89-09-56-AC/230.89.67.59 RARP SERVER

The data base of server Physical address

Protocol Address

00-45-EF-B4-D6-89 00-56-78-90-A4-BD 00-AD-EE-67-45-89

230.89.67.56 230.89.67.57 230.89.67.58

00-D8-C8-78-99-65

230.89.67.60

A is sending RARP request RARP request

Source Hardware Address (00-AD-EE 67-45-89)

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Source Protocol Address (???????)

Destination Hardware Address (00-67-89-09-56-AC)

Destination Protocol Address (230.89.67.255)

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RARP server is sending reply to A RARP reply

Source Hardware Address (00-67-89

Source Protocol Address

Destination Hardware Address

Destination Protocol Address

90-56-AC)

(230.89.67.59)

(00-AD-EE-67-45-89)

(230.89.67.58)

Fig. 41: Illustration of RARP

BOX 12

QUESTIONS 1.

2. 3. 4. 5. 6.

One of the reasons for using leading bits to classify IP addresses into different classes instead of using a range of value is that the use of bit can decrease the computational time. Hence a table of first four bits may be used to compute the classes of address. Design table. Is it advisable to redesign IP address with higher bit size (> 32 bits) to eliminate the address classes? Give a comparative table of different ARP technique in term of your defined parameters. It is often argued that “message passing ARP is hopelessly inefficient” ----------------- why? Suggest a solution to overcome the problem. ARP is called local. Can it be remote? If more than one different ARP replies with different hardware addresses are received for a single ARP request; how will it be processed?

Hints for Question (1) First Four Bits of Address

Table Index (in decimal)

Class of Address

0000 0001 0010 0011 0100 0101 0110 0111 1000 1001 1010 1011 1100 1101 1110 1111

0 1 02 3 4 5 6 7 8 9 10 11 12 13 14 15

A A A A A A A A A A A A A A A A

Table that can be used to compute the class of an address. The first four bits of an address are extracted and used as an index into the table.

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5.10 IPV4 TO IPV6 5.10.1 Ipv4 addressing review IP provides the basic service of getting the datagram to their destinations. It provides this service in best effort protocol. IP on receiving the TCP packets, forms it own packet known as IP packets. Each IP packet is associated with IP header. IP header has source IP address and destination IP address. These addresses are used to route the datagrams. IP addresses are universal machine identifiers, which are shared by all the machines in the Internet. IP uses 3 modes of addressing[2-6] for the purpose of routing packets. These are known as class A, class B, and class C. An IP address is a 4 bytes or 32 bits binary number. IP address has a Network Address Field (often known as Network ID) and Local Address Field (often known as Node or Host ID). Network address ID identifies the network to which a node or host is attached, and host ID identifies the individual node or host attached to the network already identified by the network ID. The difference in classes is due to how many bits are used for network identification and how many bits are used for host identification. The different address formats under different classes and the address space under different classes were illustrated in earlier sections. The different classes are identified by the left most bits of the address: for class A left most bit is 0, for class B left most two consecutive bits are 1 & 0 and for class C left most three consecutive bits are 1, 1 & 0. The 4-bytes of an IP address are usually mentioned as four sets of dot-separated octets in decimal notation. For example 10.0.16.9 is a valid IP address, and in binary notation this is: 00001010.00000000.00001000.00001001. The address in the example refers to class A address as the left most bit is 0. This address points to a network number 10 (the decimal value of first octet), and to the host or node number 4105 (the decimal value of second, third and fourth octets taken together as a binary number) attached to network number 10. For class A, most significant bit of first octet is always 0. Therefore next seven bits are used for network identification, and this means that only 27 = 128 networks can be of class A. As address 0.0.0.0 is used for system initialization, the class A has network address field from 00000001 to 01111111, that means from 1 to 127. But the packets with first octet as 127 is used for network testing. As an example: the address 127.0.01 is used for loop back address. Thus address for the network 127 is not open to use. Therefore for addressing purposes, class A uses network addresses from 1 to 126. Each of the network under class A address will have node or host address field of 3 bytes or 24 bits. Each network may have maximum number of nodes or hosts equal to 224 . The identification of a network under any class is made by filling the local part or node/ host field of address with zeros. Therefore the following addresses are reserved (x is a binary bit and may be either 0 or 1): Class A: 0xxxxxxx.00000000.00000000.00000000 = Network ID.0.0.0 Class B: 10xxxxxx.xxxxxxxx.00000000.00000000 = Network ID.0.0 Class C: 110xxxxx.xxxxxxxx.xxxxxxxx.00000000 = Network ID.0 Again the broadcast, under a specific network, is done by filling local part or host/node field of address with ones. Therefore the following addresses are reserved: Class A: 0xxxxxxx.11111111.11111111.11111111 = Network ID.255.255.255 Class B: 10xxxxxx.xxxxxxxx.11111111.11111111 = Network ID.255.255 Class C: 110xxxxx.xxxxxxxx.xxxxxxxx.11111111 = Network ID.255 These mean that for identification of networks and for internal broadcasting in networks, two numbers of host or node addresses under each class are kept reserved. Thus the host or node addresses available under class A = 224 – 2 = 16777214 – 6 = 16777214

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For class-B address, first two bits of first octet must be 1 & 0, and these make first octet to be from 10000000 to 10111111 i.e. from 128 to 191. The second octet will be from 00000000 to 11111111 i.e. from 0 to 255. The first and second octet in combination provide Class B network addresses to range from 1000000000.00000000 (or 128.0) to 10111111.11111111 (or 191.255). Total networks and hosts/nodes under a network of address class B are respectively 214 and 65534 (=216 – 2). For class-C address, first three bits of first octet is 110. This means the first octet will be form 11000000 to 11011111 i.e. from 192 to 255. Class C network address will range from 11000000.00000000.00000000 to 110111111.11111111 i.e. from 192.0.0 to 223.255.255. Class-C address has 221 networks and 254 (=28 – 2) hosts or nodes per network. The total maximum hosts under different addressing modes are: Class A = (27 – 2 ) × (224 – 2) = 126 × 16777214 = 2113928964 Class B = 214 × (216 – 2) = 16384 × 65534 = 1073709056 Class C = 221 × (28 – 2) = 2097152 × 254 = 532677708 The address scheme under class-A to class-C is for unicast communication. The other addressing schemes are for multicast communication. Class-D belongs to the multicast scheme. Multicasting allows a stream of data to be sent simultaneously to a designated subset of network users. This is a very effective way of transmitting data to many receivers. This is contrast to uni-casting and broadcasting. In uni-casting, separate data packet is sent to each receiver. In broadcasting, all packets are sent to everyone. Class D addresses are identified by four left most consecutive bits as 1110. In class-D, all the four octets are used to identify the group of nodes designated to receive a multicast. Class-D address does not specify the network. Class-D addresses are in the range of 11100000.00000000.00000000.00000000 to 11101111.11111111.11111111.11111111 i.e. 224.0.0.0 to 239.255.255.255. Class-E addresses are identified by the consecutive four left most bits as 1111 and next bit as 0, a n d t h e r e f o re the addresses range from 11110000.000000.00000000.00000000 to 11110111.11111111.11111111.11111111 i.e. from 240.0.0.0 to 255.255.255.255. But 247.255.255.255 is used for broadcast for all networks. Class E address is still under experimentation. The special addressing schemes are as below: 1. 255.255.255.255 is used as a broadcast packet for all networks. 2. For broadcast in a specific network, local part or Node/Host field of the address is set to 255. 3. IP address 0.0.0.0 is used for system initialization. IP address 0.0.0.0 is called “This Host, This Network” address. 4. Packets with first octet as 127 are used for network testing. 5. IP address 127.0.0.1 is the Loop back Address. 6. An entire network is specified by providing only the network identification and with 0s in all other octets such as: 124.0.0.0 for class A, 129.155.0.0 for class B and 200.127.11.0 for class C. 7. Subnet mask is special addressing scheme. It is used for two purposes: to show the class of addressing in use, and to divide a network into different sub networks to control traffic. For first purpose, a subnet mask determines which part is for network ID and which part is for hosts ID. A subnet mask for class A network is: 11111111.00000000.00000000.00000000 (=255.0.0.0). All 1s indicate network ID and all 0s indicate host ID. For second purposes, subnet mask is used to divide the network within network administrator. Always a network number is given to an organization.

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If an organization has applied for a Class B address, the organization may be given a number 140.80.X.X. Now 140.80 is what that bothers to Internet regulator. What your organization does with host/node field of address (X.X) that is up to your organization. For example: when you are given a number 140.80.X.X; you may use the entire third octet of this address space for subnet. In that case your mask for subnet identification will be 11111111.11111111.11111111.00000000 (=255.255.255.0). Another example: of subnet ID mask under class B could be: 11111111.11111111.11110000.00000000 (= 255.255.240.0). In this example 4 left hand side bits of the third octet are used for subnet networks ID, whereas other 4 bits plus last octet is kept for host ID. The use of subnet ID is to expand your organizational network. Natural question may then arise: Why Subnet and Why not different network ID. Network space is limited, therefore it is not desirable to allow your organization, say 10 to 20 network address-space. Second allowing a number of network addresses to a particular organization creates complexity in routing table. But with subnet concept, as the subnet is under your network control, it will not hamper the main routing table of the Internet.

5.10.2 Ipv6 Addresses The address scheme defined so far is up to IP version 4 (Ipv4). Ipv4 is facing some serious problems. First, Ipv4 has 32 bits address scheme. But due to exponential growth of Internet users[11], literally this address space is running out of addresses. In IP Next generation (Ipng) or IP version 6 (Ipv6), 128 bits address space (in place of 32 bits of Ipv4) and 40-byte IP header (in place of 20 byte of Ipv4) have been proposed to cope up with the increased demands. IP addressing with 128 bits or 16 bytes is called IP version 6 because of the reason illustrated below: 2 (or 4 bytes) bytes address space refers to IP version 4, then 23 bytes address space may refer to IP version 5 if any one there, and therefore 24 bytes address space would refer to IP version 6. Second, due to huge growth of Internet users, organizations are being assigned C addresses. This is causing exploration in routing table of Internet. Third, Ipv4 is for best effort delivery of packet. It does not guarantee packet sequence integrity and consistent latency in delivery, and hence inherently unsuitable for real time services. Thus with Ipv4, voice, video and multimedia services will not be possible at the required level of quality. On the other hand, growth in voice IP traffic is tremendous. Ipv6 is being explored to avoid the problems being faced by Ipv4 to support real time services like voice, video and multimedia services. Ipv6 is proposed to have the following features: 1. 128 bits address size. This means that a total of 2128 = 256 × 2(10 × 12) = 256 × 10(3 × 12) = 256 × 1036 different addresses would be available in the address space. The inefficiency in the allocation and administration of the address space is measured by the H factor. The H factor is defined as the ratio between the log (number of addresses) and the number of bits in addresses. H factor is usually 0.22 to 0.26. Taking into account the H factor, Ipv6 is believed to support 1015 or quadrillion of networks and 1012 or trillion of networks.

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2. The notion for writing Ipv6 address is each double bytes of the 16 bytes field are separated by colon. The bytes are written in hexadecimal symbols. Thus one example of the address could be: ABCD:23D5:7893:C07E:3425:9BAC:6754:CED6. The proposed notation has the provision to skip leading zeros so that 0000 can be written as 0 or 0056 can be written as 56. It has also the provision of removing a 0 leaving the colons by the technique of double colon notation. For example, an address like ABCD:0:0:0:0:0:7896:DE45 can be written as ABCD::7896:DE45. The double colon notation can be used at the beginning and at the end of the address but only once. In the transition period Ipv4 address would be converted to Ipv6 address by pretending 12 bytes of 0s. Thus an Ipv4 address 102.23.78.10 would be written an Ipv6 address as 0:0:0: 0:0:0:0:0:0:0:0:0:102.23.78.10 or ::102.23.78.10. Ipv6 has a provision of a single address associated with multiple interfaces, 3. Ipv6 has auto configuration facility, 4. Ipv6 provide QoS (Quality of Service) service support to real time services like voice and video, support to mobility, 5. The flow level and priority in the header of Ipv6 facilitate the support of real time data, 6. Ipv6 has an efficient header format. In Fig. (42) we find overhead bits of Ipv6 are less than that of Ipv4. The overhead bits in Ipv4 is 12 bytes in the header format of 20 bytes, whereas the overhead bits in Ipv6 is 8 bytes in the header format of 40 bytes. The less header in Ipv6 helps in achieving higher data rates required for voice and multimedia services [13]. 7. The header of Ipv6 is different from that of the Ipv4 in many aspects: (a) Ipv6 has a fixed header size unlike Ipv4. Options and padding are the variable fields in Ipv4. These are removed in the Ipv6. This makes Ipv6 to act like ATM (Asynchronous Transfer Mode) cell comparison to Ipv4 which acts as a conventional packet. (b) There is no header checksum in Ipv6. This modification is made based on experience that there is no need of checksum at each intermediate node. (c) there is no hop by hop segmentation procedure in Ipv6. Therefore there is no fragment offset, flags and identification fields in the Ipv6. There is also no need of header length field as with next header field the chaining of IP headers is possible in Ipv6 (Fig. 43). (d) the TOS (Type Of Service) field of Ipv4 is removed in Ipv6. This removal is also based on experience. Hardly the TOS field is used in IPV4. 8. Ipv6 defines six extension headers (Fig. 43): Hop by hop options header, Routing header, Fragment header, Encrypted security payload header, Authentication header, and Destination options header. 9. Ipv6 has different options, security, and easy transition facility. We can compare Ipv6 with Ipv4 in terms of two important parameters, namely address space and IP header coding efficiency among others. Like gain bandwidth product of the amplifiers, the address space (AS) and header coding efficiency (CE) relative to address space only, could be used for comparing the versions. The comparison is shown in Table (20): Table 20: Comparison of Ipv6 with Ipv4 in terms of AS X CE Versions

AS (a rough estimate) 109

Ipv4

232 = 4 ×

Ipv6

2128 = 256 × 1036

The achieved benefit is in the ratio of 1:128 × 1027.

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AS × CE

8/20

1.6 × 109

32/40

204.8 × 1036

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5.10.3 How Did IPv6 Come Up Although a few people are projecting the death of the Internet due to increased traffic, yet Internet access is growing exponentially over time and is providing the better services. Forget about the death of the Internet, a next generation gigabit Internet was proposed in 1996 and is being experimented now. Internet is on no-stop move from kilo bits per second to giga bit per second/tera bits per second. The world may enter into the age of the Internet2 in future. Internet2 shall be next generation of the Internet, when the present Internet may have to be designated as Internet1. It is often said that Quality X Quantity = Constant , meaning that if quantity increases, the quality is bound to fall and vice-versa. But this theory has failed in case of the Internet. As per research firm International Data Corporation of USA “the number of people accessing the World Wide Web will hit almost 100 million by the year-end 1998 and 320 million by 2002.” John S Quarteman, President of Matrix Information and Directory Services said ” The quality of Internet is actually getting better – not worse ……… the number of servers supplying information increased rapidly and massively to cope with the demand for the information. In this way Internet is even better than traditional media” Craig Partridge, Chief Scientist, BBN Technologies, Cambridge said” Recent years have seen a tremendous focus on improving access to the Internet. We’ve seen a push to make the Net easier to use, and to improve data rates with 56-kb modems and with new access technologies like cable modems and asynchronous digital subscriber loop.” Commenting over Internet business in Jauary’99, IDC Senior Vice President John Gantz said “ Within five years, every dollar of Internet investment in the United States will be paying back $1.50.” Internet is going to have several changes and version like IPv4 to IPv6, Internet2, Wireless Internet and Cable Internet. But most immediate is from IPv4 to IPv6. Presently, Internet works on IPv4 (Internet Protocol Version 4) as defined in RFC791. By the middle of 1990s, by the time of which the IPv4 became about 15 years old, it was recognized that there are several limitations in the IPv4. Table (XXI) lists the major studies on the run up of IPv4. Two important limitations are the inadequate address space available with 32-bit address space of IPv4 and inability of the IPv4 to support real time services or time-sensitive services. The 32-bit address space is not sufficient to cope up with the growing Internet users. Since it is estimated that the Internet has been growing by a factor of two every year, the underlying principles and assumptions based on which IPv4 was designed are going to be invalid. What was duly sufficient for a few million users or a few thousands of networks will no longer can support a world with tens of billions of nodes and hundreds of millions of networks. Inability of IPv4 to support real time services was the stumbling block to realize Internet telephone. IPng (Internet Protocol Next Generation) initiative (RFC 1752) was then, started by the Internet Engineering Task Force (IETF). By 1996, the IETF proposes IPv6 (Internet Protocol Version 6) under IPng initiatives, which is supposed to solve the problems of IPv4 including the two major limitations mentioned above. IPv6 is therefore the future replacement of IPv4. From the experience over IPv4, it was felt that new version should take care of: More addresses, Reduced overhead, Better routing, Support for address renumbering, Improved header processing, Reasonable security and Support for mobility. Under the IPng initiatives the main techniques investigated were: • TUBA that refers to TCP (Transmission Control Protocol) and UDP (Users’ Datagram Protocol) with bigger addresses • CATNIP that means common architecture for the Internet. The main idea is to define a common packet format that will be compatible to IP, CLNP (Connectionless Network Protocol) and IPX (Internet work Packet Exchange). CLNP has been proposed by OSI (Open System Interconnection) as a new protocol to replace IP, but never been adopted because of its inefficiency

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• SIPP (Simple Internet Protocol Plus) that proposes to increase of the number of address bits from 32 to 64, and to get rid of unused fields of IPv4 header As none of the above three was seen to be suitable. As such, a mixture of all these three along with other modifications was suggested in RFC 1883. The RFC 1883 suggested the modifications as below: • Expanded Addressing in suggesting 128 bits for address that may allow more levels of address hierarchy, increased address space and simpler auto configurable addressing • Improved IP header format by dropping the least used options • Improved support for Extensions that will bring flexibility in operations • Flow Label that will make the real time services possible over Internet Based on the experience gained in operation of IPv4 over about 20 years, the design of IPv6 has considered four major simplifications: • assigning fixed format to each header. This ensures the removal of header length field that is essential in IPv4 • removing header checksum. The main advantage in removing header checksum is to diminish the cost and the time delay in header processing. This may cause the data to get misrouted. But experience has shown that the risk is minimal as most of data pack is encapsulated by the packet checksum at other layers like MAC (Media Access Control) procedure in IEEE 802.X and in adoption layer of ATM (Asynchronous Transfer Mode) etc. • removing the hop by hop segmentation procedure • removing TOS (Type Of Service) field that IPv4 provides, since experience has shown that this field has ever been set by applications. On the other hand, IPv6 has considered two new fields, flow label and priority. These are included to facilitate the handling of real time services like voice, video and high quality multimedia etc.

5.10.4 IPv6 in Details Thus the IPv6 was finally come up with packet format as in the Fig. (42). The final specifications of IPv6 was produced in RFC 1883. The new features of IPv6 are: • A fixed and streamlined 40-byte header: IPv6 is having fixed header bytes like that in ATM (Asynchronous Transfer Mode) cell. This makes the node processing delay to minimize, and thereby becomes more suitable for real time services like voice, video and multimedia. • Expanded addressing capabilities: A 128-bit address space in IPv6 instead of 32 bit as in IPv4, is believed to ensure that the world won’t run out of IP addresses. The 128 bit address size gives rise to a total of 256 × 10 36 different addresses. It is expected the Internet under IPv6 to support 10 15 (quadrillion) hosts and 10 12 (trillion) networks. The Internet under IPv4 can support maximum 2 32 hosts. Therefore the IPv 6 address space is about 64 × 10 9 times more that that of IPv4. This is why it is expected that future and exponential growing demand for Internet connection be met with IPv6. • New Address Class: Besides unicast and multicast, IPv6 has the provision of anycast addressing. Anycast address allows a packet addressed to an anycast address to be delivered to any one of a group of hosts.

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• A single address associated with multiple interfaces • Address auto configuration and CIDR (Classless Inter-domain Routing) addressing • Provision of extension header by which special needs like checksum, security options may be introduced. • Flow labeling and priority: Flow level and priority headers are used to comfortably support the real time services. By assigning higher priority to the real time packets, the necessity of time sensitiveness is restored. Data packets and for that purpose time insensitive packets are assigned low priority and serviced by the best effort approach. As per RFC 1752 and RFC 2460, this new feature allows “ labeling of packets belonging to particular flows for which the sender requests special handling, such as a non-default quality of service or real-time service.” Hence video and audio may be treated as flows whereas traditional data, file transfer and e-mail may not be treated as flows. • Support for real time services • Security support that could be eventually seen as the biggest advantage of IPv6. Today, billion dollars business is done over Internet. To keep the business secure, public crypto system has emerged out as one of the important tools. IPv6 with its ancillary security protocol has provided a better communication tool for transacting business over Internet • Enhanced routing capability including support for mobile hosts. IPv6 as such is not simple extension of IPv4, but a definite improvement over IPv4 in order to meet growing demand of Internet connectivity and the services of real time communication via Internet. Version (4 bits)

Priority (4 bits)

Payload Length (16 bits)

Flow label (24 bits) Next Header (8 bits)

Hop Limit (8 bits)

Source Address (128 bits) Destination Address (128 bits) Variable length TCP pack/ UDP pack (which is TCP header + Payload or UDP header + Payload)…….. Fig. 42: IPv6 Packet Format

The functions of IPv6 headers that is of base headers of fixed 40 bytes are: • Version field (4 bits): It contains the version number. Versions are 4 and 6. For version 6, this field is 6 (i.e. 0110). The various assigned values for IP version label are shown in table (XXI). But it must be remembered that just putting a number “6” or “4” does not make the corresponding IP packet. For the corresponding IP packet the proper format is required to be made. • Priority (4 bits): The bits in the field indicate the priority of the datagram. The priority levels are 16 from 0 to 15. The first 8 priority levels (from 0 to 7) are for the services that provide congestion control. If the congestion occurs, the traffic is backed off. These are suitable for non-real time services like data. The different priority levels under the first 8 levels are: 0 that defines no priority, 1 that defines background

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traffic like Netnews, 2 that defines unattended transfer like e-mail, 3 remains reserved, 4 that defines attended bulk transfer like FTP (File Transfer Protocol), NFS, 5 remains reserved, 6 that defines interactive traffic such as Telnet, X-windows, and 7 that defines control traffic such as SNMP (Simple Network Management Protocol) and routing protocols. The higher 8 priority levels (from 8 to 15) are used for services that will not back off in response to congestion. Real time traffics are examples of such services. The lowest priority level of this group 8 refers to traffic that most willing to be discarded on congestion and the highest priority level 15 is for traffic that is least willing to be discarded. • Flow level (24 bits): It is proposed to be used to identify different data flow characteristics, which will be assigned by the source and can be used to label packets. The packet labels may be required to provide special handling of packet by IPv6 routers, such as defined quality of service (QoS) or real time services. The combination of the sender IP address and the flow label creates a unique path identifier that can be used to route the datagrams more efficiently. The field is still being experimented. Flow is actually a sequence of packets coming from a particular source and destined for a particular destination. A flow may require a special handling by routers. Each flow is uniquely defined by the combination of the source address and a non-zero flow label. The flow label can be from (000001)H to (FFFFFF)H in hex. The packets having no flow label are given a zero label. All packets in the same flow must have same flow label, same source and destination addresses and same priority level. The initial flow label is obtained by the source by pseudo random generator, and the subsequent flow numbers are obtained sequentially. • Payload length (16 bits): The field indicates the total size of the payload of the IP data gram that excludes header fields. It can define up to 65,536 bytes of payload. • Next header (8 bits): The field indicates which header follows the IP header. The next header can be either one of the optional extension headers used by IP or the header for an upper layer protocols such as UDP or TCP. The field defines the type of extension header. For example 0 defines IP information, 1 defines ICMP (Internet Control Message Protocol) information, 6 define TCP information, 44 defines fragmentation header, 51 defines authentication header and 80 defines ISO (International Standard Organization) /IP information. Each extension header again contains an Extension Header Field and a Header Length Field (Fig. 43). When there is no other extension header, the next header will be TCP and hence the next header field will contain 6. The length of the base header is fixed 40 bytes. The extension header gives the functional flexibility to the IPv6 datagram. Maximum six extension headers can be used. The extension headers may be source routing, fragmentation, authentication and security etc. IPv6 currently defines six extension headers: (1) Hop by hop option header, (2) Routing header, (3) Fragment header, (4) Authentication header, (5) Encrypted security payload header and (6) Destination options header. If one or more extension headers are used, they must be in order in which they are presented above. For example, if Authentication header and routing extension header are to be used, the extension header fields must follow as: (1) main IPv6 header, (2) routing extension header (3) Authentication header and (4) TCP header with data. Each extension header must have one 8-bit next header field. For all extension headers except the fragment header (as in case of fragment header the flags and offset is 16 bits fixed), the next header field is immediately followed by a 8-bit extension header length that indicates

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the length of current extension header in multiple of 8 bytes. In the last extension header the next header field contains the value 59. The example that we considered earlier, the next header in main IPv6 packet will contain the routing extension header, the next header field in the routing header will show the authentication extension header, and the next header field of the authentication header will contain the value 59 (A comprehensive list is shown in Fig. (43). By specifying the next header field as TCP, the last of the extension header can be indicated. Why then the need of the number “59?” This is because IP does not only carry TCP segment, it carries UDP pack, ICMP pack etc. To standardized, the end of all the value “59” has been recommended. • Hop Limit (8 bits): This field indicates the maximum number of hops that the datagram is allowed to traverse in the network before it reaches its destination. If after traversing this maximum number of hops the datagram does not reach the destination, the datagram is discarded from the network. The field is used to avoid the congestion that may be caused by the datagram. Each router decreases the hop limit by 1 while releasing the datagram to the network. When the hop limit reaches 0, it is deleted. The hop limit of IPv6 is exactly what is called Time To Live in IPv4. The new name of Hop Limit has been given as the name suits better to its function. • Source Address and Destination Address (Each 128 bits): Both the addresses can be called IP address and are described in RFC 2373. IP address that defines the original source of datagram is called source address. The IP address that defines the final destination of the datagram is called the destination address. The three main groups of IP addresses are: unicast, multicast and anycast. Unicast address defines a particular host. A unicast packet is identified by its unique single address for a single interface NIC (Network Interface Card), and is transmitted point-to-point. A multicast address defines all the hosts of a particular group to receive the datagram. The anycast address will be addressed to a number of interfaces on a single multicast address. The anycast packet therefore goes to the closer interface and does not attempt to reach the other interfaces with the same address. A multicast packet, like anycast packet has a destination address that is associated with number of interfaces, but unlike the anycast packet, it is destined to each interfaces with that address. Unlike IPv4, IPv6 addresses do not have classes. But the address space of IPv6 is subdivided in various ways for the purpose of use. The sub division is done based on leading bits of addresses. The present division of IPv6 address space is as shown in table (22). The IPv6 address space is huge enough. So a portion of the IPv6 is reserved for computer system using Novell’s Internet Packet Exchange (IPX) network layer protocol, as well as the Connection Less Network Protocol (CLNP). Details on Use of Extension Headers Hop by hop extension header: The payload length header field of IPv6 defines the maximum payload size in bytes; and that is 65,535 bytes or 64 K bytes. Some applications like multimedia services or for use of super computers, may require the larger payloads than 65,535 bytes. By using hop by hop extension headers, the size of the payloads may be increased to 219 bytes (why not any size? The header length of extension header specifies the data in extension in bytes, but must be in multiple of 8 bytes. Therefore each extension header can accommodate about 28 bytes. Maximum 6 extension headers are allowed. So the total bytes that can be accommodated = 6 × 28 by the extension headers plus 216 for the payload field ≅ 219 bytes).

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However the large packet can be constructed with the jumbo packet extension header (see Fig. 43 d), where a value 194 specifies the jumbo packet. The option length field identifies the jumbo or large packet. This indicates the jumbo payload length field in bytes. Finally the 32 bits payload length field specifies the payload size that could be maximum 4,294,967,295 bytes,. Routing extension header: The use of the extension header is typically that of the LSRR (Loose Source and Record Routing) option field of the IPv4. This extension header is used to provide router addresses in order that the IPv6 packets may follow to reach the destination node. This is useful for sending packets by the defined routing paths without any variation. This is also useful when the default routing link or router is out of order, in which case the forwarding node defines/ adds the new routing address in the extension headers so as to make the packet to reach the destination. The illustration is given in Fig. (43 f) Fragment extension header: It typically allows fragmentation of the packets for the purposes as in IPv4. The default minimum IPv6 packet size is 1280 bytes. When the sender discovers that the receiving node is on a network that has MTU (Maximum Transfer Unit) is less than 1280 bytes, the sender fragments the packets to make it possible for transfer over the receiving network. The fragmentation information is conveyed to the receiving node by the fragment extension header. For example when the packet is fragmented, each part of that packet is assigned the same identification number. The identification number is of 32 bits data in the variable extension header field. Illustration is given in Fig. (43 e). Is there any difference between the fragmentation in IPv4 and IPv6? Yes, there is. In IPv4, sender or any intermediate node/router may fragment the packet if the network over which the packet to travel has lower MTU size than the packet size. But in IPv6, it is only the sender or the original source can fragment the packet if so required. Actually there is a technique known as path MTU discovery technique by which the source can find the smallest MTU on the path. Using the information so obtained the source may fragment the packet. However if the path MTU discovery is not run, the source may fragment the packet to smallest MTU size of 576 bytes that is the minimum size the networks support those are connected in the Internet. Authentication extension header: This is used to authenticate the receiving node about the received IPv6 packet, meaning that the original packet as was sent by the sender is received in tact. The extension header if included, the variable extension header fields will include the authenticated code (may be MD5 digest or digest by hash function- see chapter of data security) of all headers, except the headers that change and the payload. Note that the Hop Count field will change at each hop. That is why for digest, the hop count field is not included. Or otherwise the authenticated code for hop count is taken as 0. Illustration of this header is given in Fig. (43 g) Encapsulating security payload(ESP) extension header: This is used to provide data security to the payload only. This extension header supports the secret key encryption namely DES. If the extension header is included, the sender provides the security information in the variable extension header fields. The receiver using the information in the variable extension header fields, decrypts the payload. The ESP uses a header and tailor as in Fig. (43h). Very often, the authentication and the encapsulating security extensions are used together. Whereas authentication ensures the integrity of the packet as a whole, the security ensures the confidentiality of the payload. Illustration is given in Fig. (43 i) It is found that several fields present in IPv4 are no longer present in the IPv6; and notably among them are:

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• Checksum field: The main issue of designing IPv6 was the fast processing of packets. This results in designing with fixed header fields and removing the redundant fields. The error check is done at upper layers namely TCP/UDP. As such the check sum field further at IP layer was assumed as redundant and accordingly it was removed from IPv6. Again with check sum at IPv4 packet, the error checking at every node was essential. It was a very time consuming and costly thing and duly unwanted at IPv6. • Options field: Dropping of options field has made the IPV6 a fixed header packet. Of course if required the IPv6 packet may use next header field for the purpose of header extension. • Fragmentation: The IPv6 version has dropped the fragmentation and reassembly feature at intermediate routers. The data is fragmented for packetization at the source only. The reassembly is done at destination only. If an IP packet received by any intermediate router is found as too large to be forwarded on the outgoing link, the router simply drops the packet; and in turn send a ICMP error message of “Packet Too Big” to the sender. Sender on receiving the ICMP error message of “Packet Too Big”, retransmit the data with smaller packet size. Actually the fragmentation and the reassembling the datagram at routers is a time consuming matter; and removing these from routers’ functions to end users’ functions, makes the network to speed up.

5.11 ICMP (INTERNET CONTROL MESSAGE PROTOCOL) ICMP for version IPv4 is used by hosts, nodes, routers and gateways to communicate network layer information to each other. ICMP is specified in RFC 792. ICMP information is carried as IP payload like TCP or UDP information. ICMP messages are basically used for error reporting among others (Table 24). An ICMP message is made of a type field and a code field and also the first eight bytes of the IP datagram for which the ICMP message is to be generated in the first place so that the sender can know the packet that caused the error. The ICMP message is sent as IP datagram (Fig. 44) with TOS field set to 0 and protocol field set to 1 that defines an ICMP pack. After the headers, ICMP message is included as payload. The ICMP message format (Fig. 44) has four parts: Type: 1 byte field specifies type of the message as in table (24) Code: 1 byte code field is specified as per table (24) Checksum: 2 bytes field is used to provide check sum (1’s complement addition) of all 16 bits headers. The field may be filled in all 0s if required. The additional information field depends upon the header of ICMP messages: • For timestamp request, the additional information is made of 2 bytes identifier, 2 bytes sequence number and 4 bytes originating timestamp • For echo request and reply, the additional information is made of 1 byte identifier, 1 byte sequence number and the original IP header • For destination not reachable etc., the additional information contains 4 bytes unused fields followed by the original IP header. A new version of ICMP defined (Fig. 44d) for IPv6 in RFC 2463. The new ICMP has the reorganized existing types and codes as well as added new types and codes. The added new ICNP type includes “ Packet Too Big”, and “unrecognized IPV6 options” among others.

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The different error reporting ICMP messages are as below: • Destination unreachable: When due to some reason (table 23), a node or a host is unable to deliver data to the destination, the node or host send this type of the ICMP to the source host. The typical problem that may arise is that when a router/node receives a data gram with DF (don’t fragment) bit set to 1, but network does not support size of the received datagram, ICMP message of type = 3 and code = 4 (see table 24) is sent. • Source quench message: The source host does not have any mechanism to know whether a transmitted datagram has actually reached the desired destination or not. This is because IP is a connection less protocol. A node/router or host may discard a data when there is congestion in the network or a condition of buffer full arises. In that case the node/router or host that is discarding the datagram sends the source quench (source slowdown) ICMP message to the sender. The sender on receiving the source quench message understand that there is a congestion in the network and accordingly slows down the transmission. • Time exceeded: A router/node will send the time exceeded ICMP message to the sender under any of the following conditions : (a) when a data gram reaches the router/node with TTL field set to 0 yet the datagram has not reached the destination, and (b) when all the fragments of a datagram are not reached within the time limit. • Parameter problem: Any error or ambiguity (semantic or syntactic) in the IP header creates problem in routing or forwarding the datagram. In that case, the datagram is discarded and the ICMP parameter problem message is sent to the sender. • Redirect message: Both routers/nodes and hosts use routing table in order to route the datagram. The routing table in routers and huge and the tables are constantly updated. The routing table of hosts are limited and static in nature. So the hosts may not route the data appropriately. The routers may suggest redirect ICMP message to inform better routing path (see question 5 latter in this section) The different ICVMP messages for management query are as follows: • Echo request and reply: This is used to check whether a communication is possible between entities ( routers/nodes or hosts). Echo request is obliged by echo reply. Both use identification and sequence numbers for correct acknowledgement and synchronization. • Time stamp request and reply: This is used to measure the delay in transfer of the datagram. • Address mask request and reply: When sub networks are in use, this is used to exchange subnet mask. • Router request and advertisement: This is used to add capability to the host to known the routing information. This is also used to check whether a router is alive / working or not. The router advertisement ICMP message takes the shape as in Fig. (43 c-7). It is as per RFC 1256 and includes the fields of (a) number addresses—the number of router addresses advertised in the current message, (b) address entry size—the number of 32 bits addresses for each router address, which is 2, (c) lifetime—The maximum time in seconds that the router advertisement remains valid, the default value of which is 1800 seconds, (d) router address i (1<= i <= number addresses)—the IP address or addresses of the sending router’s interface where from the message is sent, (e) preference level i (1<= i <= number addresses)—the preference of each router as a

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default router in relative to all other routers on the network. The value is given in 2’s complement signed scheme. The higher the value, the higher is the preference (see problem 6 in latter of this section). The router solicitation or reply (Fig. 43 d-9) is issued when a router becomes just active. It accepts advertisement of solicit the information from all other attached routers to it. BOX 13

QUESTIONS 1. What layer protocol is the ICMP? It is a network layer protocol. But it does not directly pass data to the data link layer. The ICMP data is encapsulated in IP datagram before being passed to data link layer, as in Fig. (1). ICMP Message/Data IP Header

IP data/payload Fig. 1

2. Why does the Internet require ICMP? IP has no provision of error reporting and has no mechanism for management query. The ICMP supplements these deficiencies of the IP as in Fig. (2). ICMP functions → Error reporting 1. Destination unreachable 2. Source quench message 3. Redirect message 4. Parameter problems 5. TTL field expires / All fragmentation do not reach before within time out → Query Messages 1. Echo request & reply 2. Time stamp request and reply 3. Information / Address mask request and reply 4. Router advertisement and reply Fig. 2

3. Where to ICMP message is sent? ICMP error reporting message is always addressed to the original source. Any node in the network may send ICMP query message to any desired destination. 4. How does a receiving station understand that an IP datagram carries an ICMP message. The version field of IP is set to 1, when the IP payload is an ICMP message. This indicates that the IP datagram is carrying an ICMP message.

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5. In reference to Fig. (3), say the host F on LAN 3 likes to send a message to host A on LAN 1. When and what ICMP message may arrive at such transfer. If F sends data to Router 2 for transport to destination host A on LAN1, the Router 2 will find that it will be easy for this transport via Router 3. Then Router 2 will send an ICMP redirect message to the sender with advice that the data may be sent via shorter path of Router 3. The address of Router 3 is contained in the parameter field of the redirect message. A

Router 3

LAN 1 Router 1 B Internet

Router 2

LAN 2

LAN 2

C E

D

E

Fig. 3

6. How does a recently activated or alive host issues router solicitation? It may use the broadcast address to alert all other routers by using 255.255.255.255. 7. Do you think IPv6 may meet some of the requirements of Personal Communication Network (PCN)? Yes, definitely. The great philosophy behind the concept of Personal Communication Network (PCN) is to connect people rather than machine. But how? This is following the natural communication we do in our day to day life. With the type of communication we do in our day to day life, out of many others one unique thing is that: if you are called by “Sweet” at Kolkata, your are called by the same name “Sweet” elsewhere in the world. Similarly if your name is “Sabuj”, you will be known by “ Sabuj” everywhere in the planet. But your telephone number changes from place to place. The unique global addressing is a fundamental requirement of the PCN. IPv6 quadruples the addresses from 32 bits to 128 bits with which it is predicted that unique IP addresses for every network device on the planet are possible. Thus it may meet a component of requirement of PCN.

5.12 BOOTSTRAP PROTOCOL (BOOTP) By the RARP protocol, the information about the IP address corresponding to a given physical address of the machine is obtained. But the RARP does not provide the router or gateway

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address, server address etc. BOOTP is therefore developed to provide additional information to a caller. The BOOTP message goes as a payload of UDP layer. The message format is shown in Fig. (45a). The fields are as below: (a) Operation field is set 1 and 2 respectively for a request and a reply (b) The hop field is set to 0 in the request message. When the message is passed from one server to another server the hop count is incremented by 1. (c) Transaction ID field is used for the sequential co ordination between the request and the reply messages just like the sequence number in the IP datagram (d) The seconds field is used to count the time in seconds since the host starts BOOTP. (e) Others field are self explanatory. The vendor specific fields are not yet standardized. With the start of the BOOTP procedure, a request is sent to the server, and a timer is made on. If no reply is reached within a defined time period i.e the time limits expires, BOOTP should attempt retransmission.

5.12.1 Auto configuration and multiple IP addresses/DHCP IPv4 address structure is a stateful address structure, which means that if a node moves from one subnet to another the user has either to reconfigure the IP address or to request for a new IP address from DHCP (Dynamic Host Configuration Protocol). With DHCP, an IP address is leased to a particular host or computer for a defined period of time. But IPv6 supports a stateless auto configuration whereby on moving from a subnet to another subnet a host can construct its own IP address. This is done by host on adding its MAC (Media Access Control) address to the subnet prefix. IPv6 also supports multiple addresses for each host. The addresses can either be valid, deprecated or invalid. With valid address new and existing communication may be done. With deprecated address, the existing communication may be done. With invalid address no communication is done. When a host desires to get an IP address, a DHCP discover message is broadcasted over the physical network with broadcast IP address 255.255.255.255 (Fig. 45 c). The discover message will be received by all the routers and other hosts and they will ignore it. Only the DHCP server/router shall reply. Others routers will discard the discover message thereby preventing the whole of Internet flooded with discover message. The DHCP server in the network may reply with a DHCP offer message. The offer message provides an IP address and other information relating to configuration. It is not required that each network shall have a DHCP server. The network may have DHCP relay agent. The relay agent may request the remote server for reply to be subsequently sent to the requesting host. If the requesting host receives a number of offers from a number of DHCP servers, the host will accept one offer and acknowledges the same to the corresponding server. The corresponding server then acknowledges the reply of host and now the host uses the IP for address for communication.

5.13 IPV6 AND IPV4 ADDRESS COMPATIBILITY Earlier we discussed the address notation of IpV6. Like IPv4, the IPv6 has special notation for representing the IP addresses. The IPv6 address is represented by hexadecimal colon notation. The 128 bits are divided into eight sections each of two bytes in length. Each of the eight sections is represented in four hex digits (or a pair of hexagonal numbers separated by a colon. A pair of hex means a byte) and is separated by a colon. One example is: AB12:0978:CF56:00FE: 1234:127E:CB65:7890

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The notion allows to drop leading zeros. This means and for example 0045 can be just represented as 45, and 0A456 can similarly be represented as A456, and 0000 as simply 0. The notion also allows removing a zero leaving a colon, and therefore for example 2456:AC67:0:0:67:D4E5:A456:A678 can be written as 2456:AC67::67:D4E5:A456:A678. The stated double colon notation can be used at the beginning or at the end of an address but only once. The double colon at the start indicates leading zeros and that at the end indicates contiguous zeros at the end. If more that one location double colons are used, it will not be possible to know how many zeros are there at a particular double colon location. This is why double colon notation is used only once. By counting the other bytes, the number of zeros at the single double colon location can be found out. For a long interim period, the IPv6 and the IPv4 have to coexist. During this period, an IPv4 address can be converted to an IPv6 address by pre pending 12 bytes of zero. For example, an IPv4 address 126.34.67.10 will be converted to an IPv6 address as 0:0:0:0:0:0:0:0:0:0:0:0:126.34.67.10 or::126.34.67.10. Similarly a host having an IPv4 address as 128.67.56.9 may be mapped (read as IPv4 mapped IPv6) could have an IPv6 address as ::AC45:128.67.56.9. The different special notations of version 4 and version 6 will make them separable. Version (4 bits)

Priority (4 bits)

Flow label (24 bits)

Payload Length (16 bits)

Next Header (8 bits)

Hop Limit (8 bits)

Source Address (128 bits) Destination Address (128 bits) Next header

Header length (8 bits)

Variable header fields

… …………………………… Next Header

Header length

Variable header fields………….

…………………………………… Variable length TCP pack (which is TCP header + Payload)…….

(a) Illustration of use of Next Header Fields IPv6 Header Next Header = TCP

TCP Header +Data

IPv6 Header Next Header = Routing

Routing Header Next Header = TCP

TCP Header + Data

IPv6 Header Next Header = Routing

Routing Header Next Header = Fragment

Fragment Header Next Header

Fragment TCP Header

= TCP

+ Data

(b) Illustration of the next header extension

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Code Value

Next Header

0

Hop by Hop

2

ICMP

6

TCP

17

UDP

43

Source Routing/Routing Header

44

Fragmentation/Fragment Header

50

Encryption of payload

51

Authentication Header

52

Encapsulating Security Payload Header

59

No next header/Null

60

Destination option

(c) Code values for Next Header Field Next Header (8 bits)

0 (8 bits)

194 (8 bits)

Option length (8 bits) = 4

Payload Length (32 bits)

(d) Extension header for large/jumbo packet Next Header (8 bits)

Reserved (8 bits)

Fragment offset (13 bits)

Reserved (2 bits)

M (1 bit)

Identification (32 bits)

Fragment offset, one bit more fragment (M) and Identification are used exactly as in Ipv4. Only the size of the identification field has increased to 32 bits. NOTE THAT THE RESERVED FIELD OF 8 BITS OCCUPY THE HEADER LENGTH FIELD. WE TOLD EARLIER that THE HEADER LENGTH FIELD IS NOT USED in fragmentation header. (e) Fragmentation extension header Next Header (8 bits)

Header length (8 bits)

Reserved (8 bits)

Strict/Loose bit mask (24 bits)

Address Address ……………………… Address

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Header length specify the length of routing extension header in unit of 64 bits, excluding the first 64 bits. The segment field specifies the number of route segments left before the destination is reached. The maximum permissible value is 23. Initial segment value will be the total number of route segment from source to destination. The bits in the strict / loose bit mask indicate the type whether the next routing be followed strictly or loosely. If all bits are 1s, it is strictly. If all bits are 0s, it is loosely. (f) Routing extension header. Next Header (8 bits)

Header Length (8 bits)

Reserved (16 bits)

Security Parameter Index(SPI) (16 bits) Sequence Number (16 bits) Authenticated code/Digest (Variable length in multiple of 32 bits)

SPI defines virtual circuit identifier and is same for all packets under a connection Sequence Number is used to avoid playback. A sequence number does not change even if the packet is retransmitted. (g) Authentication extension header format IP header Security (SPI)

ESP Header Parameter

Payload

ESP tailor

Index

Sequence Number Information

Information + padding

+ padding

Header length (8 bits)

Next Header (8 bits)

(h) ESP extension header ← Encrypted → IP Header

ESP Header

Payload

ESP Tailor

Authenticated code

← Authenticated → (i) Authentication and ESP together Fig. 43: Extension of Header in Ipv6 / Possible values of next header field

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Table 21: Reports of different studies on IPv4 address space run up Study group

Recommendation

Two leaders of IETF Address Lifetime Expectations (ALE)’s recommendation

Pv4 address space would be exhausted I in 2008 and 20018 respectively

Final recommendation of ALE in 1994

IPv4 address space will be exhausted at some time between 2005 to 2011.

American Registry for Internet Numbers (ARIN)’s report in 1996

All class A Address has been assigned; 62% of class B address and 37% of Class C address have been assigned

Table 22: Different IP version labels Value

Key

Description

0 4

IP

Reserved Internet Protocol (RFC 791)

5

ST

ST datagram Mode (RFC 1190)

6

SIP

Simple Internet Protocol (IPv6)

7

TP/IX

TP/IX: The Next Internet

8

PIP

The P Internet Protocol

9

TUBA

TUBA

10-14

Unassigned

15

Reserved

Table 23: IPv6 address space subdivision based on prefix assignments of bits Prefixed bits

Use of address space

0000 0000

Reserved

0000 0001

Unassigned

0000 001

Reserved for NSAP application

0000 010

Reserved for IPX application

0000 011 0000 1 0001

Unassigned Unassigned Unassigned

001

Aggregatable Global Unicast address

010 011 100

Unassigned Unassigned Unassigned

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101 110 1110 1111 1111 1111 1111

0 10 110 1110 0

267

Unassigned Unassigned Unassigned Unassigned Unassigned Unassigned Unassigned

1111 1110 10

Addresses for Link Local use

1111 1110 11

Addresses for Site Local use

1111 1111

Multicast addresses

Table 24: Selected ICMP messages ICMP

CODE

Remarks

Additional information

0

0

Echo reply (to ping)

16 bit identifier, 16 bit sequence number

3

0–network unreachable 1-host unreachable 2-protocol unreachable 3-port unreachable 4-fragmentation needed 5-source route failed

Destination unreachable

32 bits unused, Original IP headers, 64 bits of original payload

4

0

Source quench message (Congestion Control)

As in Destination unreachable

5

0-for the network 1-for the host 2-for the type of service & network 3-for the type of the service and host

Redirect message

32 bits gateway address Internet Header, 64 bits original payload

8

0

Echo Request/Reply

16 bit identifier, 16 bit sequence number, optional data

11

0-time to live exceeded on transit 1-fragmentation reassembly time exceeded

Expiry of TTL (Time to live)

32 bits unused, Internet Header, 64 bits of original payload

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12

0-indicates error

Parameter problem

8 bits pointer, rest 24 bits unused, Internet Header, 64 bits original payload

13

0

Time stamp Request

16 bits identifier, 16 bits sequence number, 32 bits original time stamp,

14

0

Time stamp reply

16 bits identifier, 16 bits sequence number, 32 bits original time stamp, 32 bits receive timestamp, 32 bit transit timestamp

15

0

Information request for address mask

16 bits identifier, 16 bits sequence number

16

0

Information reply

As in information request

for address mask

0 (bit)

3,4

7,8

Version (4)

Length(4)

15,16 Type of service (8) =0

Identification (16) Lifetime (8)

31

Total length (16) Flags (3)

Protocol (8) = 1

Fragment offset (13)

Header checksum (16)

IP source address (32) IP destination address (32) Options (variable)

……

…….

Padding (variable)

(a) IPv4 Headers for ICMP message IP headers

Type (8 bits)

Code (8 bits)

Checksum (16 bits)

Additional information

(variable)

(b) ICMP Message format that is a payload in the ICMP based IP datagram Type

Code

Checksum

Unused IP header + 64 bits original datagram (1) Destination Unreachable; time exceeded; source quench

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Code

Identifier

Checksum Sequence number

Originatetimestamp (2) Timestamp

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Type

Code

Checksum

Pointer

Type

Unused

Code

Checksum

Identifier

IP header+ 64 bits original datagram

Sequence number

Originate timestamp Receive timestamp

(3) Parameter problem Type

Code

Checksum

Transmit timestamp

Gateway IP address

(4) Timestamp reply

IP header + 64 bits of original datagram (5) Redirect Message Type Num Addrs

Code

Checksum

Entry size

Lifetime

Type

Code

Checksum

Identifier

Sequence number

Router address 1

Optional data

Preference level 1

(6) Echo; echo reply Type

M Router address n

Code

Checksum

Identifier

Preference level n

Sequence number

(8) Address mask/Information request

(7) Router advertisement. Type

Code

Checksums

Type

Unused

Code

Identifier

Checksum Sequence number

Address mask (9) Router Discovery

(10) Address mask/information reply.

(c) Different ICMP Illustration IGMP

ICMP

ICMPv6

IPv4 ARP

IPv6 RARP

Network layer in version 4

Network layer in version 6

(d) ICMP in IPv4 and in IPv6 Fig. 44: ICMP Message format

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IPv6 defines a set of addressing that is different from IPv4 addressing. IPV6 doses not include special addressing for broadcasting on a given network rather IPv6 address includes one of the following three types: Unicast, Multicast and Anycast. Unicast: The address is meant for a single computer or node. A datagram under such addressing is routed through the shortest path. Multicast: The address refers to a set of computers or nodes possibly on different geographic locations. The membership of the set may change from time to time. One copy of the datagram under such addressing is delivered to each member of the set. Anycast: The address refers to a set of computer or nodes that all reside in the same location. The members of the set share a common address prefix. A datagram under this addressing is routed through the shortest path and is delivered to one of the members of the set i.e the closet member to the sender. Anycast addressing was originally known as cluster addressing. Operation (1 byte) Hardware Type (1 byte) Hardware Length (1 byte) Hops ( 1 byte) Transaction ID (4 bytes) Seconds (2 bytes) Client IP address (4 bytes) Your IP Address (4 bytes) Server IP Address (4 bytes) Router/Gateway IP Address (4 bytes) Client Hardware Address (16 bytes) Server Host Name (64 bytes) Boot file name (128 bytes) Vendor Specific Field (64 bytes)

(a) BOOTP Message format/ BOOTP message is sent using UDP Operation

Hardware type

Hardware

lengthHops

Xid (Transactions ID) Seconds

Flags Client IP address Your IP address Server IP address

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Router/Gateway IP address Client hardware address (16 bytes) Server host name (64 bytes) File (128 bytes) Operation

(b) DHCP packet format/DHCP message sent using UDP Other network Broadcast Host

DHCP relay agent

DHCP server Router

Other network

Host

(c) A relay agent receives a broadcast DHCPDISCOVER message from a host and sends a unicast DHCPDISCOVER to the DHCP server. Fig. 45: BOOTP/DHCP

Question. We find after IPv4 it is IPv6. Where is then IPv5? Why IPv5 is missing?

5.14 CO EXISTENCE OF IPV4 AND IPV6/DUAL STACK & TUNNELING How will the existing IPv4 Internet be made to work with coming up IPv6? What will be the technique of migration from IPv4 to IPv6? In the intermediate phase how do IPv4 and IPv6 co exist and work in cohesion? There are several options to these problems. Application

TCP Ipv4

Ipv6 Ethernet

Fig. 46: Typical Dual Stack Configuration

The first and the most straight forward solution will be to declare a off day when all machines, hosts and users will not use Internet, and IPv4 will be made all upgraded to IPv6. But this is not technically feasible and unthinkable solution considering the involvement of millions and millions users, machines, and networks.

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RFC 1933 proposes two other solutions to tackle the problems: Dual stack (Fig. 46) and Tunneling. While deploying, the IPv6 nodes can be made “backward compatible” thereby making it capable of sending IPv4 datagram, but already existing IPv4 nodes are not capable of sending IPv6 datagram. In the dual stack, each node is made IPv6/Ipv4 compatible. The IPV6 node is made to implement a complete IPv4 as well. Such node can work both in IPv4 and in IPv6 has both Ipv4 and Ipv6 addresses. The node also has the capability to determine whether the next node of routing is IPv4 or Ipv6. This may be done by the DNS. The dual stack is in IPv6 nodes, but existing IPv4 does not have this provision The dual stack approach is pictorially be looked as in Fig. (46). Let us consider a network as in Fig. (47). The network is made of a few IPv6 and a few IPv4 nodes. The network is chosen such that all possible transitions are possible namely: IPv6 to IPv6 : Node N1 to Node N2 Ipv6 to Ipv4: Node n2 to Node N3 Ipv4 to Ipv4: Node N3 to Node N4 Ipv4 to Ipv6: Node N4 to Node N5

N1/IPv6

N2/IPv6

N3/IPv4

Logical tunnel

N5/IPv6

N4/IPv4

Fig. 47: Example Network

Assume that the node N1 sends a IPv6 datagram to node N5. the node N1 will send IPV6 datagram to node N2. The node N2 will apply dual stack provision to convert IPv6 pack to Ipv4 pack as the next node N3 is IPv4. By the process some IPV6 compatible information like flow label will be missed. The node N3 will send IPv4 pack to next node N4 that is an IPv4 one. Now the node N4 will send IPv4 pack to the node N5. The node N5 being an IPv6 node will be able to receive the pack under dual stack operation. But the pack is received as IPv4 not as original IPv6. So the received pack is with some loss of data. This is the disadvantage of the dual stack operation. The operation between the node N2 to node N5 is as if a logical tunnel. The problem of the dual stack as illustrated in the above example may be solved with the concept of tunneling. In the tunneling the IPv4 nodes treats any IPv6 packet as a complete payload for subsequent routing. In reference to Fig. (47), when the node N2 sends the IPv6 data gram to the node N3, the whole IPv6 is taken as a payload and the IPv4 data gram is made with the payload. The node N3 and node N4 remain ignorant about this and transfer the packet with IPv4 header information to the node N5. The node N5 extracts the IPv6 pack from the payload of received IPv4, and takes subsequent decision for forwarding or accepting.

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BOX 14

Interoperability-Some Idea Version Field Indicated Tunneling for Interoperability between Ipv4 and Ipv6 Introduction As of today the vast majority of Internet installation is based on IPv4. To cope up with the exponential growth of Internet users that may exhaust the IPv4 address space shortly, the new version of IP namely IPv6 has been specified [1-10]. During the transition period from IPv4 to IPv6, both versions of hosts, nodes, routers and networks will coexist. Therefore there emerges an area of investigation on the routing techniques in the mixed environment. The two techniques that have been recommended in RFC 1933 are dual stack and tunneling. This short paper reviews these techniques and proposes a new technique. Existing techniques for interoperability between IPv4 and IPv6/Dual Stack & Tunneling How will the existing IPv4 Internet be made to work with coming up IPv6? What will be the technique of migration from IPv4 to IPv6? In the intermediate phase how do IPv4 and IPv6 co exist and work in cohesion? There are several options to these problems. The first and the most straight forward solution will be to declare an “off day” when all machines, hosts and users will not use Internet, and all IPv4 bases will be made upgraded to IPv6. But this is not technically feasible and is an unthinkable solution considering the involvement of millions and millions users, machines, and networks. Application TCP Ipv4 Ipv6 Ethernet Application

TCP Ipv4

Ipv6 Ethernet

Fig. 1: Typical Dual Stack Configuration

RFC 1933 proposes two other solutions to tackle the problems: Dual stack and Tunneling [9-16]. In the dual stack approach it is proposed that all the IPv6 nodes can be made “backward compatible” thereby making it capable of sending IPv4 datagram too. In the dual stack, each IPv6 node is made IPv6/IPv4 compatible (Fig. 1). The IPv6 node is made to implement a complete IPv4 as well. Such node can work both in IPv4 and in IPv6 has both Ipv4 and Ipv6 addresses. The node also has the capability to determine whether the next node of routing is IPv4 or IPv6. This may be done by the DNS. The dual stack concept is for IPv6 nodes, but existing IPv4 does not have this provision. We may consider a network as in Fig. (2). The network is made of a few IPv6 and a few IPv4 nodes. The network is chosen such that all possible transitions are possible namely:

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IPv6 IPv6 IPv4 IPv4

to to to to

IPv6 : Node N1 to Node N2 IPv4: Node n2 to Node N3 IPv4: Node N3 to Node N4 IPv6: Node N4 to Node N5

N1/IPv6

N2/IPv6

N3/IPv4

Logical tunnel

N5/IPv6

N4/IPv4

Fig. 2: Example network

Assume that the node N1 sends a IPv6 datagram to node N5. the node N1 will send IPV6 datagram to node N2. The node N2 will apply dual stack proviso to convert IPv6 pack to Ipv4 pack as the next node N3 is IPv4. By the process some IPV6 compatible information like flow label will be missed. The node N3 will send IPv4 pack to next node N4 that is an IPv4 one. Now the node N4 will send IPv4 pack to the node N5. The node N5 being an IPv6 node will be able to receive the pack under dual stack operation. But the pack is received as IPv4 not as original IPv6. So the received pack is with some loss of data. This is the disadvantage of the dual stack operation. The operation between the node N2 to node N5 is as if a logical tunnel. The problem of the dual stack as illustrated in the above example may be solved with the concept of tunneling. In the tunneling the IPv4 nodes treats any IPv6 packet as a complete payload for subsequent routing. In reference to Fig. (2), when the node N2 sends the IPv6 data gram to the node N3, the whole IPv6 is taken as a payload and the IPv4 data gram is made with the payload. The node N3 and node N4 remain ignorant about this and transfer the packet with IPv4 header information to the node N5. The node N5 extracts the IPv6 pack from the payload of received IPv4, and takes subsequent decision for forwarding or accepting. The problem of recognizing the payload, received under tunneling as a full IPv6 pack rather than conventional payload is to be addressed. One possible solution is that when any packet is received at IPv6 from IPv4 node, the payload be treated as IPv6 pack. But that may create two other problems: (1) any Ipv4 packet may by the process be lost out that is undesirable, and (2) for the correct operation the entire the mixed environment must use the tunneling technique alone which may not be the case due to different political, administrative and geographical reasons.

Proposed Technique of Version Field Indicated Tunneling We propose a solution based on the version field, the first 4-bits header both in IPv4 and IPv6 (Fig. 3). The version field indicates the version of IP used in the datagram as per conventional notation shown in table (1). Two reserved version namely 0 (0000) and 15 (1111), may be used as follows: 0000 to indicate a data gram in IPv4 version when payload is a full IPv6 pack 1111 to indicate a data gram in IPv6 version when payload is a full IPv4 pack. where use is made of reserved fields

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or

alternatively other options may be used like: 0001 to indicate a data gram in IPv4 version when payload is a full IPv6 pack 0010 to indicate a data gram in IPv6 version when payload is a full IPv4 pack. where use is made of fields 1 and 2 (3 and 4 may also be used) or 1010 to indicate a data gram in IPv4 version when payload is a full IPv6 pack 1011 to indicate a data gram in IPv6 version when payload is a full IPv4 pack. where use is made of unassigned fields (12,13 and 14 may also be used). Comparison of the Techniques The natural attractions of the proposed technique are many: any node irrespective of version may tunnel any IP pack to any other node of any version, necessity of dual protocol as in dual stack is absent, no loss of any information occurs, no need of knowing the version of sending node to recognize the payload as in conventional tunneling exist. The only limitation of the proposed technique may be the increase in overhead bits due overhead bits being combined in the payload for the tunneling. The increase in overhead bits decreases the coding efficiency. For an analytical comparison of the decreased coding efficiency, we assume: (1) the traffic load is equally shared (50% both) between IPv4 and Ipv6 datagrams in the network, (2) the network has equal number of IPv4 and IPv6 nodes, and (3) all the datagrams of both versions carry minimum header fields (20 bytes for IPv4 and 40 bytes for IPv6), then for a payload of m bytes: Coding efficiency for conventional tunneling = [{{m/(m + 20)} + {m/(m + (40 + 20))}]/2 × 100 % Coding efficiency for proposed tunneling = {m/(m + (40 + 20)} × 100% Thus in the proposed technique the coding efficiency decreases in relative to the conventional tunneling by [m/(m + 1)(m + 3)] × 100%. Our first two assumptions are reasonable and the third assumption is also reasonable as this provides the best coding efficiency. The derivation shows that with higher traffic load, the relative decrease is insignificant, which would be the case. Therefore this limitation is not so bearing on the technique. A comparison of the techniques is given in table (2). Besides, the proposed solution has to take care of address translation for tunneling that may be done by DNS or by the look up table as used in ARP resolution. The proposed tunneling may be used as a simple solution for all purposes of interoperability without geographic restrictions. 0 3,4 7,8 15,16 31 (bit) Version (4)

Length(4)

Type of service (8)

Identification (16) Lifetime (8)

Total length (16) Flags (3)

Protocol (8)

Fragment offset (13)

Header checksum (16)

IP source address (32) IP destination address (32) Options (variable)

……

…….

(a) IPv4

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Version (4 bits)

Priority (4 bits)

Flow label (24 bits)

Payload Length (16 bits)

Next Header (8 bits)

Hop Limit (8 bits)

Source Address (128 bits) Destination Address (128 bits) Variable length TCP pack (which is TCP header + Payload)……..

(b) IPv6 Fig. 3: IP Packet Formats

Table 1: Different IP version labels Value

Key

0

Description Reserved

4

IP

Internet Protocol (RFC 791)

5

ST

ST datagram Mode (RFC 1190)

6

SIP

Simple Internet Protocol (IPv6)

7

TP/IX

TP/IX: The Next Internet

8

PIP

The P Internet Protocol

9

TUBA

TUBA

10-14

Unassigned

15

Reserved

Table 2: A comparison of the techniques for interoperability Loss of Information

Possibility of pack wrongly accepted

Tunneling al through

One solution for global coverage

Overhead

Dual Stack

Inherently associated

Possible

Not applicable

May not be possible

Overall may not increase

Conventional tunneling

No

Possible

No, only over IPv4

May not be possible

May increase

Proposed Tunneling

No

No

Yes

Possible

Increases

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REFERENCES 1. http://www.fnc.gov/Internet_res.html 2. U. Black, TCP/IP and Related Protocols, 2nd edition, McGrawhill, 1995. 3. D. Comer, Internetworking with TCP/IP: Principles, Protocols and Architecture, Prentice Hall, 1988. 4. C. Huitema, Ipv6:The New Internet Protocol, Prentice Hall, 1996. 5. D. Sanghi, The Internet Protocol: From v4 to v6, Proceedings of COMNAM-2000, 21-22, December’2000, pp. 12-15. 6. C.T. Bhunia, Internet to Internet2, Electronics for You, Jan. 2000. 7. C.T. Bhunia, Personal Communication, IETE Journal of Education, Vol. 38, No. 2, April-June 1997, pp. 109-118. 8. C.T. Bhunia, The World of Narrow band to Broadband Networks, EFY, June 1996, pp. 96-105. 9. C.T. Bhunia, Communication World Over, EFY, Aug. 2000, pp. 89-104. 10. C.T. Bhunia, A Global Network for Integrated Services, CSI Communication, Sept. 1995, pp. 25-35. 11. C.T. Bhunia, Packet Switched Data Networks, CSI Communication, May 1995, pp. 17-21. 12. Ammar Rayes et al, Integrated Management Architecture for IP-Based Networks, IEEE Communication Magazine, Vol. 38, No. 4, April 2000, pp. 48-53. 13. Larry Lange, The Internet, IEEE Spectrum, Jan. 1999 pp. 35-40. 14. Huitema, IPv6: New Internet protocol, Prentice Hall, 1997. 15. Paul T. Ammann, Managing Dynamic IP Networks, Tata McGrawhill, 2000. 16. Peterson and Davie, Computer Networks, Asis Harcourt Ltd. 2001.

BOX 15

OBJECTIVE TYPE QUESTIONS Objective Questions on: Data/Queue modeling Tick the correct answer : 1.

2.

3.

4.

Under Poisson process, the probability of arrival of 1 message in 0.1 second when average arrival rate is 10 messages per second is : (a) 1/10e (b) e/10 (c) 1/e (d) none of the above. If the average arrival rate and the average service rate are respectively 10 messages per second and 100 messages per second , the load factor is : (a) 10 (b) 0.01 (c) 0.1 (d) none of the above. Under M/M/1 model, load factor is (a) the probability that the system/queue is not empty (b) the probability that the system/queue is empty (c) the probability that the system has 10 messages in queue (d) none of the above. If N, T and ? are the average number of messages waiting in the system, the average delay a message faces in the system, and average arrival rate; the Little’s formula says: (a) T = N. λ (b) λ = N.T (c) N = λ .T (d) none of the above.

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INFORMATION TECHNOLOGY, NETWORK AND INTERNET

For a stable system : (a) average arrival rate (λ) > average service rate (µ) (b) λ = µ (c) λ >= µ (d) µ > λ. In a M/M/1 node, the probability that there are k messages or more is given by : ∝

∝

(a)

(b)

∑

(d) (1 – ρ)

k

k=0 ∝

(c)

∑ (1 − ρ)ρ

∑P

k

k=0

∝

(1 − ρ)/ρk

k

∑ρ

k

k

where ρ = load factor. 7.

8.

9.

10.

11.

12.

13.

14.

In M/M/1 node, the average number of the customers waiting in the queue (λ = average arrival rate, µ = average service rate) is : (a) [{1/(µ – λ)} – (1/µ)] (b) {1/(µ – λ)} (c) [{1/(m – λ)} – (1/λ)] (d) none of the above. The service rate is c bits per second, and the packet size is µ packets per bit. What is the average delay suffered by the packet arriving at a rate of ?, if the system is M/M/1 type : (a) 1/(µc – λ) (b) 1/{(c/µ) – λ} (c) 1/(µ – λc) (d) 1/{1 – (c/λ)}. The average arrival rate and the service rate to a node are respectively 6000 packets per minutes, and 24 KBPS. The packet size is 200 bits. The average number of the bits and the messages laying in the system of M/M/1 type are respectively. (a) 10 and 5 (b) 20 and 10 (c) 1000 and 5 (d) 5 and 5. For the system of question (9), the average delay suffered by a packet and a bit are respectively : (a) 0.5 see and 0.25 × 10–2 sec (b) 0.05 sec and 0.25 × 10–3 sec (c) 0.5 sec And 0.5 sec (d) 0.25 × 10–3 sec and 0.25 × 10–3 sec. M/M/1 refers to a : (a) multiserver queuing model with Poisson arrivals and exponential service times, (b) multiserver queuing model with exponential arrivals and Poisson service times, (c) single server queuing model with Poisson arrivals and exponential service times, (d) none of the above. a multiserver queuing model with general independent arrivals and exponential service times is denoted by : (a) M/M/N (b) M/G/N (c) G/M/N (d) G/M/1. If two independent Poisson streams with average arrival rate of λ1 and λ2 are merged, the resulting stream is : (a) Poisson with an average arrival rate of λ1. λ2 (b) Poisson with an average arrival rate of λ1 + λ2 (c) General independent stream with an average arrival rate of λ1. λ2 (d) General independent stream with an average arrival rate of λ1 + λ2. In Fig. 1), Poisson stream with average arrival rate is divided among two links. If the division is done by randomization, each link behaves like a queuing model of type : (a) M/M/1 (b) M/M/2 (c) M/G/1 (d) M/G/2.

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15. In Fig. (1), if the division is done by metering, the whole system behaves like a queuing model of type : (a) M/M/1 (b) M/M/2 (c) M/G/1 (d) M/G/2. /2



A

B

/2

Fig. 1 16. Two M/M/1 queues are in tandem as in Fig. (2). The probability that in the steady state, there are n and m customers respectively at queue – 1 and queue – 2 is given by : (µ1 and µ2 – service times are mutually independent as well as independent of the arrival process. ρ1 = λ/µ1 and ρ2 = λ/µ2)

Queue 1 1



Queue 2



Fig. 2 (a) ρ1n (1 – ρ1) ρ2m (1 – ρ2) (c)

ρ1n

(1 –

ρ2)

ρ2m

(1 – ρ1)

(b) ρ1m (1 – ρ1) ρ2n (1 – ρ2) (d) ρ2n (1 – ρ1) ρ1m (1 – ρ2).

17. For Fig. (2), by Burke’s theorem, the departure process from queue-1 is : (a) Poisson

(b) General Independent

(c) Deterministic

(d) none of the above.

18. In Fig. 3), queuing networks of (a), (b) and (c) are respectively of the type of : (a) Simple tandem queue, traffic merging and traffic partitioning (b) Traffic partitioning, traffic merging and simple tandem queue (c) Traffic partitioning, simple tandem queue and traffic merging (d) Traffic merging, traffic partitioning and simple tandem queue. (Customers depart queue via path a with probability P) 19. For a M/M/N with all N identical servers, if is the utilization of each server, the utilization of the entire system can be considered as: (a) (N/ρ)

(b) Nρ

(c) Nρ

(d) (ρ/N).

20. For question (19), the theoretical maximum input rate is : (a) λmax = N/µ

(b) λmax = Mµ

(c) λmax = Nµ

(d) λmax = µ/N.

Customers depart queue via path A with probability P, thus path B with probability (1-P)

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A Queue B

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1

(a)

Queue

1 + 2

2

(b)



Queue 1 1

Queue 2





Fig. 3

Answers 1. (c) 9. (d) 17. (a)

2. (c) 10. (b) 18. (b)

3. (a) 11. (c) 19. (c)

4. (c) 12. (c) 20. (c)

5. (d) 13. (b)

6. (d) 14. (a)

7. (a) 15. (b)

8. (a) 16. (a)

OBJECTIVE TYPE QUESTIONS ON PROTOCOLS Choose the correct answer : 1.

2.

3.

4.

5.

6.

7.

ISO–OSI protocol has : (a) 4–layers (b) 3–layers (c) 7–layers (d) 5–layers ISO-OSI stands for : (a) International Standard Organization–Open system Interconnection (b) Indian Standard Organization–Open system Interconnection (c) International Standard Organization–Operating system Interface (d) Indian standard Organization–Operating system Interface Under OSI - ISO model, frame formation is made at : (a) data link layer (b) network layer (c) transport layer (d) physical layer Error - checking in OSI - ISO model is done at : (a) data link layer (b) network layer (c) transport layer (d) physical layer for telephone line, the physical layer interface is : (a) X-21 (b) X-25 (c) IEEE802.3 (d) RS-232-C RS-232-C is the interface between (a) Data link layer and Network layer (b) Data link layer and Physical layer (c) DTE-DCE (d) None of the above For digital link, the physical layer interface is : (a) X-21 (b) X-25 (c) IEEE802.3 (d) RS-232-C

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9.

DTE stands for : (a) Data Transmission Equipment (c) Data Transport Equipment

281

(b) Data Terminal Equipment (d) None of the above

DCE stands for : (a) Data circuit Equipment (b) Data Circuit-terminating Equipment / Data Communications Equipment (c) Data connection Equipment (d) None of the above.

10. MODEM is a : (a) DTE (b) DCE (c) Both DTE and DCE (d) None of the above 11. RS-232-C interface standard was recommended by : (a) IEEE of USA (c) CCITI 12. RS-232-C standard uses a (a) 12-pin connector

(b) ISO (d) Electronics Industries Association of USA (b) 9-pin connector

(c) 20-pin connector (d) 25-pin connector 13. RS-232-C signals are carried between DTE and DCE over a maximum distance of : (a) 30 m / 100 ft (b) 7.5 m / 25 ft (c) 15 m / 50 ft (d) none of the above 14. The maximum data rate under RS-232-C signals carried between DTE-DEC is (a) 200 KBPS (b) 100 KBPS (c) 20 KBPS (d) none of the above 15. HDLC and SDLC are each a (a) character oriented protocol (b) bit oriented protocol (c) code oriented protocol (d) none of the above 16. HDLC and SDLC protocols were recommended respectively by : (a) ISO and IBM (b) IBM and ISO (c) IBM and ITU (d) ITU and ISO 17. HDLC and SDLC respectively stand for : (a) High level Data Logical Control and Synchronous Data Logical Control (b) High level Data Link Control and Synchronous Data Link Control (c) High level Data Link Control and Synchronous Data Logic Control (d) High level Data Logic Control and Synchronous Data Link Control 18. The valid minimum size of a HDLC or a SDLC frame is (a) 6 bytes (b) 40 bits (c) 53 bytes (d) 48 bytes 19. The flag byte of a HDLC or a SDLC frame is : (a) 01111110 (b)10101010 (c) 01010101 (d) none of the above 20. To avoid the occurrence of opening and ending flag within the information and other fields, the technique used in practice is called : (a) frame check sequence (b) piggybacking (c) stuffing/destuffing (d) none of the above.

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21. An information byte, 01111110 when is stuffed at the transmitter before transmission, it looks like: (a) 10000001 (b) 01111110 (c) 011111010 (d) none of the above 22. A SDLC frame is an information frame when the control field has : (a) a “0” as first bit (b) “10” as the first two consecutive bits (c) “11” as the first two consecutive bits (d) none of the above. 23. HDLC has three types of frames. These are known as : (a) Small frame, medium frame and long frame (b) Control frame, information frame and management frame (c) Information frame, supervisory frame and unnumbered frame (d) None of the above. 24. The Fig. (1) is the block diagram of the : (a) Information frame (b) Supervisory frame (c) Control frame (d) Unnumbered frame 1 0

2

3

4

Send Sequence

5 P/F

6

7

8

bits

Receive Sequence

Number

Number Fig. 1

25. Supervisory commands and responses of supervisory frame are indicated by : (a) First and Second bits (b) Second and Third bits (c) Third and fourth bits (d) Fourth and Fifth bits 26. Supervisory frames of the types of Receive Ready, Reject, Receive Not Ready and selective Reject are indicated respectively by the bit-pairs as : (a) 11, 10, 01 and 00 (b) 10, 01, 11 and 00 (c) 11, 10, 01 and 00 (d) 00, 01, 10 and 11 27. The channel usually remains in active state between the transmission of frame. In this state, the transmitter continuously sends : (a) 10000001 (b) 11110000 (c) 01111110 (flag byte) (d) 00001111 28. The transmitter can abort a frame that it has started to send, by transmitting at least seven (a) contiguous 1s (b) contiguous 0s (c) 0110011 (d) 1001100 29. A channel enters idle state, when the transmission of 15 or more contiguous : (a) 1s are detected (b) 011001100110101 is detected (c) 111111110000000 is detected (d) 000000001111111 is detected 30. PDU stands for : (a) Packet Data Unit (c) Packet Data Unnumbered

(b) Protocol Data Unit (d) Protocol Data Unnumbered

31. Under OSI-ISO model, a host computer shall have : (a) all 7 layers (c) upper 4 layers

(b) lower 3 layers (d) lower 4 layers

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32. Under OSI-ISO protocol a node has functional coverage over : (a) all 7 layers (b) low e layers (c) upper 4 layers (d) lower 4 layers 33. Congestion control and flow control in ISO-OSI reference are done respectively by : (a) Data link layer and Network layer (b) Session layer and Transport layer (c) Transport layer and Network layer (d) Network layer and Transport layer 34. Security measure is taken in OSI-ISO model under : (a) application layer (b) Presentation layer (c) Session layer (d) Transport layer 35. The main function of network layer of OSI-ISO protocol is : (a) Framing and Error Detection (b) Framing and Switching (c) Switching and Routing’ (d) Routing and Framing 36. TCP / IP suite has : (a) 7 layers (b) 4 layers (c) 3 layers (d) none of the above 37. TCP and IP are respectively : (a) connection oriented and connectionless protocols (b) connectionless and connection oriented protocols (c) both connectionless protocols (d) both connection oriented protocols 38. IP address space is : (a) A0 64 BITS (b) 48 BITS (c) 32 BITS (d) 16 BITS 39. The layers of TCP / IP suite are : (a) Physical layer, Network layer, TCP layer and IP layer (b) Application layer, Transport layer, TCP layer and IP layer (c) Presentation layer, TCP layer, IP layer and Physical layer (d) Application layer, TCP layer, IP layer and Data link & physical layer 40. a gateway bridge has functional coverage over : (a) TCP layer and IP layer (b) IP layer and Application layer (c) Application layer and TCP layer (d) IP layer and Data link & Physical layer 41. Fig. (a), (b), (c) and (d) of Fig. (2) respectively depict the IP address scheme of (a) A, B, c and D class (b) B, C, D and A class (c) C, D, B and a class (d) D, C, B and a class 1110

28 bits (Multicast) (a)

110

21 bits (Networks)

8 bits (hosts)

(b) 10

14 bits (Networks)

16 bits (hosts)

(c) 0

7 bits (Networks) (d) Fig. 2

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42. The most important traditional TCP/IP services are : (a) file transfer, remote login and electronics mail (b) conventional data communication and electronics mail (c) conventional data and Internet telephony (d) none of the above. 43. an internet gateway connects two class c networks whose net-ids are 201.12.1.0 and 152.12.3.0 which of the following is a valid address for the gateway ? (a) Both the address : 201.12.1.0 and 152.12.3.0 (b) Any of the above given two addresses (c) Only 201.12.1.0 address (d) None of the above. 44. The PDUs exchanged between two TCP modules are known as segments. The sizes of source port, destination port, sequence number and acknowledgement number in a TCP segment are respectively : (a) all 16 bits (b) all 32 bits (c) 16 bits, 16 bits, 32 (d) 32 bits, 32 bits, 16 bits and 16 bits 45. Class A IP-address scheme provides (a) 224 hosts and 28 networks (b) 224 hosts and 27 networks (c) 216 hosts and 214 networks (d) 28 hosts and 221 networks

Answers 1. 9. 17. 25. 33. 41.

(c) (b) (b) (c) (d) (d)

2. 10. 18. 26. 34. 42.

(a) (b) (a) (d) (b) (a)

3. 11. 19. 27. 35. 43.

(a) (d) (a) (c) (c) (a)

4. 12. 20. 28. 36. 44.

(a) (d) (c) (a) (b) (c)

5. 13. 21. 29. 37. 45.

(d) (c) (c) (a) (a) (b)

6. 14. 22. 30. 38.

(c) (c) (a) (b) (c)

7. 15. 23. 31. 39.

(a) (b) (c) (a) (d)

8. 16. 24. 32. 40.

(b) (a) (a) (b) (d)

OBJECTIVE TYPE QUESTIONS ON SWITCHING & ROUTING Tick the correct answer : 1.

2.

3.

4.

5.

The trade-off parameter of the packet switching is : (a) overhead vs. speed (b) overhead vs. memory size (c) speed vs. memory size (d) speed vs. pipe-line effect The charge for customers in packet switching is estimated based on (a) time duration of the call (b) volume of data transferred during the call (c) both time duration and volume of data (d) none of the above. The link utilization is better in packet switching compared to circuit switching (a) True (b) False (c) Neither true nor false (d) None of the above The examples of message switching services are : (a) Conventional telephone communication (b) T.V. broadcasting (c) Cable radio and video (d) Telegraph, E-mail As the packet size becomes shorter, the packet switching becomes more suitable for (a) Data communication (b) Voice and video communication (c) Data, voice and vide communication (d) None of the above

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6. As the packet size becomes longer, the packet switching becomes more suitable for : (a) Data communication (b) Voice and video communication (c) Data, voice and video communication (d) None of the above 7. The cost of a call is calculated based on time duration of the call in : (a) packet switching only (b) ATM switching only (c) Circuit switching only (d) None of the above 8. While delay in message switching may be a few minutes to a few hrs around, the delay in packet switching may be a few seconds to a few minutes around. (a) False (b) True (c) None of the above 9. Datagram service of packet switching is suitable for : (a) High volume of traffic (b) Low volume of traffic (c) None of the above 10. Virtual circuit service of packet switching is suitable for : (a) High volume of traffic (b) Low volume of traffic (c) None of the above. 11. Virtual circuit refers to a (a) dedicated physical link (b) dedicated logical link (c) logical link that may be shared by others, also, during a call session (d) none of the above. 12. The compromise switching technique between the Circuit Switching and the Packet Switching is known as : (a) virtual circuit switching (b) datagram service switching (c) message switching (d) ATM cell switching. 13. ATM cell switching is suitable for : (a) data only (b) voice and video only (c) data and video only (d) all services 14. The cell size of ATM is : (a) 53 bytes (b) 48 bytes (c) 5 bytes (d) none of the above. 15. The header size of ATM cell is (a) 53 bytes (b) 48 bytes (c) 5 bytes (d) none of the above. 16. The information field size of ATM cell is (a) 53 bytes (b) 48 bytes (c) 5 bytes (d) none of the above. 17. The appropriate transport technique for multimedia service is : (a) packet switching (b) message switching (c) ATM cell switching (d) None of the above. 18. The Datagram service and the virtual circuit service are respectively : (a) the connection oriented and the connectionless services (b) the connectionless and the connection oriented services (c) both the connectionless services (d) both the connection oriented services

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19. Probability that a packet is in error (a) increases with increase of packet size (b) decreases with increase of packet size (c) does not depend on the packet size (d) none of the above. 20. As the size of the packet increases, the probability of the retransmission of the packet : (a) remains same (b) decreases (c) increases (d) none of the above. 21. When bit error rate = 1; the probability that the packet in error, irrespective of the packet size , is : (a) zero (b) 0.5 (c) 1 (d) none of the above 22. The probability that the packet is in error becomes zero when : (a) bit error rate = 1 (b) bit error rate = 0.75 (c) bit error rate = 0.5 (d) bit error rate = 0 23. a message transferred through a network of Fig. (1) under message switching takes a transmission time of 2t sec. Message is now broken into two equal size packets, and is proposed to be sent under packet switching. How much transmission time shall the packets take to transfer message? (a) 2t sec (b) 1.5t sec (c) 1t sec (d) none of the above 1

1

Source

Tination

r = link speed in BPS c = message size in bits t = c/r sec (Assume overheads are negligible) Fig. 1 24. If for the problem (23), the overhead bits (h) required for the message is same for each of the packets; the coding efficiency under message switching and packet switching respectively could have been : (a) c/h and c/(h = 2) (b) c/(c + h) and c/[c + (h/2)] (c) c/(c + h) and c/2()/[(c/2) + (h/2)] (d) c/(c + h) and c/(c + 2h). 25. The basic objects of routing are : (a) maximize transmission times, maximize transmission costs and minimize the network throughput (b) maximize transmission times, minimize transmission costs and minimize the network throughput (c) maximize transmission times, transmission costs as well as the network throughput (d) maximize transmission times, and the transmission costs; and maximize the network throughput 26. The routing strategy must avoid : (a) oscillations only (b) loops only (c) either oscillations or loops (d) oscillations and loops

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27. For a network of Fig. (2), oscillation occurs when a packet follows the path as : (a) 1-2-3-4 (b) 1-2-2-2-2-2---------(c) both of (a) and (b) (d) none of the above. 28. For a network of Fig. (2), loop occurs when a packet follows path as : (a) 1-2-3-4 (b) 1-2-2-2-2------(c) both of (a) and (b) (d) none of the above. 29. The Floyd’s algorithm and the Bellman ford algorithm are the two important routing algorithms. They are both shortest path routing. (a) true (b) false (c) neither true nor false (d) none of the above. 1

A

8

D

4

7

6

B

C

3

F

E

5

Fig. 2 30. We can make perfect routing decision (a) true (c) neither true nor false

(b) false (d) none of the above.

Answers 1. 9. 17. 25.

(a) (b) (c) (d)

2. 10. 18. 26.

(b) (a) (b) (d)

3. 11. 19. 27.

(a) (c) (a) (b)

4. 12. 20. 28.

(d) (d) (c) (a)

5. 13. 21. 29.

(b) (d) (c) (a)

6. 14. 22. 30.

(a) (a) (d) (b).

7. (c) 15. (c) 23. (b)

8. (b) 16. (b) 24. (d)

5.15 DOMAIN NAME SERVICE Most human beings are better with names than numbers. It is really difficult for average user to remember and keep track of the IP address of a host in numeric from like an address 204.78.100.70. But it is relatively easy for human beings to remember and keep track of a name like [email protected] or www.ictp.it or [email protected]. On the other hand, computers (hosts/nodes) need numeric decimal dotted IP address rather than textual address. To tackle with this issue, domain name was introduced in the Internet architecture. A domain name is textual name of a host/node mapped to its IP address. TCP/IP protocol suite includes a Domain Name Service (DNS) that provides an address resolution. The address resolution converts an IP address into a domain name or a domain name into an IP address. DNS uses a lookup tables for this purpose. Similar to the two parts, network ID and host ID, of an IP address; the user’s name in textual form is associated with two parts. The one part is the name of individual user and the other part is the name of the organization and/or host. The two parts are separated by a symbol, @. For example in [email protected], “cnaa” is the user’s name and “vsnl.com” is the organizational host name. In this naming there is no host name. Based on the user’s name, the organizational

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server will identify the host of users “cnaa”. On the other hand, in [email protected], the user’s name is “cnaa”, the host is “ca13” and the organizational domain name is “vsnl.net.in” .The organization part of the name may have subparts each separated by period (.).The subparts are used to indicate the name of the organization, the type of the organization and the country to which it belong. The organization part of the user’s name is known as organization domain name. The organization domain name, “vsnl.com” name has two parts: “vsnl” is the name of the organization, the type of the organization is “com” (commercial). In www.icpt.it, the organization domain name is “icpt.it”. The domain name has two parts: “ictp” is the name of the organizational and it belongs to country “it” (Italy).In the [email protected] , there is host domain name. Internet host domain name comprises many parts - a Top level Domain (TLD) such as country domain and others TLDs referring to type of the organization etc (See tables 25 and 26). In this example, the host domain name is “ ca13.vsnl.net.in” in which the name of host is “ca13” and its organizational domain name is “vsnl.net.in”. The name of the organization is “vsnl”, its type is “net” (network provider) and it belongs to “in” (India). The IP address of the host domain name “ca13.vsnl.net.in “is 202.54.9.23. It is a class C address with network ID as 202.54.9 and host ID as23. A TLD may have many sub domains as shown in the Fig. (48). In the Fig. we find “com” has a sub domain of “ent”. In this case, the address for sob domain shall be “vsnl.ent.com”. net – – – – – – gov – – – – – – com – – – – – – edu

Ent

Ent 1

Bus

Ent 2

TLD

Com subdomins

Ent 3

Ent dubdomains

Fig. 48: DNS tree

Table 25: Conventional Domain Name based on type of organization Type of the Organization

Conventional Domain Name

Educational Commercial Government Network Provider Non profit Organization

edu com gov net org

Table 26: Conventional Country Domain Name Name of the country Australia Canada Japan USA Finland Poland

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Name of the country

au ca jp us fi po

India in Italy Sweden UK Chile Hungary

Conventional domain name it Se uk ch hu

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The Internet Assigned Number Authority (INNA) assigns and coordinates the IP addresses. TLD naming authority for a specific region assigns the domain names. Presently the International Ad Hoc Committee (IADHC) is proposing a global TLD naming convention (Table 27). For proper working of DNS, a domain name server at the user or client, and a domain name server at one or more hosts are used. There may be more than one server in a domain of large networks. This helps in distribution of traffic. In a network there is always a root domain server that provides a master table of names and IP addresses, and other servers if exit they are called secondary servers. The root server regularly distributes and update the look up tables of secondary servers. The other way of controlling traffic is the use of “forwarder”. DNS software has option to set a “forwarder” for an IP address. For example when a particular web is being accessed by many people, in order to avoid traffic, “forwarder” may be used to point to DNS of other network from where to find an IP address and domain name resolution. The look up tables for domain name resolution are maintained by network administrations. Let us look at how a domain name server retrieves IP address of a given domain name. Let us say the given client or user name is [email protected]. Here the organizational domain name is “vsnl.com”. So first a root server will be contacted that would provide a list of server of different types like edu, com, net, and gov etc. Second, one of the com servers will be contacted to search for “vsnl”. If com servers locates the “vsnl”. If com servers locates the “vsnl”, its IP address is returned from the look up table. Table 27: Proposed global TLD names Type of the organization

Proposed global domain name

Arts related organization General firms Recreational and entertainment organizational Products selling organization Individual unique organization Information services related organization Organizations that offer web activities

arts firm rec store nom info web

5.16 VOICE OVER INTERNET OR INTERNET TELEPHONY Internet has established itself as the most important and the single most tool of global information age. It was developed for transporting packet data, a non real-time service. But today, Internet telephony has emerged as an important technology. Internet telephony is supposed to carry real time and jitter-free voice over Internet. Active and hectic researches are being carried over the subject of VoIP (Voice over Internet Protocol). Generally speaking the use of Internet for all real time services, like voice, video, and multimedia is being explored. Table (28) [12] shows a growth estimation of VoIP traffic. There are several motivations [11] for transmitting voice over IP. These are: (1) long distance calls at low cost and may be of low quality, (2) cheaper two in one service, (3) Use of PC as a true multimedia terminal, (4) one connection for all services, (5) local exchanges can support telephone with Internet as backbone and without high investment in expensive back bone infrastructure, and (6) use of packetized voice allows voice compression that in turn decreases transmission time and cost. Earlier, telecommunication traffic or telephony connections outnumbered the data traffic. The future

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is to see the explosion of data traffic. When there will be crossover is debatable, but sooner or later data traffic will dominant the telecommunication traffic. “Consequently, now should be the time for datacom to act as a carrier for telecom.” But Internet, as such cannot be used to carry real time service as it was designed to carry data and as the characteristics of real time services like voice and video are different from data. Table 29 shows the different characteristics and different requirements of voice, video and data The need to deploy Internet for the real time services like voice and video, have lead to redesign some features of Internet. The important two features related to this emerging issue are: (i) redesign of IP datagram format, and (ii) to use RTP (Real Time data transfer Protocol) and IP for carrying voice over conventional IP datagram and Internet. It is believed that with deployment of Ipv6, VoIP will be reached. Table 28: Projected growth of IP telephone (A) As per [12] Voice IP Traffic 1998 1999 2004 (expected)

310 million minutes 2.7 billion minutes 135 billion minutes

(B) As per [16] Year

Average unit (millions per year)

2000 2002 2004 2006

3987.2 22,386.2 167,896.2 587,636.9

Unit growth rate (%) 256 162 114 75

Yearly revenues (millions)

Yearly revenue growth rate (%)

388.75 1511.07 8814.55 22036.38

209 136 88 46

Table 29: Characteristics of different services Voice

LAN data

Transactional Data

Video

Predictability

Constant/On-Off

Bursty

Highly bursty

Constant/Bursty

Bandwidth/ Bit rate

Very Low to Low

Medium to High

Low to Medium

High

Delay/Jitter

Sensitive

Tolerant

Tolerant

Sensitive

Loss

Sensitive/ No recovery

Sensitive but can recover

Sensitive but can recover

Very sensitive/ No recovery

Error/Integrity

Can tolerate

Can not tolerate

Can not tolerate

May tolerate

Technical Problems of Voice Packet Transmission Over Internet PSTN (Public Switched Telephone Network) based on circuit switching provides voice service with guaranteed quality of service. This is not the case in case of voice service provided by Internet that acts on packet switching. Many technical challenges the voice packet faces while

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in transition over packet switching network like Internet. These include packet loss, packet transfer delay and jittering delay. Voice communication is involved with human interaction. As such, a few losses of the voice packets could be tolerated due to human intelligence and perception involved in recovery. But too much loss of the voice packets may seriously degrade the voice quality. Moreover, PSTN is a reliable voice service provider whereas Internet is not, as because Internet is datagram based. Table 30: End to end voice packet latency delay Delay source

Typical value (end to end or Phone to Phone) in ms

Recording

10-40

Encoding/Decoding(CODEC)

Each 5-10/Both together 10-20

Compression/Decompression (SPEECH)

Each 5-10 / Both together 10-20

Internet Delivery

70-120

Jitter buffer

50-200

Average

150-400

Delay is the more serious issue for real time interactive services like voice. By delay it is meant that the time difference between the time the sender releases the packet to the network and the time at which the receiver receives the packet from the network. Delay refers to : (1) total transfer delay of a packet that includes coding/decoding delay, propagation delay, transmission delay, node processing and queue delay, switching and routing delay; and (2) jittering delay that refers to the phase delay between two successive packets. Typical delay from different sources are as in Table (30)[12]. If the total delay exceeds a certain value, customers may get irritated to the service. A statistic says that a delay up to 80 msec between the caller and callee is acceptable but beyond it causes irritations to the users. The total delay is a variable quantity, and it varies from packet to packet. The jittering delay is very serious issue. If the phase lags between the voice packets at the source and destination varies, the service quality degrades. The phase lag between packets differs from the source end to the destination end because the total transfer delay varies from packet to packet. Due to jittering problem, a sending voice “I shall go home” may be received as “I shall go home”. Compared to the transmitter, the phase delay between “i” and “shall” has increased and that between “shall” and “go” has reduced to zero at the receiver. While the total delay could be limited by increasing the bit rate capacities of the link and by adopting efficient routing technique among others, the jittering effect can not be solved so simply. There are several techniques to reduce the affect of the jittering problem. One such technique is known as accelerating and de accelerating. In fact the jittering problem is due (Di+1 – Di) which is finite and a variable. Here, Di+1 and Di are both variable quantities and represent respectively the total transfer delay of (i + 1)th packet and ith packet. To avoid the jittering effect, it is required that Di + 1 – Di = 0. In the accelerating and de accelerating technique, at the receiver end a variable delay (say Wi for ith packet) is caused to each packet such that Di + Wi = K, a constant for all packets (i.e. for i = 0,1,2,3…..) before delivery of the packets to the terminal equipment for play back. By the process, the variable delay caused by the network between two successive packets is made zero as (Di + 1 + Wi + 1) – (Di + Wi) = 0. This ensures that the phase delay between packets at the transmitter remains same at the receiver. The scheme is illustrated in the Table (31). As illustrated in the table, the success of the technique depends on the choice of K.

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Table 31: Illustrating of accelerating and de accelerating technique to cope up with the problems of jittering Instant at which a packet is released at the transmitter (xi) in ms th.

Variable delay with which the packet reaches the receiving node in ms (Di)

Variable delay (Wi) caused at the receiving buffer (100 – Di) in ms (K has been chosen as 100 ms)

Delay with which the packet is delivered to the terminal device (xi + 100 ms)

Packet-1

0

80

20

100

Packet-2

10

70

30

110

Packet-3

15

85

15

115

Packet-4

25

100

0

125

Packet-5

30

110

– 10

130

(packet-4 is the marginal case. Packet-5 is the failed case. Both could have been avoided had the constant K been chosen more than 110 ms in this case. So the success of the technique depends on the choice of fixing K) VoIP is going to be a dominant service issue of IP. VoIP has several motivations as we discussed earlier. PSTN supports only toll-quality sound (4 KHz sound), and not suitable for high-fidelity sound. VoIP can support higher grades of sound. This will be another major driving factor for VoIP. But there are several issues that need to be resolved before VoIP is used. Standards are still not finalized, although H.323 of ITU is being projected as a possible standard. H.323 may be under new version 2 be used for interoperability between different service networks like PSTN and Internet to support voice. The standard H.323 is for multimedia or videoconferencing. The audio G.7xx standard of H.323 may be many based on choice of xs. The choice of xs will define the intelligibility of the voice service provided. IPv6 for Real Time Services The conventional packet switching is not appropriate to carry real time services. There are many reasons for this. For example HDLC or SDLC packets are variable in size. To synchronize and identify a packet, flags are required to be located. To avoid occurrence of flag byte in the payload, stuffing and de stuffing are done. These cause huge node processing delay, and hence packet transfer delay. ATM was proposed as the replacement of packet switching to support real time services. The problems of conventional packet switching were solved in ATM by making ATM packet, called cell simpler. The simplicity in ATM is in two respects: (1) shorter cell and (2) fixed size cell. This philosophy was extended to design Ipv6 datagram to replace Ipv4 datagram so that IP can carry real time services. IPv6 has a simple and basically fixed header format. The overhead bits of Ipv6 are less than that of Ipv4. The overhead bits in Ipv4 is 12 bytes in the header format of 20 bytes (8 bytes are for address), whereas the overhead bits in Ipv6 is 8 bytes in the header format of 40 bytes (32 bytes are for address). IPv 6 proposes to provide QoS (Quality of Service) service support to real time services like voice and video. The flow level and priority in the header of Ipv6 facilitate the support of real time data. Ipv6 has an efficient header format compared to Ipv4.

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Several major modifications of Internet have been projected in literatures. These if are implemented successfully may replace the today’s Internet, and certainly these are long term ventures. But techno-economically, the existing Internet with Ipv6 and VoIP may go a long run, and may be the path for migration to new generation of Internet by cohesion and coexistence rather than replacement Two important proposals for development of the Internet are : (i) under sea super speed Internet and (ii) wireless Internet. A proposal for a global optical-fiber under sea cable network called Project Oxygen has significant industry support and financial backing. This project is called “the best of bandwidth on demand” project as per the company release. Experts say “ Project Oxygen is the most ambitious communication project in the 20th century…..The Internet and video transmission are the major drivers for the expansion…..a global optical fiber network could erase the boundaries between Internet and the traditional communications, and shift the profit model from voice service to data and video." ”Construction of the under sea network began in September’98. In the first phase, the cable shall be stretched over 158000 Km in 74 countries with three major network management centers in USA, Spain and Singapore. The major transatlantic and transpacific links are likely to be operational by 2000. Phase two shall start in 2002 and cover the whole of the world. The speed of cable is projected at 1920 GBPS with minimum capacity of 640 GBPS. It is reported that with under sea Internet, a video-based Internet shall come with over 10,000 video channels. A revolution in wireless has reached. In 10 years time, we can have mostly wireless devices, equipments and computers. These need to be connected to the Internet. For this, a situation may require to have all wireless networks including wireless Internet. BOX 16

Physical Layer Interface Introduction Within a span of few years, the computer through communication has invaded all spheres of life. Any computer communication system needs three basic hardware elements, namely 1. A physical medium such as twisted wire pair, co-axial cable and fibre optics. 2. A control system often known as CC (communication controller). 3. An electromechanical often known as CI (communication interface). The CI is to connect data terminating equipment known as DTE with data circuit-terminating equipment known as DCE as shown in Fig. (1). Typically, DTEs are the host computer, printer, plotter, terminal etc. and DCEs are modem, nodes of networks etc. But under different circumstances, any one of these may either be a DTE or a DCE. C

C

C I

C Physical line

DTE

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Standard There are different standards of different organizations for CI. A comparative study of some of these standards has been given in Table (1). Till now most successful, acceptable, useable and common interface is RS–232–C interface as because in realistic terms of cost and technical constraints it is effective where relatively average performance is acceptable by the users. This article, is hence set to fully discuss the RS –232 –C serial interface. Historically ‘the proliferation of non-computer manufacturing companies building terminals’ motivated the EIA of USA to publish a voluntary standard for the general convenience of computer and electronic equipment manufacturers. As the EIA has no legal right to publish such standard, its committee TR30 responsible for standardization, calls it Recommended Standard, hence the prefix ‘RS’ is there in the name of interface. The latest third (C) revision designation it as RS–232–C; 232 being the number of the standard. The features of RS-232-C interface are 1. 2. 3. 4.

It is a serial interface. It can handle both asynchronous and synchronous operation. It can be used in unbalanced system. It treats binary ‘YES’ also known as mark and binary ‘NO’ also known as space by a voltage in the range of –3 volts to –25 volts and in the range of +3 volts to +25 volts respectively. 5. It transmits/receives data bits (number of bits / characters may be 5, 6, 7, or 8) enclosed between a start bit and a stop bit (S) (in rare case 11/02 bits are used). The voltage transition to the space is defined as start bit, whereas voltage transition to the mark is defined as stop bit. Parity may be used for error detection. Parity may be even, odd or none. 6. It may use nineteen signals for transmission/reception/handshaking and two signals for grounds (Fig. 2). 7. The industry has generally accepted a D-shaped 25 pin connector. (Fig. 2) for RS-232-C standard. Such connector has two forms: male and female. Male form also known as plug end has 25 pins. The Rs-232-C side which is on DTE side uses the male connector. The female form also known as socket end has 25 receptacles. The RS-232-C side that is on DCE side uses the female connector. But there is exception, which is explained in next section. Application of RS-232-C The major applications of this CI are in the following cases: (a) In local use (Fig. 3a) 1. In connecting a computer with terminals (Keyboard and Screen and Local Printer) 2. In connecting a computer with specialized device such as monitoring system etc. (Fig. 3b) (b) In remote use (Fig. 4) 1. In connecting a computer with modem. 2. In connecting a computer with a node etc. While RS-232-C connector is being used; users may have to face basically two problems. Firstly, there is no hard and fast rule that a device will always be a DTE or DCE. In some

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occasions a DTE may be used as a DCE and vice-versa. Thus there may be needs for DTE-DTE or DCE-DCE connections. However, straightway connections in these cases is not possible as a male connector can’t be connected within a male connector and so on. The solution lies in use of a special adapter known as null modem (Fig. 5). This adapter is the special cabling scheme that interchanges some of the wires to make a DCE look like a DTE and vice-versa. Secondly, although the 25 pin connector is most common, Yet there are different forms of small connector. Such as 3 pin economic connector, 9 pin full duplex connector (Fig. 6) etc. In order to connect say any DTE with 9 pin male connector with any DCE with 25 pin female connector; again some special adapter cable is to be used. 12

13

14

15

Pin

EIA Ref.

Description

1

AA

Protective ground (PG)

2

BA

Transmitted data (TD)

3

BB

Received data (RD)

4

CA

Request to send (RTS)

5

CB

Clear to send (CTS)

6

CC

Data set ready (DSR)

7

AB

Signal ground (SG)

8

CF

Data carrier detect (DCD)

9

—

Positive DC test voltage (PDS)

10

—

Negative DC test voltage (NEG)

11

—

(unassigned)

12

SCF

Secondary DCD (SDCD)

13

SCB

Secondary CTS (SCTS)

14

STD

Secondary TD (STD)

15

TB

Transmit clock (TC)

16

SBB

Secondary RD (SRD)

17

DD

Received clock

18

—

(unassigned)

19

SCA

Secondary RTS (SRTS)

20

CD

Data terminal ready (DTR)

21

CG

Signal quality detect (SQ)

22

CE

Ring (calling) indication (RI)

23

CH/CI

Data signaling rate selector

24

DA

Serial clock transmit (external) (SCTE)

25

—

Busy (BUSY) Fig. 2

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IC s for RS-232-C Interface In addition to Rs-232-C electromechanical interface, a typical CI uses other two ICs (Fig. 7): formatting & protocol IC and physical interface IC (line drivers and receivers). However, the discussion of formatting & protocol IC is beyond the scope of the article. Physical interface IC is required for RS-232-C port. The voltage levels of YES and NO state of RS-232-C standard as discussed previously are not compatible with the voltage level of many digital electronic circuit and TTL gates etc. Physical interface IC performs the conversion of these voltage levels removing the problem of incompatibility. Typical physical interface ICs are the MC 1488 line - driver and MC 1489 line - receiver pair. Equivalent CCITT Standards The Rs-232-C is equivalent to CCITT V.24 in description of interchange circuit and to CCITT V28 in electrical signal characteristics. The major difference between Rs-232-C and V.24 recommendations is that the identification of each interchange circuit is made by a two or there alphabetic character reference by Rs-232-C and the same is done by three numeric character by V.24. Limitations and Remedies The important limitations of Rs-232-C standard are : 1. The coverage of only 50 ft is not at all enough even for many small plants and/or large establishments. 2. Maximum data rate of 20 K-baud is not sufficient for many cases as large amount of data could not be transmitted quickly. 3. Multidrop communication is not possible except the point-to-point communication between a transmitter and a receiver. 4. The system is susceptible to noise as it operates only under single ended connection. 5. It has the problem of voltage incompatibility with TTL gates as discussed previously, and 6. It left some unassigned connections which provides a free hand to vendors to use these pins to perform special function to make their products more attractive to the consumers. Thus different pieces of Rs-232-C standard equipment of different vendors may be incompatible in operation. Thus this standard may fail to provide standardness. This means that though Rs-232-C is good for many applications or just marginal for many situations. In order to overcome the limitations of Rs-232-C, some expansion on Rs-232-C are made by EIA in order to design new standards such as RS-423, RS-422 and RS-485. These new standards change the details of electrical interface such as voltage, signal types etc. to long distance communication with lower, higher baud-rate etc., but they do not change the basic data transfer technique i.e. start/stop nature of RS-232-C standard. The RS-422, RS-423 and RS-485 solve many problems of RS-232-C to a greater extent (Table 1). All these standard use special ICs for removal of voltage incompatibilities. However, the RS-422 and RS-485 have the better control over noise as they use differential signal. The major enhancement done by the RS-485 is to support Multidrop operation upto 32 transmitters and 32 receivers. It has the highest operational performance and flexibility over other three standards of the above noted general purpose EIA standards. The two distinct limitation of the above stated group of EIA standards are: (1) none of them can use the maximum bit rate to be used upto maximum distance (Fig. 8) and (2) they

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can’t use current loop for binary representation in order to meet the situations where voltage representations become complex. Conclusion In order to provide improved performance to systems, the designers are in increasing needs of new CI standard. The general-purpose (a 37 pin connector with an additional 9 pin connector for a secondary communication) EIA-RS-449 standard which covers two separate specifications : an unbalanced electrical interface (Rs-423-A) and a balanced electrical interface (RS-422-A) is a very good competitor of RS-232-C. But as at present, the huge number of RS-232-C systems are already in operation and are being manufactured by large number of companies; it will take time to replace RS-232-C.

C

C

I

I

DTE (Computer)

DCE (Printer)

Fig. 3(a)

Tranducer

Meter + A/D converter

C

C

I

I

Computer

Fig. 3(b)

C

C

I

I

DTE (Computer)

N long-haul link DCE (Modem)

DCE (Modem)

C

C

I

I

DTE (Terminal)

Fig. 4 2

2

2

2

3

3

3

3

Fig. 5a: Simplest Null Modem

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2

4

2

3

3

5

3

4

2

6

6

5

3

7

7

6

6

8

8

7

7

20

20

8

8

22

22

20

20

(b) Loop Back Null Modem

(c) Double-Cross Null Modem Fig. 5 02, TD 3, RD 4, RTS

Connector

5, CTS 6, DSR 02, TD

7, SG

3, RD

8, DCD

7, SG

20, DTR 22, RF

(a) 3-pin

(b) 9-pin

Fig. 6 Computer or terminal etc.

8251 UART

TD

RD

RD

TD

1488 1489 Link

Formatting & Protocol IC (a part Of CC)

Physical Interface IC

RS-232-C Line Driver / Receiver Fig. 7

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8251

1488

UART

Computer or terminal etc.

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Maximum band

Bit rate

Distance

Maximum distance

Fig. 8

BOX 17

Transmission Media Option A communication system is made of a transmitter, a medium and a receiver. The medium actually carries the signal. The transmission medium is the physical path between transmitter and receiver in a communication system. Cables, optical fibers and even air can serve as excellent medium for sending data and signals from one point to another. In the electrical and optical communication, electrical and optical signals respectively are used as the carriers. They use two kinds of transmission media (Fig. 1): Closed and Open. Closed medium transmits the bounded (guided) wave whereas open medium transmits the unbounded (unguided) wave. Both the nature of the signal to be carried and the medium to be used determine the characteristics and the quality of the transmission. However, in case of closed medium, the nature of the medium is more important than the nature of the signal in determining the quality and the characteristics of transmission, while in case of open medium the nature of the signal is more important than the nature of the medium. Accordingly, in different applications, the designers have very little options to choose a particular medium, based on technical grounds. For example, for communication in remote locations and hill areas, air and vacuum are the only practical choice. From data transmission point of view, the characteristics of open and closed media are given in Table (1).

Closed Medium The examples of this class are twisted-pair wire, co-axial cables and optical fibers. The twistedpair wire has two insulated copper wires arranged in a regular continuous spiral pattern with 2 to 6 inches per 360 degrees twist. Through spiral wrapping, the susceptibility to external electrical noise is minimized, because through such warping the noise appears nearly identical on two wires of pairs and the continuous circuitry can suppress most of the noise. Twisting is useful in minimizing the interference created when adjacent pairs are combined in multi-pair cables. The diameters of the wires range from 0.016 to 0.051 inches (26 to 16 gauges). Common numbers are 22, 24 and 26 gauges. However, 16 and 19 gauges are also used. When wires are wounded or twisted around each other inside a cable, we have twisted pair cables made of twisted pairs. A number of twisted pairs are packed in a cable. The cable size varies from 3 mm

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diameter to about 88 mm. A cable can carry from a few to several thousands individual pairs. The twisted-pair wire is the simplest and low cost medium. It is easy to install and good for low-performance applications. Twisted wire pair is one of the original wire types used in telephone. The problems of twisted pairs are: (1) the signal distortion caused by the skin effect, (2) the interference caused by crossed talk due to other signals induced from wire pair, (3) signal attention over length and (4) signed distortion at higher data rate. The skin effect refers to the phenomenon that the high frequency signal are carried only by the outer surface of cable and this makes the high frequency signal to be contaminated with the electrical and the magnetic fields from adjacent wires. To minimize interference due to cross talk two techniques are used: (1) screening the cable with a light metal foil usually wounded into the cable sheath; this produces what is called Shielded Twisted Pair (STP), and (2) by isolating receive pair from transmit pair. Different standardized UTP (Unshielded Twisted Pair) or STP cable types are given in Table (2). The UTP/STP cables are classified into different categories based on quality. As per EIA (Electronics Industries Association of USA), there are 6 categories. For data communication; categories 1 and 2 are not used. Categories 3, 4 and 5 are known as voice-grade but are also used for data. The transmission speed of category 3 is up to 16 MHz. The category 4 goes up to 20 MHz. Category 5 is known as data-grade, and supports upto 100 MHz transmission. Most data cables are of category 5. But with proposition of gigabit Ethernet and ATM-to-desk, the transmission in the region of 300 MHz to 600 MHz is on demand. To meet with, a new cabling known as category 6 is being experimented. Most installed category 5 cable does carry data at a speed well bellow its rating of 100 MHz. Yet there is demand for high rate cable. Category 6 under experimental stage is proposed at least to double the rate of category 5. To do so, and to optimize the performance of category 6; it is proposed to ensure individual screening to each of four pairs. This will minimize cross-talk and inference, but will increase the physical size, weight and cost of the cable. UTP cables are connected to the network devices via snap-in plug connector (Fig. 2). Connectors may be either male or female. Male connector is RJ45 with eight conductors, one for each pair of four twisted pairs. Transmission media

Closed (For bounded waves)

Twisted pair wire

Optical fibre

Open (For bounded waves)

Air

Co-axial cable

Fig. 1: Classification of transmission media.

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Vacuum

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The co-axial cable, like the twisted-pair wire, has two wires but constructed differently to offer substantially larger bandwidth with high immunity to electrical interference and a low error probability. It consists of a single inner wire (solid or stranded), centered in a hollow outer cylindrical conductor (solid or stranded), which acts as second wire of the pair. The inner conductor is held in the center of the outer conductor by either a solid dielectric material or regularly spaced insulating rings; thereby insulating the outer conductor from inner conductor. A jacket or shield covers the outer conductor. The co-axial cable diameter ranges from 0.15 to 0.75 inch, and the cable are identified by RG numbers. The common numbers are RG-58, RG-8, RG-11, RG-9 and RG-174. The followings are a few common use: RG-8, RG-9, RG-11 -used in the dc. RG-56-- used in the Ethernet, RG-75-- used in TV. Although the cables with larger diameter have wider bandwidth and lesser attenuation, they are costly, bulkier, rugged and harder to install. The co-axial cable is more expensive than twisted pair wire as well as more difficult to install because of its diameter and mechanical stiffness. However, its high bandwidth and high data rate are attractions for may data communication systems. Its high noise immunity makes it suitable for long distance/haul communication. Coaxial cables are widely used in telephone, under sea communication and in cable TV. The optical fibers are designed by using different types of glass and plastics. A hair-thin stand, or fiber or glass or plastic is used as a light pipe. The light that enters one end of the pipe, stays in the pipe and travel to the other end by total internal reflection. Typically, an optical fiber is between 0.125 and 0.5 mm in diameter, not much thicker than a human hair. A fiber pipe comprises an inner core which carries the light signals; and a cladding which reflects and traps the light inside the core. The core has a refractive index about 1% larger than the cladding to ensure total internal reflection. In fiber optics, light is the electromagnetic wave, rather than the conventional electrical wave. There are two most common types of fibers: stepindex and graded-index. It offers wider bandwidth, higher data rate and offer considerable long haul communication. With fiber optics, it is possible to handle data at gigabits per second (1 Gbit = 1000 Mbits). Silica fiber has an unprecedented bandwidth of 25,000 GHz or 25 THz (Tera hertz). There are numerous advantages of fiber: extremely wide bandwidth, light-weight and smaller size cables, lack of cross-talk and interference, greater security, longer life span, greater safety, use of common natural sources for medium etc. The major problem of optical fiber is the termination of cable. Flexibility in terms of addition or deletion of users is less. There are different types of fibers like step index fiber and graded index fiber based on physical design or mono-mode fiber and multi-mode fibers based on the ability to support modes of propagation.

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Wireline

Wireless Cost

Time

Fig. 3: Cost Comparison of wire line and wireless communication.

Table 1: Characteristics of closed and open transmission media from the data transmission point of view Type of medium

Medium

Closed

Open

Bandwidth

Data rate

Repeater

Uses spacing

Twistedpair wire

250 kHz

1MBPS

2-10km

Analogue and digital communication, telephone, low-cost and lowperformance LANs.

Coaxial cable

350 kHz

500MBPS

1-10 km

Long distance telephones, cable TV, CATV, and LANs

Optical fiber

1GHz

1GBPS

Air and vacuum

Moderate to very high

Moderate to very high

1-100 km

variable

Long distance and high data rate point-to-point communication. Satellite, cellular, terrestrial, and international communication

Open Medium The air (the earth's atmosphere) and the vacuum of space are very useful, low cost and attractive means of transmitting electromagnetic waves. They are used to transmit bounded waves known as radio waves, microwaves and infrared optical waves. However, the entire electromagnetic spectrum ranging from the extremely low frequency to the extremely high frequency can be used in both air and vacuum as the bandwidth of both the media is very wide. The electromagnetic waves can travel more distance in vacuum than in air, because the various molecules of air like oxygen, carbon dioxide, nitrogen and others absorb the electromagnetic energy. Cost (installation, system and operation & maintenance etc) wise the comparison of wire line (UTP /STP/Coax/Fibre) and wireless (air/vacuum) is shown in Fig. (3). In Fig. (4), the frequency range of different media is shown.

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Twisted wire pair Coaxial cable Radio communication 3 KHz 100 KHz

100 KHz 5 KHz

500 MHz

300 GHz

Satellite communication C band = 4/6 GHz Ku band = 11/14 GHz Ka band = 17/31 GHz

Fig. 4

Radio, microwave and satellite channels use open space for carrying electromagnetic signals. The attenuation is relatively lower compared to wire line links. Hence either repeater is not used or used at a longer interval of links. However, we refer electromagnetic waves of frequency in the rang of 30 MHz to 1 GHz (covering VHF and part of UHF) as the radio waves, while microwaves cover the range of about 1 GHz to 40 GHz, which include a part of UHF band and all of SHF bands. The satellite communication uses microwave (4/6 GHz band and also UHF and VHF band) and air and vacuum as the media. FM ratio and also UHF and VHF television use air medium with radio waves. Mobile radio telephone also uses radio waves (for example: 800 to 900 MHz) in air. Infrared optical wave is yet to find a practical use in air or vacuum. Below 30 MHz the signal propagation is done by ionospheric reflection. Microwave link , usually above 100 MHz uses line of sight propagation. Satellites use microwave frequencies and a satellite as a reflector or/and repeater. From data communication and network point of view, the selection of media depends on several factors e.g. type of network, transmission distance, transmission speed, cost etc. The transmission distance or rang refers to the distance over which the medium can be used. Transmission rate measures the data rate that media can support. Error rates refer to the BER (Bit Error Rate) of data link. If the medium is resistant to the unauthorized users to access and /or modify the data, the medium is more secured. Cost refers to the installation cost of medium. Table (3) gives a media summary for application to different networks like LAN (Local Area Network), WAN (wide Area Network) etc. Table 2(a): Different categories of UTP/STP cable Type UTP or STP cable type

Rate/Possible application

Category 1 (Cat. 1) and Category 2 (Cat. 2)

voice and low data

Category 3 (Cat.3)

specification allows use up to 16 MHz typically allowing use for combined voice and Ethernet or 4 Mbit /s token ring LANs

Category 4 (Cat.4)

specification allows use up to 20 MHz allowing for 16 Mbit/s token ring LAN use as well as voice.

Category 5(Cat.5)

specification allows use up to 100 MHz, allowing use for voice or data up to 100 Mbit /s

Category 6 (Cat.6)

under proposal stage.

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(b) Use of UTP /STP Name

Type

Data Rate (Mbps)

Distance (Meters)

Often Used by

Category 1*

UTP

1

90

Modern

Category 2

UTP

4

90

Token Ring-4

Category 3

UTP/STP

10

100

10 Base T Ethernet

Category 4

UTP/STP

16

100

Token Ring-16

100

200

100 Base T Ethernet

10

185

10 Base 2 Ethernet 10

Category 5

UTP/STP

RG-58

Coax

RG-8

Coax

10

500

X3T9.5

Fiber

100

2000

Base 5 Ethernet

FDDI

(c) Transmission media performance Medium

Cost

Speed

UTP STP Coax Optical fiber Radio Microwave Satellite Cellular

Low Moderate Moderate High Moderate High High High

1-100 Mbps 1-150 Mbps 1Mbps-1Gbps 10 Mbps-2 Gbps 1-10 Mbps 1-Mbps-10 Gbps 1 Mbps-10 Gbps 9.6-19.2 Kbps

Attenuation High High Moderate Low Low-high Variable Variable Low

Table 3: Comparison of media for data/computer networks

Network Type Cost Transmission Distance or rang Security Error rates Transmission Speed

Network Type Cost Transmission Distance or rang Security Error rates Transmission Speed

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Twisted Pair

Coaxial Cable

Fiber Optics

LAN Low Short Good Low Low-High

LAN Moderate Short Good Low Low-High

any High Moderate Very Good Very Low High-VeryHigh

Infrared

Microwave

Satellite

LAN Low Moderate Poor Moderate Low

WAN Moderate Long Poor Low-Moderate Moderate

WAN Moderate Long Poor Low-Moderate Moderate

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Table 4: Media option for sources of various services Telephony

Internet

Broadcast

Download

Copper wire or UTP /STP

YES

YES

NO

YES

Coaxial cable

YES

YES

YES

NO

Fiber

YES

YES

YES

YES

Radio communication

YES

YES

NO

YES

Terrestrial communication

NO

NO

YES

YES

Satellite communication

NO

NO

YES

YES

A matrix of better media options for carrying different services is shown in Table (4): telephony for point to point real time voice communication, Internet for data communication for both OFF and ON line, Broadcast for voice, video and data; and down load for data.

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1. INTRODUCTION The communication service has three basic types: Data Service, Video Service and Voice Service. The requirements of data service are different from that of the other two services, as because the characteristics of data are different from those of the voice and video. Table (1) shows the different characteristics of voice, video and data. Data is known as “BAD IT”[1], meaning Burst, Asymmetric, Delicate and Insensitive to Time. Burst refers to the arrival of data in bursts, meaning for a long time there may not be any data and suddenly there may arise a bulk of data. Asymmetric refers to the more flow of data in one direction than that in the other direction. Execution of one short command at a source may cause the transfer of a very large file from a sink to a source. One example is the transfer of the bank accounts from a branch bank to the head office of the bank, on requisition from the head office. Delicate feature is serious in data service due to handling of data services mostly by machines rather than men. This is not the case for voice and video services where human perceptions are involved. If a data ‘11001111’ transmitted from a source machine is received as ‘01001111’ in a sink machine, the error will be there in the receiving machine until and unless it is corrected by some means. This is what is the delicacy in data communication. And exactly for this, data communication needs a powerful data correction technique for reliable and faithful service. Delicacy feature is not a so serious issue in voice and video communication and this is due to human perception involved in these services. For example, if a person over telephone hears from the other side “I shall go home toomorrow”, he can easily perceive that in place of “toomorrow” what is heard, it is actually to be heard as “tomorrow.” Insensitivity of the data to time means that the real time communication is not essential for data service unlike voice. Data services can tolerate delay but not error. Table 1: Comparison of characteristics of different communication services Service type

Burst

Asymmetric

Delicate

Time sensitive

required bit rate

Data

yes

yes

yes

no

Variable Bit Rate (VBR)

Voice

no

no

no

yes

Continuous Bit Rate (CBR)

Video

no

moderate

no

usually

Continuous Bit Rate(CBR)

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The issue of delicate feature of data is related to the “accuracy” or “reliability” of data communication. The accuracy of data communication is maintained by the basic two techniques: FEC (Forward Error Correction) technique and BEC (Backward Error Correction) technique [1]. The techniques use respectively error correction codes (ECC) and error detection codes (EDC). ECC and EDC are made of redundant check bits appended with the actual informative data bits. Redundant check bits are used for error correction and detection. “Redundancy increases reliability” is the philosophy behind any ECC and EDC. For example, if we want to send two data, x and y to our friend, we can send two redundant data (x + y) and (x – y) to our friend, in addition to x and y. Our friend can compute x and y from received (x + y) and (x – y), and then compare these computed x and y respectively with received x and y to examine whether there is any mismatch and, hence, error. If data size is n bits, any ECC or EDC may be of size m(m > n) bits where number of check bits in the code is (m – n); and the code is called (m, n) code. The use of FEC and that of the ECC and the EDC was reported by Claude E Shannon as early as 1948 [50]. Thereafter a number of codes were introduced: (a) RW Hamming introduced one-bit error correcting codes in 1950. (7, 4) code and (13,8) code are the examples of one-bit ECC. (b) P Elias developed convolution codes in 1955. (c) In 1959 RC Bose and DK Chaudhuri proposed multiple error correcting codes. These are very powerful codes and known as generalized Hamming codes. A Hocquenghem independently designed the codes proposed by Bose and Chaudhuri. That is why these codes are known as BCH codes. (d) In 1960 IS Reed and G Solomon designed a powerful block codes particularly for burst errors. The codes are known as Reed Solomon codes. (e) In 1960 GD Fornery introduced the concept of concatenated codes. (f) In 1967 AJ Viterbi introduced an important convolution code known as Viterbi code Turbo code. (g) Turbo code, Low Density Parity Code[51], combined Turbo Code[52] and Punctured Turbo Code [53] are other important codes. A code tree will look like as below: ECC/EDC

Trellis Codes

Linear Codes

Block Codes/Algebraic Codes

Nonlinear Codes

(Convolution Codes) Linear

Cyclic (CRC codes)

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Noncyclic (BCH/Hamming codes)

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ECCs are used in FEC[39-43]. EDCs are used in BEC[44-46]. More the check bits in a code, more powerful the code becomes both for error correction and detection. But with more the check bits, more the bandwidth is required for the transmission of the code. For a (m, n) code, the required additional bandwidth will be [{(m – n)X Original Bandwidth}/n]. The trade off between increased bandwidth and capability of the code acts behind the selection of a code for a particular application. The trade-off is shown in Table (2): Table 2: A brief comparison of a few EDCs and ECCs Code Type

Code Name

Detection/Correction capability of the code

% of additional bandwidth requirement

EDC

(5,4)–one bit parity check

Basically one bit error detection–but in general odd numbers error detection

25%

EDC

CRC-32 (4 bytes check bits) as used in IEEE 802.3 LAN-maximum packet size 1526 bytes and minimum packet size 72 bytes.

Single bit error detection, also double bit error detection on a large scale, Most burst error detection etc.

0.26% with maximum packet size. 5.55% with minimum packet size.

ECC

(7, 4) Hamming Code

One bit Error correction

75%

ECC

(13, 8) Hamming Code

One bit Error Correction

62.5%

ECC

(23, 12) Golay Code

Three bit Error Correction

92%

ECC always have more check bits than EDC and hence requires more bandwidth. Code capability and complexity in system design are the other parameters for selection of a code for particular applications. Transmission errors are typically two types: Random and Burst. Error is called random, if the bits in error are randomly distributed over the code. Burst error occurs when bits in error are clustered together over the code. For a transmitted byte 01010101, the examples of random error and burst error may be as below (underlined bits are in error): 01110111—random error—errors are distributed and in second and sixth bit locations, and 01101101—burst error—errors are clustered on fourth, fifth and sixth bit locations. CRC is the code that can detect both the burst and the random error, unlike Hamming code or parity code that can basically detect random errors only. Hamming code or parity code may be used for detection of burst errors; but that will increase the design complexity to a large extent. Two-dimensional Hamming Code may be used for detection of burst errors; but in this case not only the complexity increases but the block wise data transmission becomes imperative also. CRC code does not create any such problem. For example, CCITT_CRC with generator polynomial, x16 + x12 + x5 + 1 can detect all single bit errors, all double bit errors, all errors over odd number of bits, all burst errors of length less than or equal to 16, 99.97% of burst errors of length 17 and 99.998% of burst errors of length greater than or equal to 18. The system design of CRC is very simple. These are the reasons behind using CRC in most of the data communication and computer communication networks including ARPANET, Ethernet LAN, and Internet etc. Performance of CRC is quite high[47].

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In FEC, a data is transmitted for one time only. If error occurs in the data, the same is corrected at the receiver. Naturally ECC is used in FEC. In BEC, if data is received with error, the receiver requests the transmitter for the retransmission of that data. Unlike in FEC, in BEC the receiver does not correct but only detect the presence of error in the data. In the BEC technique, the bits-in-error of the data are corrected by the means of retransmission. BEC thus uses EDC and needs a feedback path for requesting the transmitter to retransmit the data or packet in error. As FEC uses ECC, it requires more check bits and more bandwidth for a particular data size, than those are required in BEC that uses EDC. Accordingly in long haul communication, BEC is used rather than FEC. But with increase of channel error probability, throughput efficiency (informative data size in bits divided by actual number of bits transmitted for final correct reception of data) of BEC decreases, due to increased probability of more retransmission expected with increased error probability; whereas throughput remains constant in FEC. However BEC provides higher reliability than that of FEC (Table (3) compares FEC with BEC). Many studies[2-4] show that BEC techniques perform well in many forms of transmission errors, and offer better performance than FEC techniques for wide and practical ranges of signal to noise ratios. This is why in many real applications and in long haul communications BEC is used invariably. Table 3: Comparison of FEC with BEC Forward Error Correction 1. Uses error Correction codes (a) coding efficiency is less (b) requires more bandwidth 2. Correction is done through coding process 3. Throughput remains fairly constant 4. Mostly used in local/short distance distance communication/localized

Backward Error Correction 1. Uses error detection codes (a) coding efficiency is better (b) requires less bandwidth 2. Correction is done through retransmission of erroneous does not require any feedback path copy feedback path is essential 3. Throughput decreases with increased bit error rate 4. Mostly used in long haul/long communication/networks LAN/MAN/WAN etc.

system and wireless communication

2. BASIC BEC TECHNIQUES BEC has three basic techniques. These are: Stop-and-Wait Automatic Repeat Request (S/W), Go-Back-N Automatic Repeat Request (GBN) and Selective Repeat Request (SRQ). In S/W ARQ technique, after transmitting a packet, (a packet is a code appended at both end with flags of start and end, source address and destination addresses and other control bytes), the transmitter waits for an acknowledgement from the receiver before transmitting the next packet. On receiving a packet, the receiver checks, using the error detection technique used in the process, for any error. If no error is found, the receiver sends an acknowledgement, known as positive acknowledgement (ACK), to the transmitter through the feedback path. On the other hand, if any error is detected, the receiver sends a negative acknowledgement (NAK) to the transmitter. On receiving an ACK for the already transmitted packet, the transmitter transmits the next packet. But on receiving a NAK for any transmitted packet, the transmitter retransmits the previously sent packet. In short, until and unless a packet is received correctly

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by the receiver and it is positively acknowledged, the transmitter will not transmit the next packet. However there remain several questions to such operations. What will happen if ACK or NAK is lost in the feedback path? The transmitter waits for a period known as time out period, which is greater than twice the propagation delay between the transmitter and the receiver, for the acknowledgement. If no acknowledgement is received within the time out period, the transmitter retransmits the previous packet. The receiver understands the received packet as retransmitted one by checking the sequence number of the packet and takes decision accordingly. What shall happen if ACK is changed to NAK or vice-versa during the transmission through the feedback path? The change of ACK to NAK is tackled by the same technique, as that is used in case of loss of acknowledgement. When NAK is changed to ACK, the receiver on checking sequence number only detects the change. By this time the previous transmitted packet for which NAK was changed to ACK is not available with the transmitter. This causes a serious problem. The performance of the techniques is measured by a parameter known as throughput efficiency (ν). It is defined as number of the information bits correctly transmitted divided by the total number of bits transmitted for the purpose. If we assume (i) (m, n) code were used in the protocol. (ii) processing time at the transmitter and the receiver for ACK/NAK or packet is negligible, (iii) transmission time of ACK/NAK is negligible and (iv) feedback path is error free; ν(s/w) = n/{(m + RT)E} ...(1) where E = expected number of transmission for successful reception of a packet, R = rate of transmission, T = total round trip delay. When each packet has the same probability that it is received with error, E = 1/β ...(2) where β is the probability that a transmission for a given packet is the last transmission. If P and Pu are the probability that a packet is in error and the probability of the undetected packet error respectively, β = 1 – P – Pu = 1 – P, as Pu << P ...(3) If tp is one way propagation time and tt is the transmit time of a packet, we have: T = 2tp and R = m/tt Using these and equs. (2-3) in eqn. (1) we find: ν(s/w) = {n(1 – P)}/{m(1 + 2a)} ...(4) where a = tp/tt. The throughput efficiency of S/W ARQ is poor. It is because the successful transmission of a packet involves at least two propagation delays in between the transmitter and the receiver. In order to improve throughput, GBN ARQ was developed. In GBN ARQ technique, the transmitter continuously transmits a block of N (N is often known as window size) packets without waiting for the acknowledgement for the individual packet; and keeps the packets in its memory or buffer. The receiver sends only the negative acknowledgement if so detected. The transmitter on receiving NAK for the first time, stops transmission and retransmits all the packets which were transmitted prior to stopping of transmission but starting from the packet for which

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NAK is received; and discards the packets transmitted prior to the packet in error from the memory. For example when 5th (assuming N > 5) packet of the block is the first negatively acknowledged packet when up to Nth packet has been transmitted, the transmitter will then discard first to fourth packets from its memory, and now will retransmit all the packets from 5th to Nth. Worst situation in GBN ARQ occurs, when the first packet of the block is negatively acknowledged, the case of when the whole block of N packets requires retransmission. Best situation occurs when none of the packet is negatively acknowledged, thereby successful transmission of N packets involves with minimum two propagation delay rather one packet being involved with two-propagation time as in S/W ARQ. This gives throughput advantage to the GBN ARQ over S/W ARQ. GBN ARQ may be two types: continuous and non continuous. In the continuous scheme, after transmission of a block of N packets, the transmitter does not have to wait for the acknowledgements of these packets before starting the transmission of the next block. In the non-continuous mode, before starting the transmission of the next block, the transmitter has to wait for the acknowledgements for the packets of the previous block. If the transmit time of a packet/acknowledgement is one unit, we have: N >= (1 + 2a) for the continuous scheme and N < (1 + 2a) for the non-continuous scheme. The throughput efficiency for GBN ARQ is given as: ν(gbn) = {n(1 – P)}/{m(1 + 2aP)} for continuous scheme, = (n/m){1 + NP/(1 – P)}–1 where N = 1 + T/(m/R) = 1 + 2a, [Note that m/r is chosen so as to make N = 2, 3, 4…. of GBN technique. When T = m/R, N = 2 and transmitter goes back by two blocks.] ν(gbn) = {n . N(1 – P)}/{m(1 + 2a)(1 – P + NP)} for non-continuous scheme, ...(5) The through put of GBN ARQ technique is higher than that of the S/W ARQ but still the throughput is a function of propagation delay, a. Selective Repeat Request (SRQ) ARQ further improves the throughput. It operates like that of the GBN ARQ but retransmits only the packet for which negative acknowledgement is received. This means that theoretically infinite buffer is required at the transmitter. It has also two modes of operation, namely continuous and non-continuous. The throughput efficiency is given as: n(srq) = {n(1 – P)}/m for continuous scheme n(srq) = {n . N(1 – P)}/{m(1 + 2a)} for non-continuous scheme ...(6) The problems of the loss and/or the change of acknowledgements in GBN ARQ and in SRQ ARQ are tackled by the same techniques as in S/W ARQ. It is the trade off between buffer size and throughput that plays the role among the three basic BEC schemes, S/W ARQ, GBN ARQ and SRQ ARQ. A comparison in terms of buffer size or memory requirements and throughput efficiency of the schemes is shown in Table 4. In term of the throughput efficiency, we find the clear advantage of SRQ over other two schemes. When the parameter “a” is zero i.e. tp << tt, all the basic techniques become same in term of throughput efficiency, which would be the case. This amounts to say that propagation time, tp is the element that causes the throughput efficiency to vary for the different techniques.

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Table 4: Comparison of basic ARQ schemes Scheme

Theoretically Minimum at transmitter

Memory Space Requirement at receiver

Throughput efficiency

Stop and Wait

1

2

Low

Go-Back-N

N

0

Higher than the stop and wait

Selective Repeat Request

∞

0

Higher than both the stop and wait; and the go-back-n.

3. DIFFERENT MODIFIED TECHNIQUES Data communication is the major issue for the global connectivity. Error free and reliable data transportation is therefore a key challenge. Error is caused during transmission at physical layers. The physical layers are basically two types: Wired that includes Copper, Cable and Fiber; and Wireless that includes Radio link, Satellite link, Microwave link etc. Copper lines are highly erroneous. Early ARQ techniques were introduced for the copper links. The error rate of the fiber link is less. Yet the use of ECC/EDC greatly improves the performance of the fiber. The study[54] reveals that the key performance metrics: Distance, Cost and Capacity in fiber are all improved by FEC. On the other hand, the wireless access to Internet or the mobile computing requires FER (Frame Error Rate) below 10–8[38,55]. Such a low FER could be achieved only with an efficient error control strategies. Thus there is a need to design an efficient Error Control Strategy for reliable data communication. Several modifications have been suggested in literatures to improve throughput of the BEC techniques. Among several modifications a few important modifications are: • Sastry’s modification • Morris’s modification • Weldon’s modification • Towley’s modification • Chakraborty’s modification • Yao technique While the above stated first four modifications are applicable to static error rate channels, the remaining two modifications are applicable to variable error rate channels or dynamic channels. The modifications mentioned above have several issues that need consideration. Weldon’s suggestion for a hybrid technique, a combination of EFC and BEC was not so sound considering the low coding efficiency of ECC used in FEC. Hybrid technique using two or multilevel EDC coding was not addressed earlier. Chakraborty recently suggested a technique for locating the position of bit(s) in error of the packet, so that the receiver can correct the error rather than requesting transmitter to retransmit the erroneous packet. The technique was named as packet combining scheme. It was so named because, error location(s) is(are) detected by XOR operation of earlier received erroneous packet and requested retransmitted packet.

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3.1 Modifications of basic ARQ schemes In order to improve the throughput efficiency, a number of modifications of the basic ARQ schemes have been suggested in literatures[5-8].

3.2 Sastry’s Scheme and Morris’s Modification Sastry[5] suggested that in order to improve the throughput of S/W ARQ, i(i > 1) copies of the packet in error might be retransmitted rather than one copy as in basic ARQ techniques on receiving a NAK for the transmitted packet. Actually by this process the effect of propagation delay is distributed over i copies of packet and not over single copy of the packet as in basic S/ W ARQ technique. Besides, due to the retransmission of i copies of the packet in error, the probability of getting correct copy of the packet by the receiver increases by the power of i; and the throughput increases. The throughput under Sastry’s scheme is given by: ν(Sastry’s S/W) = [n/{(m + RT) + (im + RT)P/(1 – Pi)}] ...(7) The increased throughput is achieved only when: i < [{1 + P(RT/m)}/(1 – P)] ...(8) The maximum improvement is achieved when: ...(9) m = – (logP) . (im + RT) . {Pi/(1 – Pi)} Thus the improvement in Sastry’s S/W ARQ technique is conditional, and generally, the improvement is noticeable when RT is considerably greater than m. For improving the throughput of GBN ARQ, Sastry suggested the retransmission of the packet (or the block) in error for a number of times rather than retransmission of all the packets or the blocks which have been transmitted prior to receiving negative acknowledgement but starting from the packet or the block in error. Sastry calculated throughput efficiency for modified continuous GBN as: ν(Sastry’s GBN) = (n/m){1/(1 + P(1/(1 – P) + 2(N – 1)))}. For improvement, we need that [ν(sastry’s GBN) – ν(GBN)] > 0. This implies that (2P – 1) = 0. And hence Sastry’s technique is superior to the basic technique only when: P > 0.5 That is, when the channel is a high error rate channel like Satellite channels or mobile Multimedia channels[38], Sastry’s technique becomes effective. But in most other situations where P < 0.5, it implies therefore that basic GBN is better than Sastry’s GBN technique. Morris[6] suggested a modification of Sastry’s GBN technique by suggesting that the receiver should not wait for processing the packets or the blocks, which are received in the mean time when a NAK has been sent to the transmitter for retransmission. This improves the throughput over Sastry’s technique over practical ranges of P but improvement is noticeable only when error rate is high. We shall see later, that by modifying Sastry’s scheme, a better technique could be achieved.

3.3

Other Modifications

Weldon[7] and Towsley[8] further suggested some modifications for improving throughput. Retransmission of increasing number of copies with increasing number of repetition of NAKs is one important suggestion. This, in general, increases throughput, but it was shown that if NAK for a packet repeats for more than three times, the modifications suggested give marginal improvement only. The work of Weldon was based on the following strategy:

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1. if ACK is received, send next packet 2. if NAK is received, packet is repeated n1 times 3. if n1 copies are acknowledged as erroneous, the packet is repeated n2 times where n2 > n1. ......... ......... q . if nq–2 copies are received in error, transmit nq–1 copies ......... ......... the process continues till the buffer is full. When the buffer is full, the same number of copies is repeatedly retransmitted till the packet is received correctly. For the purpose of calculating throughput Weldon modeled the system as a link holding S number of packets where: S = [R(3tt + 2tp)]/m. As for a complete transmission, twice propagation period and thrice (two at the end of receiver—one for transmission and another for the next on receiving positive acknowledgement and one at the receiver for acknowledgement) transmission time are used. Weldon defined throughput as the average number of packets transmitted for successful transmission of a packet. Accordingly he calculated throughput. When q = 2 which is the most practical situation, the throughput as per Weldon is as below: n

ν(weldon) = 1/[1 + n1P + ((n1 + S – 1) P (1+ n1 ) )/(1 – P 1 )] To maximize throughput, the choices of n1 shall be as below: n1 = 1 when 0 < SP < 1, n1 = 2 when 1 <= SP and 8 <= SP2 < 1 n1 = >3 when 1 <= SP2 The quantity SP is the expected number of packets in error in the data link. For the most of the links SP < 1, and hence erroneous packets should be repeated once. For long haul links, SP may exceed unity, and then erroneous packets may be repeated twice. However such selection of number of copies to be transmitted is dependent on the link conditions. Moreover there is no rule on how to vary the number of copies i.e., how to select n1, n2 ….nq of the strategy of Weldon. Sastry’s technique also does not collaborate about the selection of number of copies to be sent on receipt of subsequent NAKs. The work of Weldon also suggested that in general if P > 0.1 (high error rate condition), some form of error correction scheme can only help in increasing throughput significantly. The modifications suggested by Moeneclaey and Brunnel[12] becomes superior to conventional schemes when P > 0.5 like Sastry’s modifications.

3.4 Two level coding Weldon’s suggestion for introducing a form of correction code for increasing throughput when P > 0.1 is not sound as correction coding requires more check bits, thereby reducing coding efficiency as well as throughput. To increase throughput at higher error rate conditions, we propose a two level coding scheme for ARQ operation. At the first level of coding, parts of the packet are coded individually and separately, and at the second level of coding, the whole block of the packet is coded as usually. At the receiver, the whole packet is checked for error. If the packet is found in error, parts of the packet are checked one after another to locate the

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part(s) in error. Once located, retransmission of the part(s) in error is requested unlike whole packet as in conventional scheme[1-10]. By the process, in the retransmission slot of conventional scheme, i(i > 1) copies of the part(s) in error can be retransmitted that increase the probability of getting at least a correct copy of the part(s) under retransmission. To illustrate the proposed scheme, we assume that the packet size is fixed and of 512 bytes. It is also assumed that under the conventional ARQ schemes, CRC-32 is in use for error detection. In our proposed scheme, the packet is broken into several parts each of say, 32 bytes. We do first level error detection coding by the simple one bit parity over each part of the packet. Second level of coding over whole of the packet plus the check bits of parts, is done with CRC-32. At the receiver if any packet is checked in error by the first level of checking with CRC-32 (the first level coding at the transmitter refers to second level of detection at the receiver and vice-versa), the second level of checking using single parity bit is started with each part of the packet. On the second level of error checking, if only one part is found in error, the receiver can request for the retransmission of that part only. The transmitter then can retransmit 16(=512/32) copies of the part in error. In general, if n parts are found in error, 16/n copies of each of the parts in error can be retransmitted. When n = 16, the whole packet needs to be retransmitted, and the situation refers to the conventional scheme. In the conventional scheme, the probability, P is the parameter that basically determines the required number of retransmission before finally receiving the same correctly by the receiver. The probability, PP that will be taken for the same purpose under the proposed scheme is given as: PP = (P1 + P2 + P3 + ...... + Pp)/p ...(10) = (P/p){(1 – Pp)/(1 – P)} where p = number of the parts into which a packet is broken. In the derivation we assume that the probability of a packet in error is equal to the probability that any part of the packet irrespective of size, is in error. But in reality the probability of a packet in error is always greater than the probability of the part in error. If a be the BER (Bit Error Rate), the probability (Pn) that a packet of n bits is in error will be: Pn = 1(1 – α)n whereas that the probability (Pn/2) of a packet with n/2 bits is in error will be: Pn/2 = 1 (1 – α)n/2 Therefore: Pn > Pn/2 This only ensures that the benefit of the proposed scheme will be better than that achieved by the equation(10). If α is independent of bit position, and m be the size of the packet, the exact derivation of PP will be: PP = (P11 + P22 + P33 + ...... + Ppp)/p = [(ΣPii]/p ...(11) (m/i) where Pi = 1 – (1 – α) for i = 1 to p(Note that P1 = P) However gain of the proposed scheme over the conventional scheme increases with p. But as we go on increasing, coding efficiency decreases, this is the disadvantage of the proposed scheme. Increase in decoding complexity can be measured by 2c where p <= 2c. The gain versus increase in decoding complexity is illustrated in the table (5):

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Table 5: Gain versus Decoding Complexity of Proposed Scheme-I p(parts)

P(Probability of packet in error under conventional Scheme)

Pp(Probability of packet in error under proposed Scheme)

Increase in decoding complex

16

0.1 0.01 0.001

0.00695 0.000631 0.0000626

16

8

0.1 0.01 0.001

0.014 0.00126 0.000125

8

1 (Conventional Scheme)

0.1 0.01

0.001 0.01

0

0.01

0.02

16

3.4.1 Parity Selection in Two Level Coding There is no rule for selection of parity even or odd that would be used in any scheme. We shall propose a rule. The rule is to solve a problem often faced by telecommunication engineers. A problem of lack of dc coupling[20] between data source and sink is often faced by telecommunication engineers. This lack of coupling causes a gradual “sag” of the pulse-top towards zero. To prevent such happenings, the telecommunication engineers are in need of a disparity reduction code, “disparity being the difference between the numbers of 1s and the numbers of 0s in a code word.” Besides, ‘a high disparity may be undesirable, since a predominance of 1s and 0s may lead to a large low-frequency component, or even a d.c. offset in the binary sequence, which will make propagation through a medium with no d.c. coupling’ Proper selection of the type of the parity can provide disparity reduction code (table (6)). Table 6: Table to derive a rule for selection of parity Information code

Parity bit if odd parity is selected

(with number of information bits per code n = 1) 0 1

1 0

0 0

(with n = 2)

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Number of disparity code if odd parity is used

2

Number of disparity code if even parity is used

Remark

0

Use odd parity

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00 01 10 11

1 0 0 1

0 1 1 0

(with n = 3) 000 001 010 011 100 101 110

1 0 0 1 0 1 1

0 1 1 0 1 0 0

111

0

1

317

0

0

No choice

0

6

Use even parity

When the parity is proposed to be used in one level of coding in the proposed scheme, the following rule for selection of parity may be applied when n is odd and n <= 7: (i) use odd parity if (n + 1)/2 is odd, and (ii) use even parity if (n + 1)/2 is even. For selection under the proposed rule, the disparity will be reduced to a large extent as evident from the following results that are based on finding in Table 6: (i) if n = 1, the number of zero disparity code word is 2 out of total code words 2 (ii) if n = 3, the number of zero disparity code words is 6 out of total code words 8, (iii) if n = 5, the number of zero disparity code words is 20 out of total code words 32 (iv) if n = 7, the number of zero disparity code words is 70 out of total code words 128 However when n is even, there will be not a single zero disparity code as because in that case, code word size, m (which is equal to n + 1) is odd. On the other hand as n increases, the number of the zero disparity code words decreases. Therefore it is proposed to use n = 7 for first level of coding.

3.5 Packet Combining Scheme A recent important modification of ARQ is due to Chakraborty et al.[9]. They suggested EARQ known as Extended ARQ scheme (also known as packet combining scheme), particularly suitable for SRQ technique. The EARQ technique is the extension work of Sindhu[18]. Sindhu suggested locating errors by xor operation between erroneous copies. The idea is close to hybrid type-II technique[15-17] described later in the paper. In the conventional SRQ the receiver discards the packet in error. But in SRQ-EARQ scheme, the receiver retains the packet in error. If the requested retransmission copy is correct, the receiver discards the retained packet in error from the buffer, and accepts the correct copy received under retransmission. If the retransmitted packet is found in error, the packet is xored bit wise with the erroneous retained packet stored in the receiver’s buffer. Error location(s) is (are) indicated by the presence of 1 at the output of xor operation. To obtain correct copy the bit inversion of the bit(s) in error location(s) is done one after another followed by the FCS (Frame Check Sequence) check. Chakraborty deduced that if there is no double error (double error is defined as “when two copies are erroneous, there is at least one bit position in which both copies have an error”), throughput of EARQ or modified SRQ is given as:

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ν(chakraborty) = 1/(1 + P) The operation of the Chakraborty’s technique fails when double error occurs. The probability P(i, j) that two copies with i and j errors, respectively, have a double error is given as: P(i, j) = 1 – [(n – i) ! (n – j) !/ n ! (n – i – j) !] The EARQ technique is found suitable when the packet size is small. On the other hand, the processing of the receiver increases in the technique. Besides, the technique fails if error occurs at least at the same location of both the retained erroneous packet and the retransmitted packet. The complexity of the bit inversion process to get correct copy is given as: C = 22nα – 2 where a is BER. In order to avoid the complexity, it was suggested that when number of 1s under XOR operation exceeds a certain number, Nmax, the pair is discarded, and a retransmission is requested. In such cases, P(i, j) = 1 – [(n – i) ! (n – j) !/n ! (n – i – j) !] if i + j <= Nmax = 1 if i + j > Nmax

3.6 Modified Packet Combining Scheme (MPC) We propose a Modified Packet Combining (MPC) technique. In the MPC technique, on getting a retransmission call from the receiver the transmitter can send i(i > 1) copies of the requested packet. Receiver getting i copies, can now make a pair-wise xored to locate error positions. For example if i = 2, we have three copies of the packet (Copy-1=the stored copy in receiver’s buffer, Copy-2=one of the retransmitted copies, Copy-3=another retransmitted copy) and three pairs for xor operation: Copy-1 and Copy-2 Copy-2 and Copy-3 Copy-3 and Copy-1 Assume that an actual packet 10100011 was received as: Copy-1 = 10101011 Copy-2 = 10101111 Copy-3 = 10100001 when we have under xored operation: Copy-1 xored Copy-2 (say, C12) = 00000100 (one bit in error) Copy-2 xored Copy-3 (C23) = 00001110 (three bits in error) Copy-3 xored Copy-1 (C31) = 00001010 (two bits in error). Now we have to define with which copy the bit inversion shall start and how to proceed thereafter. We define an algorithm for the purpose as below. Make a table (see Table 7) in ascending order of number of bits in error as indicated by the xor operation. The bit inversion and the FCS checking process shall begin with the common copy indicated in the last column of the table so prepared, and proceed down the table if required. If all the inversions do not yield any result, the receiver has to go for requesting further retransmission as in EARQ of Chakraborty et al. As per table (7) in this example, the detection of error location and consequent bit inversion will start with Copy-1 and if required will be followed by Copy-3 and then by Copy-2.

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Table 7: Algorithm of MPC Comparing pairs

Number of bits in error (x)

Common copy in two

Copy-1 and Copy-2

1

consecutive (x) Copy-1

Copy-1 and Copy-3

2

Common in first two xs Copy-3

Copy-3 and Copy-2

3

Common in next two xs Copy-2

On the first retransmission and subsequent application of MPC, if the result does not yield, retransmission for the second time may be done with variable i. Second retransmission may be done with i = 4 (see proposed scheme-III in next chapter). Copies received on earlier retransmission and first transmission are kept at the receiver. With i = 4 at second retransmission, we have 7 copies for comparisons and we have the followings obtained by different pairs under XOR operations: Cij for i = 1 to 7 and j = 1 to 7 but i ≠ j i.e., we are having: C12, C13, C14, C15, C16, C17, C23, C24, C25, C26, C27, C34, C35, C36, C37, C45, C46, C47, C56, C57, C67. These are typically like the parameters used in topological design of users’ sub network using Easu-William algorithm[21]. For the purpose of correction we can follow the procedure described earlier using table like that of Table 7. But in the present case we may not get common copy between two successive rows. In that case, correction process may be started with individual copy starting with the copy having lowest numbers of bit in error. We illustrate with an example. Example: We assume an original data 1010101001010100. We shall underline the bit(s) in error, but these are unknown to the receiver. We assume that on first transmission the copy was received as: Copy-1 = 1010100001010100 On the first retransmission with two copies (i = 2), we assume that the received copies were: Copy-2 = 0010101011010101 Copy-3 = 1010101101010100 On the second retransmission with four copies (i = 4), we assume that the copies were received as: Copy-4 = 1010101001010110 Copy-5 = 0110101001010101 Copy-6 = 0010101001010101 Copy-7 = 1000101001010101 Thus on XOR operation between two different pairs of copies, we have the followings. On the sides of Cij we have mentioned the number of locations having 1 (i.e., number of bit positions where difference in bits between comparing pairs of copies are identified) in the first bracket:

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C12 = 1000001010000001 (4) C13 = 0000001100000000 (2) C14 = 0000001000000010 (2) C15 = 1100001000000001 (4) C16 = 1000001000000001 (3) C17 = 0010001000000001 (3) C23 = 1000000110000001 (4) C24 = 1000000010000011 (4) C25 = 0100000010000000 (2) C26 = 0000000010000000 (1) C27 = 1010000010000000 (3) C34 = 0000000100000010 (2) C35 = 1100000100000001 (4) C36 = 1000000100000001 (3) (3) C37 = 0010000100000001 C45 = 1100000000000011 (4) C46 = 1000000000000011 (3) (3) C47 = 0010000000000011 C56 = 1000000000000000 (1) C57 = 1110000000000000 (3) (3) C67 = 1010000000000000 The list of Cijs’ with the ascending order of the number of 1s in them is, therefore: C26 C56 C13 C14 C25 ...... ...... ...... ...... ...... C24 C35 C45 The correcting process now be started as: Step I: As C26 is the first in the list, we shall have to apply bit inversion technique successively in copy 2 and in copy 6 for correction. But in applying, bit inversion on 9th bit location, as indicated by C26, on both the copies we do not get back the corrected data. Step II: We repeat the process with next in order of the list, C56, but fail to correct. Step III: We repeat the process with next in order of the list,C13, but fail to correct.

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Step IV: We repeat the process with next in order of the list, C14. Applying the bit inversion to the bit location 7th of Copy-1, (rule is applied first at a single location one after another for all the locations indicated in error, then with possible pairs, etc.), we get the correct copy. The correcting process stops now. The disadvantage of MPC is that more processing is required in the receiver. The advantages of MPC over Chakraborty’s scheme are: 1. double bit error may be corrected in MPC, 2. throughput efficiency is comparable, and 3. by varying i, variable bit rate channel may be controlled. The two level coding scheme is the extension of Weldon scheme. The advantages and the disadvantages of the two level coding scheme have also been illustrated. It is observed that the two level coding scheme is superior to the Weldon scheme in terms of throughput and the range of P, the probability of packet in error, over which the technique is applicable. Although the decoding complexity is a problem for the proposed scheme, yet if we consider the low coding efficiency of the ECC required in Weldon technique, we can comfortably say that two level coding scheme is better in all respect compared to Weldon technique. The modified packet-combining scheme is superior to the packet-combining scheme suggested by Chakraborty. The advantages are discussed. The major advantage is that by the modified packet-combining scheme, multiple bit error can be corrected by the receiver. Besides this advantage is achieved with the copies received at the receiver, thereby saving the retransmission delay. Of course this is at the cost of required additional memory at the receiver.

3.7 ARQs for Variable Error Rate Channels Basic ARQ techniques and the modified techniques discussed so far have two flaws: 1. they do not suggest anything to tackle variable error rate conditions of the channel (dynamic channels), rather assume that the channel’s error state is static; and 2. although Weldon suggested to vary the number of copies for retransmission with increasing repetition of NAKs which may sound to be appropriate to a dynamic channel, yet there is no concrete suggestion for how to vary the copies on retransmission.

3.8 Yao Technique The practical channels are always dynamic and they are all variable error rate channels. Yao[11] proposed a modified ARQ scheme to tackle error of the dynamic channels. Yao assumed a twostate channel for his scheme. The channel may stay either in the state of low error rate or in the state of high error rate, which may be respectively, denoted as L and H states. Yao proposed that when the channel is in L state, the transmitter follow the conventional non-continuous GBN ARQ scheme. When the channel is in H state, the transmitter switches to non-continuous GBN ARQ scheme with i(i > 1) copies of transmission[12,13]. The “i” copies of transmission is to mean the transmission of i copies of block of N packets in GBN scheme. At the beginning, the transmitter starts working with normal non-continuous GBN scheme, and if and when the transmitter receives two or more than two contiguous NAKs, the transmitter switches to noncontinuous GBN with i copies transmission. It amounts to say that receipt of two or more contiguous NAKs is the signal of H state of the channel. The throughput as Yao derived for the proposed scheme with the assumption that a << 1 (that is propagation time << transmit time) is given as:

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ν(yao) = (n/m){(ν1P1) + (ν2Ph)} ...(12) ν1 = Throughput efficiency when the system is in L state = Throughput efficiency of the conventional non continuous GBN = (n/m)[(1 – P)/{1 + (N – 1)P}] ν2 = Throughput efficiency when the system is in H state = (n/m)[(1 – Pi)/{i + (N – 1)Pi}] Pl = Probability that the system is in L state = Pl(1 – p1) + (1 – Pl)p2 = p2/(p1 + p2) Ph = Probability that the system is in H state = p1/(p1 + p2) p1 = Transition probability of the system from L state to H state, p2 = Transition probability of the system from H state to L state. Yao concluded that the proposed system becomes superior to the conventional scheme only when (i – 1) > N. Exact analysis of the adaptive GBN technique was reported in a recent study of Chakraborty et al[19,48]. Most of the studies on GBN of dynamic channels done before Chakraborty were based on Gilbert type channel. Gilbert channel has two states: Good state and Bad state. Good state refers to the condition that the probability that the packet is in error is zero; and bad state refers to the condition that the probability that the packet is in error is one. Real channels hardly conform to Gilbert model. Chakraborty extended his studies to the real channels by a new model of channel. Chakraborty assumed that 1. Pg is the probability that a packet is in error when the channel is in good state 2. Pb is the probability that a packet is in error when the channel is in bad state where Pb > Pg. 3. The channel crosses from one state to another at Pc, the probability that a packet is in error such that Pg < Pc < Pb. The Chakraborty’s model conforms to real channel and his studies is more appropriate to the practical channels. where

3.9 Chakraborty’s Technique In a recent work, Chakraborty et al[14] extended the packet-combining scheme for time varying channel. The work was formed over the observation that basic ARQ schemes even with optimal packet size may be inefficient in a high error rate and time varying channels like mobile and wireless systems. The work studied the dependence of efficiency of the hybrid type-II scheme over packet size in reference to EARQ scheme[9]. In hybrid type-II[15-17], a packet is coded in two parts: at the first part data is coded with parity bits for the error detection and at the second part, coding is done by parity bits for error correction by using an invertible code. Studies concluded that complex type of coding as required in hybrid type-II scheme may not be necessary for reliable and efficient data transfer so long BER is not consistently high (>> 2 × 10–2). Satisfactory performance with SRQ and with EARQ technique may be achieved up to BER of 10–3 and 4 × 10–2 respectively.

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3.10 New Schemes We propose two schemes to combat variable error rate conditions. The techniques are basically the modification of Sastry’s scheme so as to make it applicable to variable rate error channels. To do that in the proposed schemes a definite rule for selection of the parameter “i” of the Sastry’s scheme for repeated retransmission was duly addressed. The absence of a definite rule for selection of “i” under Sastry’s modification was discussed earlier. We propose two protocols for the selection of “i”. The selection of “i” under proposed protocols would be based on receipt of the number of NAKs for a particular packet. When repeated NAKs for a transmitted packet are being received, the channel is assumed to tend toward high error rate and vice -versa. The protocols are illustrated in Table 8. As the number of NAKs received consecutively increases for a transmitted packet, the number of copies of the packet to be retransmitted increases under both the proposed protocols. In one protocol increment is doubling at each step, whereas in another protocol increment follows binary exponential rule. Distribution of propagation delay over numbers of packets under Sastry’s scheme and proposed scheme is shown in table 9. Of course sending i(i > 1)copies of packet amounts to sending more and more bits that causes more bits in error; but the technique becomes superior as the probability (PA) of getting all packets in error decreases. The trade off mentioned is the trade off between the number (M) of bits that will be in error on average and ‘PA’. If a is Bit Error Rate and the average packet size is n bits, we have M = α . i . n and PA = (1 – (1 – α)n)i If we assume the average packet size is 72 bytes (n = 72 × 8 bits) (minimum packet size of IEEE 802.3 LAN); the trade off will be typically as below: Case I: α = 10–3 For i = 2, M = 1.15 and PA = 0.192 For i = 4, M = 2.30 and PA = 0.037 For i = 8, M = 4.60 and PA = 0.001 Case II: α = 10–2 For i = 2, M = 11.5 and PA = 0.993 For i = 4, M = 23.0 and PA = 0.987 For i = 8, M = 46.0 and PA = 0.975 Comparison of these two cases yields that: (a) the effect of PA overshoots that of the M and (b) the effectiveness of repeated retransmission of variable i copies is more pronounced at higher α. Table 8: Illustration in details of proposed scheme-III Number of copies of packet to be retransmitted Number of times NAK is received for a particular packet

Protocol–I

Protocol–II

First time

2

2

Second time

4

4

Third time

6

8

: : :

: : :

: : :

n-th time

2n

2n

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Table 9: Distribution of propagation delay over packets Number of consecutive NAKS before packet of received correctly

The propagation time distributed over number of packets

Sastry With i = 2

Proposed scheme–III Protocol-I

Proposed Scheme–III Protocol-II

1

2T over 3 packets

2T over 3 packets

2T over 3 packets

2

3T over 5 packets

3T over 7 packets

3T over 7 packets

3

4T over 7 packets

4T over 13 packets

4T over 7 packets

4

(j + 1)T over (2j + 1)

(j + 1)T over j[1 + Σ2j]

(j + 1)T over j [1 + Σ2j]

packets

packets j = 1

packets j = 1

Under the proposed schemes, we have the following situations: Number of times NAK is received for a particular packet 1 2 3 4 ..... j Number of copies to be retransmitted under Protocol–I with starting copies = 2 2 4 6 8 ..... 2j In general with starting copies = i, where i > 1 i 2i 3i 4i ...... ji Number of copies to be retransmitted under Protocol–II with starting copies = 2 2 4 8 16 ...... 2j In general with starting copies = i, where i > 1 i3 i4 ...... ij i i2 Probability that the set is last set under protocol–I with starting copies = 2 1 – P2 1 – P4 1 – P6 1 – P8 ...... 1 – P2j Probability that the set is last set under protocol–II with starting copies = 2 1 – P2 1 – P4 1 – P8 1 – P16 ...... 1 – P2j Probability of the occurrence of set itself under protocol-I with starting copies = 2 j −1

1+ Σ 2 i

P P . P2 P . P2 . P4 P . P2 . P4 . P6 ...... P i = 0 Probability of the occurrence of set itself under protocol-II with starting copies = 2 P P . P2 P . P2 . P4 P . P2 . P4 . P8 ...... Pj2–1 Henceforth we shall analysis with copies under first retransmission = 2 only. Thus the probability of expected number of retransmission for successful reception of a packet becomes: ∝

E = Σ 2j.P.P j=1

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j−1

Σ

(1 – P2j)

for protocol I

ADVANCED ERROR CONTROL TECHNIEQUES IN NETWORK ∝

E = Σ 2 j . P2

j

j=1

−1

j

(1 − P 2 )

for protocol II

325 ...(3)

whereas throughput efficiency is given as: η = n/{(m + RT) + (m + RT)E} We shall compare the proposed scheme with basic scheme and Sastry’s in terms for throughput efficiency. Proposed schemes under protocol I and protocol II are identical so long the NAKs for the second time are received for a particular packet. We consider a two state model of channel: high error state (H) and low error state(L). This model conforms to simplified Gilbert model[58] of the channel. When consecutive two NAKs are received for a packet we call it state H, and when only one NAK is received for the packet we call it L state. We consider two cases of transition as in Fig. (1), with state transition probability S1 = 0.8

H

p1 = .75

S2 = .4 Case - I

S2 = 0.5

L

p2 = .75

H

p1 = .25

L

p2 = .25

S2 = .5 Case - II

Fig. 1: Illustration of proposed protocol in two state model

as shown; and p1 and p2 are the P of H state and that of L state respectively. The P for channel when Sastry mode is applied will be: Case I: P = 0.75 × 0.8 + .25 × 0.4 = 0.5 Case II: P = 0.75 × 0.5 + .25 × .25 = 0.5 The choice of i under Sastry model will be equivalently: Case I : i = 4 × 0.4 + 2 × 0.8 ≈ 3 Case II : i = 4 × 0.5 + 2 × .5 = 3 For the cases under consideration, the throughput efficiency for the schemes is : with p = 0.5 ηbasic = n(1 – P)/(m + RT) 4 ηsastry = n/((m + RT) + (3m + RT)P/(1 – P )) with p = 0.5 ηproposed = [ S2n/{(m + RT) + (3m + RT) p1/(1 – p14)} + S1n/{(m + RT) + p2/(1 – p24)}] For the given system of Sastry with n = 960, m = 980, R = 2400 bits per sec, T = 600 msec, we have the following results. Case I : ηbasic = 31.2%, ηsastry = 37.7%, ηproposed = 38%, Case II : ηbasic = 31.2%, ηsastry = 37.7%, ηproposed = 50% This shows the superiority of proposed schemes over basic and Sastry’s scheme of SW ARQ. The comparison of the proposed scheme with the Sastry’s scheme can be enumerated on practical terms also. In practice, after receiving consecutive three NAKs or the third NAK of any transmitted packet, the transmitter terminates the session. On this proposition, the number

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of times a packet is transmitted before the termination of the session by the transmitter under the schemes with i = 2 will be 5 under Sastry’s scheme and 7 under the proposed scheme for both the protocols. This being the case, the probability that out of total number of the copies transmitted for a packet before the possible termination of the session, at least one packet is received at the receiver correctly will be (1 – P5) in case of the Sastry’s scheme and (1 – P7) in case of the proposed scheme for both the protocols. In general, the probability in question will be (1 – P(1+2i) ) for the Sastry’s scheme, (1 – P(1+i+2i)) for the proposed scheme under protocol-I, and (1 – P(1+i+i*i)) for the proposed scheme under protocol-II . This clearly shows the superiority of the proposed scheme to the Sastry’s scheme. The proposed schemes provide better throughput efficiency than so far known SW ARQ protocols, and the proposed schemes are applicable to variable error rate channel. Yao suggested a protocol for variable error-rate channel in which protocol changes from SW to GNB and viceversa based on error-state of the channel. Implementation-wise proposed scheme are simpler than Yao scheme. The proposed scheme may be extended to GNB and SR ARQ protocols. The further modification are discussed in Appendix B. 3.10.1 ARQ Schemes Under Practical Situations: In practical case, if NAKs for a particular packet are received for consecutive three times, the data transmission session is terminated. Thus under each session, the average number of times (n) a packet is transmitted and or retransmitted under different schemes are as below: Conventional ARQ: nc = 1 . (1 – P) + 2 . P . (1 – P) + 3 . P2 . (1 – P) = 1 + P + P2 – 3P3 Sastry ARQ (with i = 3): ns = 1 . (1 – P) + 4 . P . (1 – P3) + 7 . P4 . (1 – P3) = 1 + 3P + 3P4 - 7P7 ARQ with variable i: np = 1.(1 – P) + 3 .P. (1 – P2) + 7 . P3 . (1 – P4) = 1 + 2P + 4P3 – 7P7 For successful transmission of a packet, a number of sessions may be required. Therefore average number(N) of transmission and/or retransmission required for successful transmission of a packet over sessions will be: Conventional ARQ: Nc = nc(1 – P3) + 2nc . P3(1 – P3) + 3nc P6(1 – P3) + ...... = nc/(1 – P3) Sastry’s ARQ (with i = 3): Ns = ns(1 – P7) + 2ns . P7 . (1 – P7) + 3ns . P14 . (1 – P7) + ...... = ns/(1 – P7) Proposed Scheme III ARQ: Np = nP(1 – P7) + 2nP . P7 . (1 – P7) + 3 nP . P14(1 – P7) + ...... = np/(1 – P7) ...(14) The Proposed Scheme is evident as the optimization between the Conventional Scheme and the Sastry’s Scheme. If P is such that higher order terms of P would be neglected, we find that: Nc = 1 + P; Ns = 1 + 3P and Np = 1 + 2P. And hence: Nc < Np < Ns The throughput in different cases becomes as below: νc = (1 – P3)/{nc . (1 + 2a)} for Conventional ARQ νs = (1 – P7)/{ns . (1 + 2a)} for Sastry’s ARQ ...(15) νp = (1 – P7)/{np . (1 + 2a)} for Proposed Scheme III ARQ Analytical results based on above findings (eqn. (15)) are as below: 1. all the schemes give same throughput when (a) P <= 10–6 at a = 0.5, (b) P <= 10–4 at a = 1 or (c) P <= 10–4 at a = 1.5.

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2. As “a” increases, the proposed scheme and the Sastry’s scheme become nearer to the conventional scheme at higher value of P. This is due to distribution of propagation delay over number of packets in the proposed and Sastry’s scheme rather than one packet as in conventional scheme. 3. for a particular “a”, for P higher than the thresholds mentioned above at (1), throughput of conventional scheme > throughput of proposed scheme > throughput of Sastry scheme. This is reverse conclusion to that we did under all possible retransmission case. But the conclusion is tenable considering infinite number of sessions. Yet we find that proposed scheme is superior to Sastry’s scheme and it is true in all possible cases. 3.10.2 GBN and SRQ under different schemes We shall derive the throughput of GBN and SRQ under Conventional, Sastry and Proposed scheme. For that purpose, under the Proposed and the Sastry’s schemes, the retransmission with a number of copies (say i, i > 1) may be modeled as the retransmission of a single copy with probability in error as Pi. Under such a model we can derive the average probability that a packet in error over a session of maximum three transmissions as below: Table 10: Average p under different cases under single packet model Initial transmission

First retransmission

Second retransmission

Average

Conventional Scheme

P

P

P

Sastry’s Scheme with i=3

P

P3

P3

(P + P3 + P3)/3 (Say equals to S)

Proposed scheme

P

P2

P4

(P + P2 + P4)/3 (Say equals to F)

P

It is found that when higher order terms starting with P2 are negligible both the Sastry’s and the Proposed schemes have same average probability (F = S), whereas probability that the packet in error of conventional scheme becomes higher than S or F. This reflects the superiority of the proposed and Sastry’s schemes to conventional scheme. Again if the higher order terms starting with P6 are neglected, the proposed scheme becomes superior to both the Sastry’s scheme and the Conventional scheme. In this case F < S < P. But when all the terms are retained it is found that Proposed scheme becomes superior to the Sastry’s Scheme when P > 0.5. This means that the proposed scheme is superior to the Sastry’s Scheme at low as well as high P. This is due to the presence of different higher order terms of P in F. Based on the average probability that a packet is in error, as modeled above and as shown in the Table 10, we can write the throughput efficiency for different cases as below after the results derived in [46]: νcg = (1 – P)/(1 + βP) for conventional GBN νsg = (1 – S)/(1 + β . S) for Sastry’s GBN νpg = (1 – F)/(1 + β . F) for proposed GBN νcs = (1 – P) for conventional SRQ νss = (1 – S) for Sastry’s SRQ νps = (1 – F) for Proposed SRQ ...(16) where β will be “2a” as per [49] on the assumption of the normalized transmit time as unit.

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The numerical results based on eqn.(16) as per run of a computer program is given: 1. in all cases, proposed scheme is superior to Sastry’s scheme and conventional scheme 2. at very low P(P < 10–4), the difference in performance of the different schemes vanishes 3. as “a” increases, performance of all the schemes goes down. [Note: For the purpose of comparison we have to consider constant “i” under Sastry’s and variable “i” under proposed scheme III such that total number of transmitted and retransmitted copies become same. This is possible if i = 3 in Sastry’s Scheme and i = 2 at the first retransmission under proposed scheme-III. Then up to second retransmission:

Total copies transmitted and retransmitted = 1 + 3 + 3 = 7 under Sastry’s Scheme Total copies transmitted and retransmitted = 1 + 2 + 4 = 7 under proposed scheme. This is why in our above calculation we have always considered Sastry’s scheme with i = 3] 3.10.3 Issues of sending different signal waveforms for repeated retransmitted copies Data is transmitted in form of electric signals in the line. Different noise signals when mixed with transmitted signal wave, error occurs. If signal waveforms are different, the same channel noise may cause different error-in different locations for example. Thus when a number of copies are being retransmitted it is better to send different waveform for them. It is possible by some fixed keying mutually known to both transmitter and receiver. For example, if key operation is XOR, transmitter and receiver can select following keys for a byte data for our proposed technique of 2i copies retransmission where i refers to the instant of retransmission and up to i = 2: A. Key for first copy of first retransmission: 11111111 B. Key for second copy of first retransmission: 10101010 C. Key for first copy of second retransmission: 00000000 D. Key for second copy of second retransmission: 01010101 E. Key for third copy of second retransmission: 00110011 F. Key for fourth copy of second retransmission: 00001111 If the data byte to be transmitted is 10011010, the situation for retransmission under normal operation and under the proposed operation will be as follows (Table 12): Table 11: Showing the case of sending different waveforms for same data under repeated retransmission Case

Retransmitted data under normal operation

Retransmitted data under proposed operation

A

10011010

01100101 (due to XOR of 10011010 and 11111111)

B

10011010

00110000 (due to XOR of 10011010 and 10101010)

C

10011010

10011010 (due to XOR of 10011010 and 00000000)

D

10011010

11001111 (due to XOR of 10011010 and 01010101)

E

10011010

10101001 (due to XOE of 10011010 and 00110011)

F

10011010

10010101 (due to XOR of 10011010 and 00001111)

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The receiver will get back the original by decoding with known keys of different retransmission positions. As under proposed scheme retransmitted data are different in form, the error location etc that may occur in the retransmitted data may be different under same noise pattern. This situation may greatly improve the error detection and correction capability of Chakraborty’s scheme and our proposed scheme of MPC.

3.6 Application of multilevel coding scheme in variable error rate channel The multilevel coding scheme was introduced earlier. In the scheme, the packet is divided into several parts. Each part is sent under a cover of error detection code, while all such parts are taken together and further covered under a detection code. Multilevel coding scheme may be applied to tackle the variable error rate channel. When the channel is in low error rate condition, the system may use two level coding. When the channel transits to high error rate condition, the system may use three levels or multi level coding. The state of the channel, for the purpose may be determined by the receipt of the number of the successive NAKs for a transmitted packet.

4.1 Issues of error in Feedback Path All the ARQ techniques discussed so far assume that feedback path is error free. When feedback path is error free, the transition matrix of the system is the transition matrix of the forward path; and it is then given as[56]: P = p00 p01, p10 p11 where 0 and 1 respectively denote successful and erroneous transmission. The model is equivalent to Gilbert model. The assumption that the feedback path is error free is far from reality particularly in high error prone situation like long haul communication and satellite communication. When feed back path is erroneous, we have to consider the transition matrix of the feedback path in order to get the transition matrix of the pair of the channels. The transition matrix of the pair of channels will actually be the transition matrix of the system. The transition matrix of the feedback path can be defined as: Q = q00 q01, q10 q11 Then, the transition matrix of the system is given as: θ = P ⊗ Q where ⊗ represents the Kronecker product between matrices[57]. In that case, the channel states can be defined as: l = 0 : that refers to (00) state meaning correct transmission and correct feedback; l = 1 : that refers to (01) state meaning correct transmission and erroneous feedback; l = 2 : that refers to (10) state meaning erroneous transmission and correct feedback; l = 3 : that refers to (11) state meaning erroneous transmission and erroneous feedback. We propose scheme to deal with the system with transition matrix of θ that has four states as defined by four different 1s mentioned above. Feedback error either can cause the loss of ACK /NAK or may change the form of the acknowledgement (ACK to NAK/NAK to ACK ). The change of acknowledgement is more serious than the loss of acknowledgement. In this chapter we shall propose a technique to deal with both the effects of the feedback error.

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4.2 Majority technique It is proposed that in order to reduce the effect of feedback error on acknowledgement, receiver can transmit multiple I(I > 1 and an odd integer) copies of ACKs / NAKs for a single reception. Transmitter will take the decision of retransmission or otherwise using majority rule. If I = 3, the operation is explained as bellow. Say, RECEIVER sends ACK thrice corresponding to a received frame. Transmitter may receive ACK as Transmitter takes decision as ACK ACK ACK (No Error) ACK ACK NAK (Single Error)

To transmit next frame To transmit next frame

ACK NAK ACK (Single Error) ACK NAK NAK (Double Error)

To transmit next frame To retransmit the previous frame

NAK ACK ACK (Single Error) NAK ACK NAK (Double Error)

To transmit next frame To transmit this previous frame

NAK NAK ACK (Double Error) NAK NAK NAK (Triple Error)

To retransmit the previous frame To retransmit the previous frame

When receiver sends NAK, the above decision table will be applicable but error will be just reversed. For this operation to cope with situation of double or triple error NAKs (NAKs convert to ACKs, it is required that the memory at the transmitter will be minimum 2 for SW ARQ. The above illustrated scheme of the proposed technique with I = 3 is like 1/3 rate code[29] used in FEC. FEC with 1/3 rate code is being used now a days in ad hoc networks[29].

4.3 Analysis of the majority scheme for SW ARQ If P1 and P2 are respectively the probability of frame in error in forward path and that of ACK/ NACK in feedback path, we can have[6], for SW ARQ. 1 = k(1 + (P1/1 – P1) + (P2/1 – P2)) ...(17) where k is a constant. In case of proposed repeated (I = 3) transmission of acknowledgement, we have : 1 = k(1 + (P1/(1 – P1)) + [3C2 P22(1 – P2)/{1 – 3C2

P22(1 – P2)}] + ((P32/(1 – P32))) ...(18) as because double or triple erroneous acknowledgements, in the scheme, contribute towards P2 and single erroneous acknowledgement has no effect. Third and fourth terms in equation (18) respectively correspond to probability of double and triple erroneous acknowledgements. Throughput is the inverse of 1. For a set of P1 and P2 , the throughput for basic SW ARQ and proposed technique (I = 3) is shown Table 12. In table 13 we have given a comparison of throughput of shutter ARQ [8] (with N = 5) with that of the shutter ARQ with proposed repeated transmission of acknowledgement (I = 3). As expected, % increases in throughput increase with P2. The increased throughput is considerable when P2. > .5 . This conforms to the study of Sastry that for higher error rate, repeated retransmission is more effective.

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Table 12: Comparison of throughput of basic SW with that of proposed Scheme-IV P1

P2

Throughput in basic SW

Throughput in proposed SW

% increase of throughput in proposed scheme

0.1

0.01

0.891

0.899

0.89

0.1

0.1

0.818

0.877

7.2

0.1

0.5

0.473

0.539

13.95

0.1

0.8

0.195

0.359

83

Table 13: Comparison of throughput of shutter SW with proposed scheme-IV P1

P2

Throughput in SW with shutter N = 5

Throughput in SW with shutter (N = 5) and proposed re-transmission

% increase of throughput in proposed scheme

0.1

0.01

0.199

0.1999

0.45

0.1

0.1

0.1956

0.1988

1.6

0.1

0.5

0.1666

0.1741

13.95

0.1

0.8

0.111

0.1498

83

REFERENCES 1. (a) C T Bhunia, Error Controls Techniques in Network, Electronics For You, 1999. (b) C T Bhunia, Error Control in Networks, Information Technology, Nov. ’99, pp. 14-18. (c) C T Bhunia, Data Security and Privacy, Information Technology, May ’93, pp. 33-34. (d) C T Bhunia et al, A cascaded Technique of Error Control, Eastern Regional Conference of CSI, 1995, Siliguri. (e) C T Bhunia, Integrated Solution to Security and Accuracy Problems of Data Communication, Indian Journal of Engineers, 1996. (f) C T Bhunia, Data Security, CSI Communication, Bombay, July 2000. (g) C T Bhunia, Error Control Strategies, Electronics For You, New Delhi, Sept. 2000. 2. S Lin and C Costello Jr, Error Control Coding : Fundamentals and Applications, Englewood cliffes, N J Prentice Hall, 1983. 3. R J Beniece and A H Frey Jr, An analysis of retransmission schemes, IEEE Trans Comm Tech, COM-12, pp. 135-145, Dec. 1964. 4. S Lin, D Costello Jr and M.J. Miller, Automatic repeat request error control schemes, IEEE Comm Mag, 22, pp. 5-17, Dec. 1984.

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5. A R K Sastry, Improving Automatic Repeat Request (ARQ) Performance on Satellite Channels Under High Error Rate Conditions, IEEE Trans Comm, April 1977, pp. 436-439. 6. Joel M Morries, On Another Go-Back-N ARQ Technique For High Error Rate Conditions, IEEE Trans Comm, Vol 26, No. 1, Jan. 1978, pp. 186-189. 7. E J Weldon Jr, An Improved Selective Repeat ARQ Strategy, IEEE Trans Comm, Vol 30, No. 3, March 1982, pp. 480-486. 8. Don Towsley, The Shutter Go Back-N ARQ Protocol, IEEE Trans Comm, Vol 27, No 6, June 1979, pp. 869-875. 9. Shyam S. Chakraborty et al., An ARQ Scheme with Packet Combining, IEEE Comm Lettters, Vol. 2, No. 7, July 1995, pp. 200-202. 10. H O Burton and D D Sullivan, Errors and Error Control, Proc IEEE, Vol. 60, Nov. 1972, pp. 12931303. 11. Yu Dong Yao, An Effective Go-Back-N ARQ Scheme for Variable Error Rate Channels, Vol. 43, No. 1, Jan. 1995, pp. 20-23. 12. N D Birrell, Pre-emptive retransmission for communication over noisy channels, IEE Proc Part F, Vol. 128, 1981, pp. 393-400. 13. H Bruneel and M Moeneclacey, On the throughput performance of some continuous ARQ strategies with repeated transmissions, IEEE Trans Comm, Vol. COM m34, 1986, pp. 244-249. 14. Shyam S Chakraborty et al, An Adaptive ARQ Scheme with Packet Combining for Time Varying Channels, IEEE Comm Letters, Vol. 3, No. 2, Feb. 1999, pp. 52-54. 15. Y Wang and S Lin, A Modified Selective Repeat Type-Ii Hybrid ARQ System and its Performance Analysis, IEEE Trans Comm, Vol. Com 31, May 1983, pp. 593-608. 16. S B Wicker and M J Bartz, Type-II Hybrid ARQ Protocol using Punctured MDS Code, IEEE Trans Comm, Vol. 42, Feb-March-April 1994, pp. 1431-1440. 17. A G Daraiseh and C W Baum, Packet Combining in Frequency Hop Spread Pectrum Communication System, IEEE Trans Comm, Vol. 46, Jan. 1998, pp. 23-33. 18. P S Sindhu, Retransmission Error Control with memory, IEEE Trans Comm, Vol. COM 25, May 1977, pp. 473-479. 19. Shyam S Chakraborty et al., An Exact Analysis of an Adaptive GBN Scheme with Sliding Observation Interval Mechanism, IEEE Comm Letters, Vol. 3, No. 5,May 1999, pp. 151-153. 20. Coates R F W, Modern Communication Systems, Macmillian, London, 1975, pp. 16, 18 and 184185. 21. V Ahuja, Design and Analysis of Computer Communication Network, McGraw Hill. 1982. 22. Guoliang, End-to-End Data Paths: Quickest or More Reliable, IEEE Communications Letters, Vol. 2, No. 6, June 1998, pp. 156-158. 23. G H Chen and Y C Hung, On the quickest path problem, Information Processing Letters, Vol. 46, 1993, pp. 125-128. 24. Y L Chen, An algorithm for finding the k quickest paths in a network, Computers Operation Research, Vol. 20, 1993, pp. 59-65. 25. Michele Zorzi and Ramesh R Rao, Perspectives on the impact of Error Statistics on Protocols for Wireless Networks, IEEE Personal Communications, Oct. 1999, pp. 32-40. 26. Norival R Figueria and Joseph Pasquale, Providing Quality of Service for Wireless Links, IEEE Personal Communications, Oct. 1999, pp. 41-50. 27. M Zorzi and R R Rao, Lateness Probability of a retransmission Scheme for error control on a twostate Markov channel, IEEE Transaction on Communication, Vol. 47, Oct. 1999.

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28. Michael J et al, Communications, IEEE Spectrum, Jan. 2000, pp. 33-43. 29. Jaap C Haartsen et al, The Bluetooth Radio System, IEEE Personal Communication, Feb’2000, Vol. 7, No. 1, pp. 28-36. 30. Report, The Net Effect: More Choices for Voice, Express Computer, June 2000 pp. 19-21. 31. C T Bhunia, Internet2, Electronics for You, New Delhi, Jan. 2000, pp. 39-42. 32. Paul Love, The Internet2 Project, IEEE Communication Magazine, Dec. 1999, Vol. 37, No. 12, pp. 3-4. 33. Richard Comerford, State of the Internet: Roundtable 4.0, IEEE Spectrum, Oct. 1998, pp. 69-75. 34. A Watson and M A Sasse, Multimedia Conferencing via Multicast: Determining the quality of the service required by the end users, Proc. Int’l Workshop Audio Visula Services over Packet Networks, Aberdeen, Scotland, Sept. 1997. 35. Mahbub Hassan et al., Internet Telephony: Services, Technical Challenges, and Products, IEEE Communication Magazine, April 2000, pp. 96-103. 36. Donna Bergmark et al., Building Blocks for IP Telephony, IEEE Communication Magazine, April 2000, pp. 88-94. 37. Guy Thomsen et al., Internet Telephony, IEEE Spectrum, May 2000, pp. 52-58. 38. What Sook Jeon et al., Improved Selective Repeat Request ARQ Scheme for Mobile Multimedia Communications, IEEE Communication Letters, Vol. 4, No. 2, Feb. 2000, pp. 46-48. 39. Tom Hardin et al., Accelerating Viterbi decoder simulations, Electronic Engineering March 1999, UK, pp. 69-76. 40. AJ Viterbi, Convolution Codes and Their Performance in Communications Systems, IEEE Trans on Communication Technology, Vol. Com. 19, No. 5, 1971, pp. 751-771. 41. Y Yasuda et al., High Rate Punctured Convolution Codes for Soft Decision Viterbi Decoding, IEEE Trans on Communication, Vol. Com-32., No. 3, 1984, pp. 315-319. 42. Robert Cottrell et al., The Implementation of a Turbo Codec in a PLD, Electronic Engineering, Jan. 2000, UK, pp. 68. 43. C Berrou et al., Near Shannon Limit Error-Correcting and Decoding Turbo codes, IEEE Intl Conf on Communications (ICCC’93), Geneva, Switzerland, May 1993, pp. 1064-1070. 44. M Boisseau et al., High Speed Networks, John Wiley and Sons, 1995, UK, Ch. 1. 45. Rainer Handel et al., ATM Networks, Addision Wesley, USA, 1999, Ch. 4. 46. Dimirti Bertsekas et al., Data Networks, Prentice Hall of India, 1992, Ch. 2. 47. P E Boudreau et al., Performance of Cyclic Redundancy Check..., IBM Journal on Research and Development, Vol. 38, No. 6, Jan. 1994, pp. 651-657. 48. Shyam S Chakraborty et al., On the Performance of an Adaptive GBN Scheme in a Time Varying Channel, IEEE Communications Letters, Vol. 4, No. 4, April 2000, pp. 143-145. 49. G E Keiser, Local Area Networks, McGrawhill, USA, 1995. 50. Wilfried Gappmair, Claude E Shannon: The 50th Anniversary of Information theory, IEEE Communication Magazine, Vol. 37, No. 4, April 1999, pp. 102-105. 51. M Lentmaier and K Sh Zigangirov, On Generalized Low-Density Parity-Check Codes Based on Hamming Component Codes, IEEE Transactions Letters, Vol. 3, No. 8, August 1999, pp. 248250. 52. Jinhong et al., Combined Turbo Codes and Interleaver Design, IEEE Transactions on Communications, Vol. 47, No. 4, April 1999, pp. 484-492. 53. Omer F Acikel et al., Punctured Turbo-Codes for BPSK/QPSK Channels, IEEE Transactions on Communications, Vol. 47, No. 9, Sept. 1999, pp. 1315-1323.

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APPENDIX-B Modified ARQs and Integrated Solution for Error Control and Security ERROR CONTROL In the network the error in packet is controlled by ARQ techniques. The basic three ARQ techniques are S/W, GBN and S/R. To improve performance, several modifications[1-6] have been suggested in literatures to improve throughput of the BEC techniques. Among several modifications a few important modificiations are: Sastry’s modification, Morris modification, Weldon’s modification, Towley’s modification, Chakraborty’s modification and Yao technique. While the above stated first four modifications are applicable to static error rate channels, the remaining two modifications are applicable to variable error rate channels or dynamic channels. The modifications mentioned above have several issues that need consideration. Sastry’s modification suggested for retransmission of i (i > 1) copies of retransmission of packet that is acknowledged as erroneous. But how to select “i” was not addressed. Shall “i” be constant for all the times of the retransmission requests ? The “i” may be constant for a static channel, but for variable rate error channel, “i” should be variable. We discuss here two techniques [7] of selection of “i” based on the number of times the retransmission is requested for a packet, and thereby suggest the application of the techniques in variable rate error channels. Weldon’s suggestion for a hybrid technique, a combination of EFC and BEC was not so sound considering the low coding efficiency of ECC used in FEC. Chakraborty [6] recently suggested a technique for locating the position of bit(s) in error of the packet, so that the receiver can correct the error rather than requesting transmitter to retransmit the erroneous packet. The technique was named as packet combining scheme. It was so named because, error location(s) is (are) detected by XOR operation of earlier received erroneous packet and requested retransmitted packet. We discuss [8] of multiple combining (MPC) of several retransmitted packets that can be a better solution for correcting error at the receiver. Several important modification we will discuss will open the areas of future research.

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Effects on previous state: In all of the studies and modifications, the retransmission strategies are adopted based on the present status of the acknowledgement. If the current acknowledgement is NAK, it is assumed that the channel is in high error condition; and if the current acknowledgement is ACK, the channel is assumed in low error condition. Accordingly and for example, in Sastry’s modification when a NAK is received; “i” (i > 1) copies of the packet acknowledged in error are retransmitted; and when ACK is received for the kth packet, (k + 1)th packet is sent in a single copy. No technique considers the previous acknowledgements to consider the state of the channels. Therefore, even if consecutive NAKs are received in immediate previous occasions, if now a single ACK is received, the channel is assumed is in low error state. This is logically not so sound. We propose that the state of channel should be decided upon the present and the previous acknowledgements, and surely not only the present single acknowledgement. We find that in the proposed case, the throughput is more than that in Sastry’s technique. We propose that the state of the channel would be decided upon by considering the immediate past two acknowledgements and present acknowledgement; and retransmission strategies would be taken accordingly. The majority rule shall be applied to the three acknowledgement, previous two and present one to decide upon the state of the channels. Thus the probability that the decision of correct state is achieved becomes 0.5 and 0.66 respectively in the Sastry’s case and in the proposed case. This is a clear advantage of the proposed scheme over Sastry’s scheme. We have shown in table (1), our proposition and the Sastry’s technique. In the three cases (bold rows in Table (1), the proposed technique is different from the Sastry’s technique. In the first instant, the number of retransmissions in the proposed and that in Sastry’s technique are respectively 1 and “i”. In the second instant, it is clear that a state of “ACK NAK NAK” can reach only after either a state of “ACK ACK NAK” or a state of “NAK ACK NAK”. In the proposed case therefore, previous transmission is one copy, retransmission will be “i” copies. But in case of Sastry’s technique, there will be 2i copies on retransmission. By then, the proposed technique gets and edge over Sastry’s technique by a factor of “2i–1”, combining the first instant and the second instant together. In the third instant of difference, the copies on retransmission in the proposed technique and in the Sastry’s technique are respectively “i” and 0, giving an edge to Sastry’s technique by a factor of “i” copies. Therefore taken together all three different cases, the proposed technique has advantage of less number of retransmission by an amount of “i”. Under the table (1), if we assume all the states of acknowledgements are equi-probable, the total six packets (Table 2) will be successfully transmitted. For this, under the proposed technique and the Sastry’s technique, the required total copies will be 8 + 5i and 8 + 6i. This will cause the throughput (η) to be proportional to: ηproposed = {6/(8 + 5i)} ηSastry = {6/(8 + 6i)} When minimum i (i > 1) i.e. i = 2 is taken, maximum ηproposed and ηSastry respectively become 33% and 30%. The constant of proportionality will be same in each case being function of P and “a” where P = probability that a packet is in error,“a” = tp/tt; and tp and tt are respectively end-to-end propagation delay and packet transmission delay. Therefore comparison as made is duly acceptable. When “i” increases, the difference between throughput increases, and the proposed technique becomes more and more superior to the Sastry’s technique. All the studies and modification suggested so far can be further studied with the project model of state based on past and present acknowledgement.

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Table 1: Illustration of proposed technique with Sastry’s technique Proposed technique Acknowledgements

Sastry’s technique

State of the channel

Transmitted copies

Retransmitted copies

State of the channel

Transmitted copies

Retransmitted copies

T0--

t0-

t0

ACK

ACK

ACK

L

1

0

L

1

0

ACK

ACK

NAK

L

1

1

H

1

i

ACK

NAK

ACK

L

1

0

L

1

0

ACK

NAK

NAK

H

1

i

H

1

i+i

NAK

ACK

ACK

L

1

0

L

1

0

NAK

ACK

NAK

H

1

I

H

1

i

NAK

NAK

ACK

H

I

0

L

1

0

NAK

NAK

NAK

H

1

I+i

H

1

2i

(T0--, t0- and t0 respectively stand for previous-to-previous, previous and current acknowledgements. H and L respectively stand for high error and low error state) {Bold rows show the difference between proposed and Sastry’s technique Total transmission under proposed technique = 7 + i Total retransmission under proposed technique = 1 + 4i Total transmission under Sastry’s technique = 8 Total retransmission under Sastry’s technique = 6i} Table 2: Illustration of six packet transmission under equi-probable states of three acknowledgements ACK

Pkt-1

ACK

ACK

ACK

ACK

NAK

ACK

NAK

ACK

NAK

ACK

NAK

ACK

NAK

NAK

NAK

NAK

NAK

NAK

NAK

ACK

NAK

ACK

ACK

—

Pkt-5

Pkt-6

Pkt-2

Pkt-3

—

Pkt-4

—

—

(Pkt-i means ith packet is successfully transmitted) MPC with Error Forecasting Decoding: The modified packet combining was discussed earlier. The future work may be carried out with MPC in combination with error

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forecasting technique[9]. The error forecasting decoding has been investigated in correction of burst error. The decoding is properly applicable to the interleaving technique of correction. We propose the application of error forecasting technique in packet combining scheme. The technique is applicable to both random and burst error correction and is illustrated with an example:

CASE OF RANDOM ERROR Original packet number 1 = 01010101 Received packet number 1 = 11010101 (erroneous copy 1) Retransmitted received packet number 1 = 00010101 (erroneous copy 2) 1. Apply original packet combining technique to correct error 2. Keep track of the error location for correction of the next packet if required. Original packet number 2 = 00110011 Received packet number 2 = 01110011 3. Apply error forecasting technique to correct error in the received erroneous second packet. Assume error is in the same location as in the first packet, and change bit from 0 to 1 or vice versa. See correction is achieved or not. If correction is made, stop; and be ready to receive third packet. If correction is not achieved as per forecasting technique repeat the process for neighboring bits. For example if the error bit location in the first packet is first bit from left, check for second from left. If it second from left, check for first and third from left.

Case of Burst Error In case of burst error, the interleaving techinque be used. Assume a (4, 5, 4) interleaving packets for the purpose of illustration in table (3) where only simple parity is applied in packet of one byte. Assume error locations on transmission as in table (4). Then we propse to apply the packet combining and forecasting as in table (5) for correction. Table 3: Original interleaved packets 01010101 11111111 00010001 10101010

Table 4: Burst error position assumed at location x 01xx0101 111xxx11 0xx10001 101xx010 Use parity on column

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Table 5: Application MPC and Forecasting 0101

No error

11x0

Error detected. Call for retransmission apply MPC

x1x1

Apply forecasting in all other erroneous packets

Xx1x 0x0x 1x00 0101 1110 0000

The advantages in hybrid application of MPC and Forecasting decoding lies in decreasing retransmission delay, but at the cost of higher decoding complexity which remains a scope of further research. Forecasting ARQ: In all of the studies and modifications, the retransmission strategies are adopted based on the present status of the acknowledgement. This looks sound so long the error rate is low and static. But the error rate of the wireless link and the broadband networks is high. Therefore there is need to adopt forecasting technique for retransmission so that probability of successful transmission is improved. We propose that the state of the channel would be decided upon by considering the immediate acknowledgement rather than the acknowledgement due from the receiver unlike the conventional ARQ. We have illustrated in table (VI) the proposed forcasting strategy. Illustration is given for stop and wait ARQ strategy, and could be extended to other ARQ strategies. In the proposed technique, as and when NAK is received, two copies of packet in error have been proposed to be transmitted. Thereafter even if ACK is received, two copies of the next packet are proposed to be transmitted. This strategy has been adopted considering the past NAK and forecasting the high error rate channel. When two consecutive ACKs are received, forecasting is made that the channel is in low error rate, and accordingly the conventional ARQ be started. The proposed method has been illustrated with the inclusion of Sastry’s modification and modification proposed by the author in the earlier works. The proposed technique may be compared with the conventional technique in term of the average number of the packets (N) required to be transmitted per successful transmission that is for conventional and proposed scheme is respectively: Nc = 1/(1 – P) Np = 1/{1 – (P + P2)/2} + 0.25/(1 – P2) The proposed technique will be better than the conventional technique when Np < Nc. It is found that proposed technique has wider benefit at the higher error state. Several modifications of ARQ techniques to combat higher error state of the broadband and wireless networks have been investigated in the literatures. The proposed technique has the viable application, and needs further investigation under real time simulation studies. This remains another scope of future work.

and

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Table 6: Illustration proposed forecasting technique Step

Transmission Strategy

Acknowledgement Received

Forecasting Remark

1

Transmit packet-1 (single copy)

ACK

Low error state

2

Transmit packet-2 (single copy)

NAK

Higher error state

3

Transmit packet-2 (two copies)

ACK

Higher error state

4

Transmit packet-3 (two copies)

ACK

Low error state

5

Transmit packet-4 (single copy)

NAK

Higher error state

6

Transmit packet-4 (two copies)

NAK

Higher error state

7

Transmit packet-4 (four copies)

ACK

Higher error state

8

Transmit packet-5 (four copies)

ACK

Higher error state

9

Transmit packet-6 (two copies)

ACK

Higher error state

10

Transmit packet-7 (single copy)

NAK

Higher error state

11

Transmit packet-7 (two copies)

ACK

Higher error state

12

Transmit packet-8 (two copies)

ACK

Low error state

13

Transmit packet-9 (single copy)

ACK

Low error state

TRUNCATED PACKET GBN The throughput of GBN depends primarily on the packet error probability, P. The low value of P provides higher throughput and vice versa. In the multi copy (m > 1) scheme, the probability that all the m packets received in error is reduced to Pm (< P). This is how the throughput increases in the multicopy mode. The enhancement is major when the channel is H-channel. We propose that instead of multicopy transmission, the packets may be truncated into parts. This will reduce the value of P itself, resulting enhanced throughput with basic (single copy transmission) GBN. The value P of depends both on the bit error rate, α of the channel and the size of the packet, k. Their relation is: P = 1 – (1 – α)k ...(1) For a given α, the value of P will decrease with decrease of the size of k. The original packet of size k may be reduced to lower size with truncated packet.

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Analysis The throughputs of basic GBN and m-copy GBN (N ≥ m) are respectively given as: η1 = and

ηm =

1– P 1 + (N – 1)P

...(2)

1 – Pm

...(3)

m + (N – 1)P m

We assume that if the transmitter receives β1 and β2 contiguous ACKs and NACKs respectively, it takes the channel as L-channel and H-channel respectively. We further assume that the transmitter acts as basic GBN when the channel is L-channel and as multicopy (m copies) transmitter when channel is H-channel, in compliance with earlier studies on adaptive GBN. The transition from good state (L-mode) to bad state (H-mode) occurs as in [6] at the crossover packet error probability of Pc such that P1 < Pc < P2 where P1 and P2 are the packet error probabilities respectively for good state and bad state. In this model, the throughput of the so far and earlier studied adaptive GBN is given as: ηadap = P1 . η1 + Ph . ηm ...(4) In our proposed scheme of the truncated packet transmission, we assume that the transmitter will transmit in the basic GBN mode when the channel is L-channel and in the truncated mode (with packet truncated into n parts each of k/n size when original packet size is k) when the channel is H-channel. Thus the throughput will be: ...(5) ηtrunc = P1 . η1 + Ph . η1(k/n) where P1 and Ph are respectively the probabilities that the channel is L-channel and the Hchannel [14]; and η1(k/n) is the throughput of basic GBN with the original packet truncated into n parts each of k/n size. Using equations (1-5), we find that: ηadap = P1 . and

ηtrunc = P1 .

(1 – α 0 ) k

1 + (N – 1)(1 – (1 – α 0 ) k ) (1 – α 0 ) k

1 + (N – 1)(1 – (1 – α 0 ) k )

+ Ph . + Ph .

e

1 – 1 – (1 – α) k

j

2

m + (N – 1)(1 – (1 – α) k ) 2 (1 – α) k / n

n + (N – 1) (1 – (1 – α) k/n )

...(6)

...(7)

The bit error rates for the good state and the bad state are respectively α0 and α, (α0 < α). The factor n in the denominator of the right hand side of equation (7) is due to transmission of n parts of the original erroneous packet, each part as a packet of k/n bits in the H-channel. For simplicity, in all above derivations, we have assumed that the feedback path is error free. Thus the gain in throughput of the proposed scheme over the existing adaptive scheme is.

LM (1 – α) MN n + (N – 1)(b1 – (1 – αg k/ n

(Gain/Ph) =

k/ n

)

–

OP m + (N – 1)(b1 – (1 – α g ) PQ 1 – (1 – (1 – α) k ) 2

k 2

...(8)

Numericals Results The gain for a set of α and k is listed in table 7. The gain (positive only that measures the actual gain) is shown by bold data in the table. The bold figures indicate the superiority of the proposed technique over the existing technique. It is found that (1) for each k, the gain is positive only when the bit error rate (at bad state) is greater that some threshold, (it is

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conclusively established that the gain is actually achieved in the proposed technique as the proposed technique is to provide benefit at the bad state when the bit error rate is high corresponding to packet error probability of 0.1 to 1 [5]) (2) the threshold value of bit error rate decreases as k increases increasing operative zone of the proposed scheme, (3) the gain in each case increases as bit error rate decreases and becomes maximum at the threshold, and (4) gain becomes zero at higher values of bit error rate and higher values for k, due to the fact that at these higher values, the throughput for the both proposed and the existing scheme is nearly zero. Table 7 (Gain/Ph) versus Bit Error Rate for m = n = 2 and N = 7 Bit Error Rate

K = 64

K = 128

K = 256

K = 512

0.5

0

0

0

0

0

0

0.1

0.00411

0.000147

0

0

0

0

0.022866

0.008553

0.000704

0.000004

0

0.004185

0.024609

0.013575

0.002021

0.02

0.016062

K = 1024

0.004

– 0.08812

– 0.05237

0.0008

– 0.04208

– 0.06842

– 0.08848

– 0.06934

– 0.01197

0.00016

– 0.00985

– 0.01895

– 0.03502

– 0.05950

– 0.08453

K = 2048

0.022802 – 0.08145

Table 8 lists the gain for different sets of n and m. It is observed for fixed m, as n increases the gain decreases; whereas for fixed n, as m increases the gain increases. This ensures that (1) theoretically the highest benefit the proposed scheme will provide, when n = 2 and m = ∞ and (2) the best choice for n is n = 2. Table 9 lists the gain for different values of window size, N = 7 and N = 10. It is observed that as N increases, the gain increases. Thus the proposed scheme may be used with higher window size. Table 8 (Gain/Ph) for different sets of n and m when k = 64. Bit Error Rate 0.004

M

n

Gain/Ph

3

2

0.205199

3

0.042469

4

– 0.041827

2

0.271808

3

0.109078

4

0.024782

2

0.167551

3

0.001682

4

– 0.081850

2

0.249950

3

0.084087

4

0.000555

4

0.0008

3

4

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Table 9 (Gain/Ph) for different values of N when k = 64, Bit Error Rate = 0.004 m

n

Gain/Ph with N = 7

Gain/Ph with N = 10

3

2

0.205199

0.215030

0.298091

3

0.042469

0.053898

0.175684

4

– 0.041827

– 0.030537

0.089481

2

0.271808

0.276492

0.311060

3

0.109078

0.115360

0.188653

4

0.024782

0.30927

0.102450

4

Gain/Ph with N = 100

The truncated packets increase overhead bits. In correct analysis the increased overhead bits in truncated packets is not the limitation if compared with the conventional adaptive GBN. if 1 is the overhead bits per packet, in the multicopy transmission with m copies , the total number of overhead bits is ml. In the truncated packets with n smaller packets, the same number is nl. As such the whole comparison lies with m and n only. Appropriate Selection of Design Parameters For efficient operation of adaptive GBN, the choice for β1 and β2 is crucial. This is because their low values will result poor performance (due to frequent, may be unnecessary, switching between L-channel and H-channel modes) and high values will fail to comply with fast varying error rate of the channel. The problem was addressed in [10]. However in all previous adaptive GBN schemes including that in [10], the optimal choice for β1 and β2 once calculated by the technique proposed in the schemes, is fixed. But the fixed β1 and β2, even if accurate optimally chosen, will truly never meet the requirement of a time varying error rate channel. For the meeting of the requirement, the optimal choices for β1 and β2 must change with channel condition. Once of the possible solutions for meeting the requirement may be to choose the time varying optimal values for β1 and β2 based on immediate previous L records of number of contiguous received ACKs and NACKs and to use weighted mean to find the optimal values for β1 and β2. For example if L = 7, and immediate past 7 records for number of contiguous ACKs and NACKs are 7, 5, 3, 5, 4, 2, 7 and 14, 9, 9, 10, 11, 12, 8 respectively, the current choice of β1 and β2 will be respectively 5 and 10. The benefit of the proposed idea is illustrated in table X with L = 5. The benefits of the proposed technique are highlighted in the “note” of the table. It is evident that the choice under the proposed scheme better coincides with the actual data. The improvement may further increase with higher size of L Many predictive alogrithms exists in literatures. We have illustrated a simplest one to justify the requirement of the time variant selection unlike previous schemes where selection is time independent fixed one.

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Table 10: Comparison of the proposed technique with previous schemes Actual

Optimal by any earlier method (fixed choice), say

Proposed scheme with L=5

Difference between actual and optimal choice as per previous studies

Difference between actual and proposed scheme

β2

β1

β2

β1

β2

β1

β2

β1

β2

β1

5

19

5

18

5

18

0

1

0

1

7

22

5

18

5

19

2

4

2

3

2

30

5

18

6

21

3

12

4

9

5

40

5

18

5

24

0

22

0

16

3

10

5

18

5

28

2

8

2

18

4

22

5

18

4

24

1

4

0

2

6

24

5

18

4

25

1

6

2

1

8

16

5

18

4

25

3

2

4

9

4

15

5

18

5

22

1

3

1

7

3

3

5

18

5

17

2

15

2

14

2

16

5

18

5

16

3

2

3

0

4

29

5

18

5

15

1

11

1

14

In two cases there is no difference between the optimal choice of β1 and the actual β2. There are three cases where there is no difference between actual β2 and proposed choice of β2. There is one case where there is no difference between actual β1 and proposed choice of β1. There is no case where there is no difference between actual β1 and optimal choice of β1. The cases of no difference is most desirable for adaptive GBN.

Analysis In [5], it was shown that the adaptive GBN under their proposal (which has been also been assumed here) will be valid only when: 1 / γ >> β 2 1 / δ >> β 1

UV W

...(9)

where γ and δ are respectively transition probability from good to bad state and that from bad to good state. They are given as: γ = P1β2 ...(10) and δ = (1 – P2)β1 ...(11) Using equations (9–11), we find that: β2 . P1β2 << 1 ...(12) β1 and β1 . (1 – P2) << 1 ...(13) Any choice for β2 and β1 must meet with conditions (12–13) for adaptive GBN to be valid. At the same time it is clear that any arbitrary choice of β1 and β2; and even if any optimal choice may not necessarily satisfy the conditions (2-13). Table 11 presents the positions for P1 = 0.01 and for P2 from 0.5 to 0.125 for different choices of β1 and β2. It is noted that the conditions (12–13) are not met at P2 = 0.125 in direct contrast to the study of [5] that P2 may be from

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0.1 to 1. This is because of the choice of β1. As the value of P2 decreases the condition (13) shall be met with higher value of β that may as high as 80 or more. However the choice of α is easy, as for any P1 and integer β2 ( > 1) will meet the condition (12). Table 11 Checking of conditions (12-13) with P1 = 0.01

β2

β1

P2

β2 . P1β2

Condition (4) meets ?

β1 . (1 – P2)β1

Condition (5) meets ?

4

12

0.5

0

Yes

0.002930

Yes

4

12

0.25

0

Yes

0.380116

Marginally

4

12

6

8

0.125

0

Yes

2.417007

No

0.5

0

Yes

0.031250

Yes

6

8

0.25

0

Yes

0.800903

Marginally

6

8

0.125

0

Yes

2.748871

No

6

10

0.5

0

Yes

0.009766

Yes

6

10

0.25

0

Yes

0.563135

Marginally

6

10

0.125

0

Yes

2.630750

No

6

12

0.5

0

Yes

0.002930

Yes

6

12

0.25

0

Yes

0.380116

Marginally

6

12

0.125

0

Yes

2.417007

No

6

40

0.5

0

Yes

0

Yes

6

40

0.25

0

Yes

0.000402

Yes

6

40

0.125

0

Yes

0.191594

Marginally

6

80

0.5

0

Yes

0

Yes

6

80

0.25

0

Yes

0

Yes

6

80

0.125

0

Yes

0.001835

Yes

Hybrid GBN In all previous studies it was assumed that the transition from good state (L-mode) to bad state (H-mode) occurs as in [5] at the crossover packet error probability of Pc such that P1 < Pc < P2 where P1 and P2 are the packet error probabilities respectively for good state and bad state. Therefore Pc is a point, which does not really resembles the real behavior of the time varying channel. We propose that when the channel is in P1 (corresponidng to BER ≤ α0), the channel is good channel when the channel is in P2 (corresponding to BER ≥ α2), the channel is bad channel; and in between these two extremes the channel in the middle level (M-channel). Therefore in conformity with the previous studies we propose that at the good state, the channel may use conventional GBN, in the bad state the channel may use the truncated scheme and in the middle state it may use the multicopy GBN. The proposed scheme may be called hybrid GBN that uses advantages of earlier studied all GBN schemes. Numerical Results We assume that if the transmitter receives β1, β2 and β3 contiguous ACKs, NACKs and NACKs respectively, it takes the channel to L-channel from M/H state, to M-channel (for analysis purpose we assume a single value of BER for M-state as α1) from L state and to H-channel

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from M state, respectively. The selection values of β may be made as per idea II. The transmitter acts as basic GBN when the channel is L-channel, as multicopy (m copies) transmitter when channel is M-channel and as truncated GBN when the channel is H-channel, in compliance with the proposed hybrid GBN. We have deduced earlier that: ηadap(basic) = P1 .

1 + ( N – 1)(1 – (1 – α 0 ) k )

ηtrunc(basic) = P1 . and

(1 – α 0 ) k

ηproposed = P1 .

(1 – α 0 ) k

1 + (N – 1)(1 – (1 – α 0 ) k ) (1 – α 0 ) k

1 + (N – 1)(1 – (1 – α 0 ) k )

1 – (1 – (1 – α 1 ) k ) 2

+ Ph .

m + ( N – 1)(1 – (1 – α 1 ) k ) 2

+ Ph . + Pm .

(1 – α 1 ) k / n

m + (N – 1)(1 – (1 – α 1 ) k / n )

...(14) ...(15)

1 – (1 – (1 – α 1 ) k ) 2

m + (N – 1)(1 – (1 – α 1 ) k ) 2

+ Ph.

(1 – α 2 ) k / n

n + (N – 1)(1 – (1 – α 2 ) k / n )

...(16) where P1, Pm and Ph are steady-state probabilities of the channel at L-state, M-state and Hstate respectively. For equations (5 and 6) [10]: pgb (transition probability from good to bad state = P1β1 pbg (transition probability from bad to good) = (1 – P2)β2 P1 = pbg/(pbg + pgb) and Ph = pgb/(pbg + pgb) For equation (7) we have: pgm (transition probability from good to middle state) = P1β1 pmb (transition probability from middle to bad state) = (1 – Pc)β2 pbg (transition probability from bad to good) = (1 – P2)β3 P1 = (pmb + pgb)/p; Pm = (pgm + pbg)/p and Ph = (pgm + pmb)/p where p = (pgm + pmb + pbg) The throughput obtained as per above equations (14–16) for a few sets of different parameters is shown in table 12 with β3 = 4, β2 = 20, β1 = 40 and m = n = 2. Table 12: Throughput comparison between different GBN schemes Throughput Multicopy

Truncated

Proposed

Set 1

N = 7, α0 = 0.01; α1 = 0.25; α2 = 0.5

K=8

0.630

0.630

0.655

K = 16

0.449

0.449

0.452

Set 2

N = 10, α0 = 0.01; α1 = 0.25; α2 = 0.5

K = 128

0.0288

Set 3

N = 20, α0 = 0.01; α1 = 0.25; α2 = 0.5

K = 128

0.0147

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Set 4

N = 20, α0 = 0.001; α1 = 0.25; α2 = 0.5

K = 128

0.267860

0.267860

0.267888

K = 512

0.067786

0.067786

0.068650

Set 5

N = 7, α0 = 0.01; α1 = 0.01; α2 = 0.5

K = 128

0.014695

Set 6

N = 20, α0 = 0.001; α1 = 0.01; α2 = 0.5

K = 128

0.2358

0.2358

0.3076

K = 512

0.0675

0.0675

0.0688

Set 7

N = 20, α0 = 0.001; α1 = 0.3; α2 = 0.5

K = 128

0.267860

0.267860

0.267888

K = 512

0.067786

0.067786

0.068650

0.014695

0.016468

In table 12 we find that for wide sets of parameters, the throughput of the proposed scheme is better than those of the multicopy GBN and the truncated GBN. The results also confirms the superiority of the truncated GBN over multicopy GBN as claimed in [8]. Although in table 1 we find, the same value of throughput for both the multicopy GBN and the truncated GBN, but this value is obtained in the truncated scheme at the higher BER. Selective ARQ In order to tackle the variable error rate channels, in all the schemes considered so far in the literatures proposed the modification of the basic technique as the error rate of the link increases. No investigation has been proposed for modification as the error rate of the link decreases. We propose that there should be a strategy for modification of the basic ARQ schemes as the link transits towards good/very good state. In the previously modified GBN schemes it is assumed that as the contiguous β1 ACKs are received the channel is modeled as L channel or good channel. As the value β1 is variable, there is no reasonable correctness to apply basic ARQ techniques for correcting error for any value β1 greater or equal to a pre assigned value, say A. We propose that when β1 is greater than any design parameter, B such that (B – A) ≥ A, not all the packets be sent under any ARQ technique but only a fraction of packets. The ACK/ NACKS received from packets sent under ARQ scheme may be used to determine the transmit behaviour of the channel. The scheme proposed may be called selective ARQ that may be used in error control scheme of [2] with modification as shown in Fig. (1).

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Start

Count contiguous NACKs No

Count of contiguous ACKs

Yes

NACK  2 ?

ACK  2A?

Yes Set count = 0

Go for selective ARQ

No

Yes Set count =0

Go for basic GBN

Set count = 0

ACK  A? No

Go for any modified GBN

(Proposed modification with selective ARQ)

Basic ARQ

(1-r) percent of all packets

Mixer r percent of all packets (MODEL OF SELECTIVE ARQ) Fig. 1: Adaptive GBN scheme with selective ARQ

Numerical Analysis In Fig. (1), we have marked the modified part of the full GBN used in error control. Thus comparison of the performance of this marked part will be made with the basic GBN, which in all previous schemes is used when β1 ≥ A. To examine the performance of the proposed selective ARQ, we choose Stop and Wait ARQ for the selective ARQ. This choice will ensure worst performance comparison of the selective scheme with all previous scheme when β1 ≥ A. The stated performance comparison shall be done with two parameters: the average transfer time

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for a success transmission of packet and the throughput that we used in measurement of performance of previous schemes in previous sections (this is actually the inverse of the average number of transmissions required for a successful transmission of a packet). We assume: Transmission time of a packet = 1 (for normalization) Propagation time = a Fraction of packets selected for transmission without any ARQ scheme = r, where (r < 1) Packet error probability when the link is in the state of β1 ≥ B = p Average transfer time = T For the average transfer time for a successful transmission, we have:

1 + 2a (1 – r) 1– p (1 + 2a)(1 – p + Np) = when N < (1 + 2a) N(1 – p) (1 + 2ap) = when N ≥ (1 + 2a) (1 – p)

Tselective = r + Tgbn Tgbn

U| || V| || W

...(17)

Based on the set of equation (17), the results are listed in table 13. It is found that the selective S/W ARQ becomes superior to basic GBN at higher p, particularly when p > 0.5. Thus the proposed technique may be applied at wireless and satellite communication. It is also found that the superiority of the proposed technique becomes more prominent at higher values of propagation delay, a. Selective S/W being superior to basic GBN, selective GBN must be superior to basic GBN. But implementation of selective S/W will be much easier. For the performance in terms of throughput, we have:

ηselective = r + (1 – r)(1 – p) (1 – p) η gbn = 1 + (N – 1) p

U| V| W

....(18)

The Fig. (2) shows the results of set of equation (18). These results also show the superiority of the proposed selective Stop and Wait ARQ over basic GBN ARQ. Table 13: Average time for successful transmission of a packet of selective stop and wait ARQ and GBN a 2

0.5

1

r 0.25

0.5

0.5

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p

Selective S/W

GBN

0.5

7.50

6.00

0.05

4.19

1.26

0.005

4.01

1.025

0.5

2.50

3.00

0.05

1.55

1.10

0.005

1.50

1.10

0.5

3.50

4.00

Remark

Gain = 0.50

Gain = 0.50

ADVANCED ERROR CONTROL TECHNIEQUES IN NETWORK

1.5

0.5

2

0.5

0.5

0.75

1

0.75

1.5

0.75

2

0.75

0.05

2.08

1.16

0.005

2.00

1.01

0.5

4.50

5.00

0.05

2.60

1.21

0.005

2.51

1.02

0.5

5.00

6.00

0.05

3.13

1.26

0.005

3.01

1.02

0.5

1.75

3.00

0.05

1.27

1.10

0.005

1.25

1.01

0.5

2.25

4.00

0.05

1.53

1.15

0.005

1.50

1.01

0.5

2.75

5.00

0.05

1.80

1.21

0.005

1.75

1.02

0.5

3.25

6.00

0.05

2.06

1.26

0.005

2.00

1.02

Gain = 0.50

Gain = 1.00

Gain 1.25

Gain = 1.75

Gain = 2.25

Gain = 2.75

Throughput as inverse of average number of transmission per success 1.2

Selective with r = 0.75

Throughput

1 0.8 Selective with r = 0.25 0.6

0.4

Selective with r = 0.1

GBN with N = 7

0.2

0

0.5

0.05

0.005

0.0005

Probability of packet error

Fig. 2: Throughput of the selective stop and wait ARQ and the basic GBN

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Solution

Problem

Solution

Problem

Solution

Problem

Future research of immense importance A few basic requirements of the information networks are the accurate or reliable (errorfree) transport, the secured transport and the high data rate transport of information. To optimally meet with these requirements, sometimes conflicting in nature is a challenge for network and communication engineers. The problems faced with meeting the stated requirements are increasingly being compounded with the increasing number of users of networks, the increasing volume of traffic of networks and the increasingly wider applications of networks day by day. First, due to increasingly wider application of networks in financial transactions, competitive global business process and classified information exchange, the security threats have been increasing exponentially [11–113] and will continue to do so that calls for investigation leading to design and development of improved crypto systems. This motivation is believed to result in developing AES (Advanced Encryption Standard) that is supposed to replace widely used and 3 DES (Triple DES) DES (Data Encryption Standard) in symmetric cryptography due to latter two’s limited level of security [14–15]. But the investigations [16–17] have duly demonstrated the AES’s problem of error or fault propagation in encryption/decryption process as larger than that of DES. The error propagation problem of AES in effect increases the gravity of the problems arisen out of the other two requirements of the networks, namely error control and high data rate. Any measure to tackle the fault or error propagation problem of AES like redundancy based technique or parity bytes operated parity based technique [16] decreases the data rate. The decrease data rate is due to requiring higher time for redundant encryption of redundancy based Security correction technique or processing of redundant check bits of parity based technique. Second, due to increasingly trend for all wireless and broadband networks that provide better and flexible services, the networks links will have higher BER (bit Error Rate). The wireless link may have BER as high as 10–2. It has been investigated in several works [18] that to tackle high BER links, FEC (Forward Error Control) techniques may be used that Error Control may use codes like 1/3 rate turbo codes. FEC uses more check bits resulting in the requirement of increased data rate or bandwidth that is well-understood and long known issue of error detection or correction codes. Third, due to integration of services over networks, high data rate networks are in demand, whereas high data rate increases BER. The stated basic three requirements of information networks are in trade off (Fig. 3) with each other. The issue, therefore remains to find a solution or a set of solutions High Data rate for meeting the stated requirements, such that such solution(s) Fig. 3: Solutions for the requiredoes(do) not generate problem(s) for meeting other requirements of information networks ments. The present discussion will be on a few solutions and are in trade off. viable approaches for the same. Basic Idea Logically the desired single solution for all requirements may be a scheme for applying the solution of one requirement to solve for one or two of the other requirements. It is well understood that the security techniques, namely encryption/decryption and the error control techniques are basically coding techniques. But there is a fundamental different between two

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coding schemes. The encryption/decryption coding does not require any additional redundant bits, whereas the essential requirement of error correction/detection codes is the use of redundant check bits. Thus for these two requirements, namely security and accuracy, the solution for security must not provide any solution for error control but the solution of error control may provide some solution for security, in the direction of which we propose a scheme. Integrating techniques of error control and that of security The desired investigation will be illustrated with the (7, 4) Hamming Error Correction Code for accuracy and the Transposition Codes for security. The (7, 4) Hamming code is a well known one bit error correcting code where the four information bits, d0, d1, d2 and d3 are coded with three check bits say, c0, c1 and c2 to produce 7 bits code word, C as: {C7C6C5C4C3C2C1} = {d3 d2 d1 c2 d0 c1 c0}. The decoding rules use the position of the bits in the code word, C for correction if a bit error occurs in the code. In transposition coding, the order of the bits/character/letter is changed with some sort of permutation. For example “CONTINUEWAR” may be encoded as “OCTNNIEUAWR” under pair wise transposition of characters. The permutation rule is the key for this encryption/ decryption coding. Thus (7, 4) code word may be coded to provide security by applying several permutations to form transposition codes, a few examples of which may be as follows: (1) d3 d2 d1 c1 d0 c2 c0 ................. pair wise transposition of check bits from left (2) d3 d2 d1 c2 d0 c0 c1 ......................pair wise transposition of check bits from right (3) d3 d2 d1 d0 c2 c0 c1 ................. pair wise transposition of first four bits from right (4) d2 d3 c2 d1 d0 c1 c0.................. pair wise transposition of first four bits from left and so on. The substitution codes and the transposition codes are two primitive and simple security codes that may easily be broken eavesdropper for which they are hardly used in today’s networks that need higher level of security. But till date Vernam code that may be seen as a variation of these two simple codes is known as the most powerful code. The requirement of unique separate key for each session in the Vernam code, a time variant key, makes Vernam code a powerful code. All other codes including sophisticated modern cryptosystems use one time key. Consider P sessions are required to transmit a message. Each session takes a time of T seconds for completion on average. Assume key size of N bits. Assume the eavesdropper applies only brute force attacks for getting the session keys. Under on time key: The eavesdropper may try on average 2N – 1 trials over a period of PT seconds. The required time for analysis of a pattern will be than (PT/2N – 1) seconds. Under Vernam time dependent variable key: The eavesdropper has to try on the average 2N –1 trials over a period of single session. T seconds. This is because the key will change from session to session. Thus the required time of analysis will be (T/2N – 1). As (PT/2N – 1) > (T/2N – 1), the attack is more effective in the one time key by an order of P. Thus the Vernam code is more secured over one time key by an order of P, the number of sessions. Thus the proposed integrated solution of applying the Transposition Coding with the Vernam’s time variant key (for each session the transposition rule/key is to change) for security over Error Correction Code of data bits will be worth applying. Only issue needs to be resolved is the key transportation/distribution between communicating parties in each session. Already the quantum tele transportation and security techniques have been demonstrated to provide solution in this era. With the quantum security techniques solving the problems of key distribution, the proposed integrated technique may become viable and potential in future.

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Integrating transposition codes with turbo codes The turbo code has the basic innovation of using one interleaver at the input and one puncture at the output. The proposed integrated solution as explained in the previous section may be applied in turbo code. We may use two permutation keys one at the input interleaver and another at the output puncture to provide two keys transposition code for integrating security code with error correction turbo code. The integrated solution being a combined coding will be fast in operation, as it will save a few cycles of processing. It will also save the hardware implementation cost. Applying of selective encryption for solving the problem of error propagation in aes Recently the selective encryption has become a subject of investigation [19] as it provides a number of advantages in information transportation. In the selective encryption, only a fraction (Fig. 4) of whole message is encrypted. Its advantages that are explored so far are: it reduces processing time and complexity, and it AES Encryption r part provides a new type of system functionality. We will report here another far appealing advantage of selective encryption that is supposed to Mixer Message be logically an integrated solution we have inLink vestigated in the current work. We mentioned 1-r part in previous section that AES suffers from the Fig. 4: Selective Encryption error propagation. The AES encryption is done at several rounds of iteration. Each round of iteration has different input data and different keys. The input data and the keys of different round are all generated from the original source data and source key respectively. Thus the input data and the keys at rounds follow a data path and a key path respectively. Any bit errors and even a single bit error at any round if occurred either at data path or at key path propagates and results in huge errors. The work [16] reported this limitation of AES in their authoritative work. We made a study on the error propagation under AES encryption. Our results are in direct conformity with the study of [16]. This limitation of AES results in low speed encryption, more processing and higher complexity, as because until and unless error free encryption is achieved the transmission of the cipher will be meaningless. Application of the selective encryption can overcome these problems thereby restoring fast encryption and reduced processing and complexity thereby supporting high data rate. For the purpose of tackling error propagation of AES, out of two important techniques, namely redundancy based technique and the byte based parity technique as mentioned earlier, we assume that AES uses redundancy-based technique. Our assumption of redundancy-based technique is realistic and more practical from the point of view of our proposed technique. We propose to suggest a technique that will guarantee the error correction for all error vectors that may generate in the AES encryption process. The redundancy-based technique only can ensure that but not the byte based parity technique. If p is the probability of the failure of encryption due to error, the average number of times encryption requires to be done for a success is 1/(1 – p). If the message is just a block of 128 bits, 1/(1–p) times of encryption are required to fully encrypt the whole message under AES with redundancy-based technique. If selective encryption were used with r fraction (0 < r < 1) of the message, the number of times encryption required could have been r/(1– p). This ensures the reduced complexity and processing and the increased data rate all by 1/r times compared with full encryption with redundancy-based technique of fault/error correction. The scheme of selective encryption with AES in redundancy-based technique logically provides an integrated solution for encryption, error control on the encryption process and fast transport.

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Only issue remains to be settled is the choice of r. The lower value of r will speed up the AES encryption but at the cost of lowering security level (a qualitative illustration is at Fig. 5). The part message encryption opens a scope of attack by “guessing.” The lower security level will not be a linear relation with decrease of the value of r, as because as r decreases larger part of the message will not be encrypted causing “guessing” attack to be effective in some exponential form. A reasonable empirical formula may be: Level of security ∝ rk Or Level of security = brk Or the loss of the security level in the proposed scheme over the full encryption scheme = b(1 – rk) where b and k are system constants. The level of the security in the original full message with AES encryption = b, and k (2 ≤ k) defines the exponential factor of “guessing”. An optimal choice of r may depend on how much loss of security level is acceptable. A reasonable assumption will be 2/3rd of the original security level. The proposed scheme then a choice of r as brk = 2b/3 or r = (2/3)1/k. For k = 2, the chosen r = 0.81. This choice of r will speed up the fully corrected encryption by 1.24 = (= /0.81) times the speed of the fully corrected full message encryption. The choice of r will be definitely grater than 0.5 as 2 ≤ k for assuring reasonable level of security in the proposed scheme. We have discussed the problems of the techniques used separately in error control and security control. To find a solution of the problem so illustrated, some scheme of integrated solution for both the issues are proposed. The proposed schemes are the basic concepts, which need to be dully addressed by further course of research for the purpose of implementing. The proposed schemes are found to be viable and potential in nature. Speed gain curve with a = 1 Speed of encryption

12 10 8 6 4 2 0 .1

.15 .2 .25 .3 .35 .4 .45 .5

.6 .7

.8

.9

1

.8

.9

1

Value of r

Security level curve for b = 1

Security level

1.2 1 0.8 0.6 0.4 0.2 0 .1

.15 .2 .25 .3 .35 .4 .45 .5

.6 .7

Value of r

Fig. 5: Qualitative curve of speed and level of security in the proposed scheme

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REFERENCES 1. A R K Sastry, Improving Automatic Repeat Request (ARQ) Performance on Satellite Channels Under High Error Rate Conditions, IEEE Trans Comm, April 1977, pp. 436-439. 2. Jol M Morries, On Another Go-Back-N ARQ Technique For High Error Rate Conditions, IEEE Trans Comm, Vol. 26, No. 1, Jan. 1978, pp. 186–189. 3. E J Weldon Jr, An Improved Selective Repeat ARQ Strategy, IEEE Trans Comm, Vol. 30, No. 3, March, 1982, pp. 480–486. 4. Don Towsley, The Shutter Go Back-N ARQ Protocol, IEEE Trans Comm, Vol. 27, No. 6, June 1979, pp. 869–875. 5. Shyam S. Chakraborty et al., An ARQ Scheme with Packet Combining, IEEE Comm Letters, Vol. 2, No. 7, July 1995, pp. 200–202. 6. Yu Dong Yao, An Effective Go-Back-N ARQ Scheme with Variable Error Rate Channels, Vol. 43, No. 1, Jan. 1995, pp. 20–23. 7. Chandan T Bhunia, “ARQ-Review and Modifications,” J IETE Tech Review, Vol. 18, pp. 381-401, Sept.-Oct. 2001. 8. Chandan T Bhunia, “ARQ With Multiple Copies on Retransmission...,” Proc Advances on Computer Communication Networks, IIT, Roorkee, pp. 81–87, February 2004. 9. Katsumi Sakakibara, Performance Analysis of the Error-Forecasting Decoding for Interleaved Block Codes on Gilbert-Elliot Channels, IEEE Trans. On Communication, Vol. 48, No. 3, March 2000, pp. 386–395. 10. A Annamalai, Vijay K Bhargava and W S Lu, “On Adaptive Go Back N ARQ Protocol for Variable Error Rate Channels”, IEEE Trans Commun, Vol. 46, pp. 1405–1408, November 1998. 11. Allen Householde et al., “Computer Attack Trends & Challenges”, Internet Security, Security and Privacy, IEEE Computer Society, pp. 5–7, 2002. 12. C T Bhunia, “Data Security Techniques”, CSI Communication, pp. 11-14, July 2000. 13. C. T. Bhunia, “Data Security”, Information Technology, pp. 69–70, Sept. 1997. 14. NIST, “Announcing the ADVANCED ENCRYPTION STANDARD (AES)”, Federal Information Processing Standards Publication, No. 197, 26, Nov. 2001. 15. B Gladman, “A specification for Rijndael, the AES Algorithm”, http://fp.gladman.plus.com/2001. 16. Guido Bertoni et al., “Error analysis and Detection Procedures for a Hardware Implementation of the Advanced Encryption Standard”, IEEE Trans on Computers, Vol. 52, No. 4, pp. 492–504, April 2004. 17. C T Bhunia et al., Project Work on AES Error Propagation at ISM, Deemed University, India, June 2004. 18. Michele Zorzi et al., “Perspectives on the Impact of Error Statistics on Protocols for Wireless Networks”, IEEE Personal Communication, Vol. 6, No. 5, pp. 32–40, Oct. 1999. 19. Tom Lookabaugh et al., “Selective Encryption for Consumer Applications”, IEEE Communication Magazine, Vol. 4.

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4

Data/Network Security Techniques and Approaches

1. INTRODUCTION The success of e-age and particularly that of the e-business and e-commerce will solely depend on how reliably and accurately the flow of data over networks and Internet is done. The flow of data often faces several problems. The gravity of the problems are compounded[1] over the years (Fig. 1), and the trend will continue because of society’s increasingly dependent over networks and Internet. Thus there have emerged needs to protect data, information and information systems including service networks for successful emergence into fully electronic and networked society. The data is often corrupted, fabricated, modified and/or lost. There are several reasons behind this. Based on the reasons, data is protected by different means. Protection mechanisms are classified into three groups: security, accuracy and privacy (SAP)[2-4]. Security refers to the protection of data against intentional modification, loss or damage and fabrication of data, and/or deliberate disclosure of data to unauthorized persons or miscreants. Once the data is out of hands, it may fall into the hands of bad people. They could modify, destroy or forge the data for their benefit or for share amusement. The data security is essential to prevent these. Accuracy refers to the protection of data against loss or modification of data due to channel or system noise. Privacy refers to keeping data inaccessible to outsiders. The password feature in UNIX system is a typical example of keeping data private. Of course, now-a-days the concept of “passfaces” for password is hot topic. In today’s perspectives the objectives of data security are protections against[5,6]: • Interception: In which case the unauthorized person gains access to the system and captures or copies the message or data in the network. The protection against interception is aimed to ensure confidentiality of the data. • Fabrication: In which case the unauthorized person inserts spurious data in a network or adds records in the files. The protection against fabrication refers to maintaining authenticity of data. • Modification: In which case the unauthorized person gets access of the system and tampers or modifies data or message. The protection is required to maintain the integrity of data. • Interruption: In which case, some parts of data or the data as a whole are destroyed. The protection is required to make the data available. • No repudiation: In which case both the sender and the receiver are prevented from denying the data sent and received respectively. The sender can prove that the alleged receiver has actually received the transmitted data as well the receiver can prove that the alleged sender has actually sent the transmitted data. 355

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The second group of security aims to give protection against the threats to computer systems and networks: • access control whereby the hackers or crackers are prevented from using unauthorized access of networks hosts or nodes and whereby software like virus and worms are prevented from damaging the systems. • availability of system whereby some parts of system or the system as a whole is destroyed. The protection is required to make the system available for transportation. The classification of security services and mechanisms based on the requirement of the stated protection is summarized [5, 7-9] in the following table(I). Internet Users Versus Security Attacks

Users in multiple of 10,000/ Attacks in numbers

60000 50000 40000 30000 20000 10000 0 1985

1990

1995

2000

2005

Year

Internet Users

Security Attacks

Fig. 1: Security problems over years.

Table 1: Security services and protection techniques Services

Techniques

Confidentiality

Coding Techniques, Cryptography, Public Key system, Digital signature, Hash Functions, IPsec

Authenticity No repudiation Integrity Access Control

System Passwords, Firewalls

Availability

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The subject of data security was initiated with the objective of “keeping data confidential and integrated”. “Keeping data confidential and integrated” has two broad aspects: prevent eavesdropping to get access of data, and in case, data is stolen to make it difficult to understand the stolen data and modify it. These objectives are met through different approaches of data security. The physical technique of data security is the oldest form of security, and is used in telephone lines. Data shall be safe, if computing equipments and lines are all physically protected. In data communication among computers, however, this technique is hardly used. Instead, logical techniques are employed. These techniques include coding methods, spread spectrum and cipher, encryption or cryptography. While these methods are mainly for keeping data confidential and maintaining integrity; the techniques of the digital signature and message digest are used for authenticity of data and no repudiation. The issue of non-repudiation is tackled by the cryptography too. We shall review all these techniques of data security and report a few developments. The reported developments refer to modified RSA algorithm and a scheme of automatic variable key. The automatic variable key is aimed to vary the key with data that may be hard to crack, as it becomes time variant key. It is well established that the accuracy of data in reference to error control, is challenged by the coding techniques. The security of data too is mainly controlled by the coding techniques. In the wireless and broadband networking, the issue of error control is a big threat. We therefore propose a future research direction on both the error control and the security control by suggesting the technique of integrated coding [10]. The paper highlights a few integrated approaches. The basic science and art of sending data or information securely is known as cryptography. The cryptography is nothing but a sort of coding the data or message in a way that makes unreceiable or unreadable to opponents or enemies. In the cryptography, the original data or message that any sender intends to send is known as plain text. The coded plain text is known as cipher text, and the cipher is obtained by the method of coding called encryption. Thus cipher text is also known as encrypted text. The receiver does deciphering or decryption on the received cipher or encrypted message in order to get back the plain text. The opponent or enemy or eavesdropper aims to break the cipher text to read the original message. The art of breaking the cipher is known as cryptanalysis. For successful cryptanalysis, several attacks are made by the eavesdroppers. The subject of Cryptology is Cryptography + Cryptanalysis. In the section we shall deal with the aspects of cryptology for all existing and proposed techniques. Besides, the issue of network security has been analyzed in the paper in particular reference to the Internet. BOX 1 TERMINOLOGIES OF DATA SECURITY • original data/message, that is meant for transmission, is known as plaintext • plaintext when is coded/enciphered/encrypted by a key what is obtained is called cipher text/encrypted/coded message or data • Cipher text/encrypted/coded message or data is sent over the communication link • Cipher text/encrypted/coded message or data when is deciphered/decoded/decrypted what is received back is called plaintext or original message or data

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Original Message or Data

Key

Encryption / Enciphering / Coding

Transmitter’s End

Cipher Text / Encrypted / Coded Message or Data Link

Possible attacks from eavesdropper

Deciphering / Decryption / Decoding

Receiver’s End

Key

Plain text / Original Message or Data

Fig. 2: Security approaches / techniques

2. DATA OR INFORMATION SECURITY Cryptography The tree of existing different security approaches and techniques is as in Fig. (2): Security Approaches

Data/Information

Physical control/ Rugged

Versus

Computers/Networks

Access control

Logical control

Contemporary methods Message Digest

Conventional methods/Secret key

Public key methods

Classical (1 & 2) (3 & 4) 1. Substitution Coding 2. Transportation Coding 3. Stream ciphers 4. Vernum/DES/AES (Blood Ciphers) 5. Code Block Ciphers

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Firewalls

Secure has function

RSA

Filter based Proxy based

They are known as asymmetric cryptography, and provide Confidential Authentic and nonrepudiation measures

DATA/NETWORK SECURITY TECHNIQUES AND APPROACHES

These are known as Symmetric Key Cryptography They provide confidential security measure

359

They provide cryptography checksums and integrity

Key Distributions is the major problem solutions are

Key distribution centre (Key transport protocol)

Diffie Hellmen protocol (Key Agreement protocol)

Quantum security (Agreement + Transport)

3. CONVENTIONAL ENCRYPTION 3.1 Classical Ciphers The classical ciphers or in general coding techniques are based on the algebraic theory. For purposes of data correction or detection, codes are designed to preserve correctness of data even if error occurs. For data security purposes, codes are designed to provide a cover of confusion so that data become unreceivable to unauthorized person. Two of the simplest and oldest coding schemes are substitution coding and transposition coding.

Substitution Codes In substitution coding[2-13], each character of the original text (often known as plain text) or data is encoded or changed to some other character, say A for X, B for Y and so on. There are many types of substitution coding as explained in Fig. (3). The most well known simple substitution cipher is Morse Codes where each letter is substituted by a series of dots and dashes. In the Caeser Cipher so named as it believed that Julius Caeser used it with a shift factor 3 (or key =3), each letter of the plain text is displaced by a fixed value. The displacement value is the key. If the key is 2, for a plain text “Continue War”, the cipher will be “Eqpvkpwg Yct”. Mathematically, Caesar cipher is expressed as: C = P + K (mod 26) where C, P and K are respectively cipher letter, original letter and key (shift/displacement) value. A generalization of Caeser cipher is known as affine cipher [12,13] that is mathematically represented as: C = a . P + b (mod 26) where a and b are chosen to make deciphering unique. Although the cipher text looks as unreceivable, yet it is easy to break the code. An improved code is monoalphabetic cipher, in which case the key is not fixed but variable for each letter so long each letter has a unique substitute letter. Fig. (4) gives an illustration of the code. The monoalphabetic cipher is superior to Caeser cipher as because Ceaser cipher is having only 25 possible keys, whereas monoalphabetic cipher has 26 ! or of the order of 1026 possible pairings of letters. The monoalphabetic cipher may be mathematically represented as: C = P + K (mod N) where K and N are variables. The polyalphabetic cipher makes use of Caeser ciphers with two different key values. The final cipher is made of pattern of two cipher that repeats. For two Caeser ciphers C1 and C2 as illustrated in Fig. (5), the pattern may be C1, C2, C2, C1, C2 that repeats for generating the final cipher message. Substitution codes Playfair, Hill, Saint Cipher Morse code Caeser Cipher

Monoalphabetic Cipher

Polyalphabetic Encryption/Vigenerre Ciphers

Fig. 3: Different types of substitution coding.

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Plain Txt Key

C +3

O –2

N +1

T 0

I +4

N –2

U –1

E +5

W +2

A +2

R 0

F

M

O

T

M

L

T

J

Y

C

R

Cipher

Fig. 4: Illustration of monoalphabetic code Original Text

C

O

N

T

I

N

U

E

W

A

R

C1 with key = 5

H

T

S

Y

N

S

Z

J

B

F

W

C2 with key = 2

E

Q

P

V

K

P

W

G

Y

C

T

Pattern

C1

C2

C2

C1

C2

C1

C2

C2

C1

C2

C1

Pattern repetition

First time pattern

Final Cipher

H

Q

P

Second time pattern Y

K

S

W

G

Third time pattern B

C

W

Fig. 5: Illustration of polyalphabetic encryption

The other improved substitution codes are playfair cipher, Hill cipher and Saint cipher. In these ciphers, the substitutions are done on a group of letters such as bigrams (a group of two letters like CO, TI, NU, EW, AR for a plain text of “CONTINUE WAR”) or trigrams (a group of three letters like CON, TIN, UEW, AR for a plain text of “CONTINUE WAR”).

Transposition Codes In transposition coding, the order of the letter is changed with some sort of permutation. For example “CONTINUEWAR” may be encoded as “OCTNNIEUAWR” under a character wise exchange.

3.2 Cryptanalysis of Classical Ciphers Cryptanalysis is the art and science of getting the key, and the encryption and decryption algorithm. This is achieved by different forms of attacks as portrayed in Fig. (6). Attacks are made to capture any, some or all of the key, the plaintext and its cipher, the encryption algorithm, and decryption algorithm. The text attacks are mainly directed towards classical and the general attempts[14, 15] are directed towards Symmetric (Secret or Private Key Cryptography) and Asymmetric cryptography (Public Key Cryptography). Ciphertext only attack The intruder or eavesdropper may only have access to cipher text, say by listening to the link. This is typical attack on message cipher under substitution / transposition coding. Substitution coding is simple and non-standard in nature. Unauthorized persons can figure out the coding scheme if they study the cipher texts with some statistics of English character. Examples are “the” is most used word, the combination of “q” and “u” is unbreakable, quite often two letter words are “in”, “it”, “is” etc., quite often three letters words are “the”, “ing”, “ion” etc.

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Known plaintext attack The intruder or eavesdropper knows the kind of plaintext and has access to record the cipher text. For example, Bob married to Trudy may be in love with Alice. Trudy knows the type of message, Bob possibly is sending to Alice. The type of the message may be like “I love you.” By pairing the letters of plaintext and cipher text, the encryption scheme may be obtained. If somehow trouble-makers get a copy of the plain text and its encoded text even once, the entire encoding scheme will be known to them. These limitations are there in classical ciphers. Chosen plaintext attack In this attack, the eavesdropper is able to choose a plaintext and to get its corresponding cipher text possibly from someone by bribes. Quick breaking is possible. Chosen plaintext and cipher text attack The intruder chooses both plaintext and cipher text, and tries to corresponding other ones by say, bribes in order to break the code. However in view of the limitations as discussed above in refer to classical codes, the classical codes are hardly used in today’s world of the sophisticated and intense attacks. The general forms of attacks are issues that relate mainly to the secret and the public key encryption mainly. Attacks for Cryptanalysis

Text Attacks

General Techniques

Cipher text only attack

Brute force attack (all keys are tried out)

Known plaintext attack

Frequency attack (Cipher text only attack)

Chosen plaintext attack

Differential cryptanalysis (Finding correlation

Chosen plaintext and cipher text attack

(between plaintext pairs & cipher text pairs)

Known to Cryptanalyst 1. Encryption Technique 2. Cipher text to be cracked

1. Encryption Technique 2. Ciphertext to be cracked

1. Encryption Technique 2. Cipher to be cracked

3. Plaintext chosen by cryptanalyst and its corresponding cipher

3. One or more plaintext–cipher text pairs 1. Encryption Technique 2. Cipher to be cracked 3. Purported cipher chosen by cryptanalyst and its decrypted plaintext 4. Plaintext chosen by cryptanalyst with corresponding cipher

Fig. 6: Different forms of attacks for cryptanalysis

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4. GENERAL ATTACKS Cryptanalysis of contemporary security techniques(Secret key and Public key Encryption) Brute force attack The attack attempts to recover the secret key by trying all possible combinations. If the key size is of n bits, there are 2n possible combinations of the keys. On average it requires to try 2n–1 combinations on deciphering. The technique is an exhaustive key search (trying every possible key till the correct one is got) algorithm. The attack may get success, but it is the cost and the time that may make the attack unacceptable. A computer that takes 1 msec to perform a single decryption may take even a few years to recover a key under this search technique; and following is a list of time involved for the same computer in different key sizes: • Key size of 32 bits may take 35.8 minutes • Key of 64 bits may take 292,271 years • Key of 128 may take 5.4 × 1024 years, a time longer than the age of the universe. It was often argued that the best attack on an algorithm is key exhaustion due to the fact that time to break the key increases with the key size. This linearity is absent in any other form of attack. On the other hand this gives an indication of selection of selecting the key size. However the current research[25] clearly demonstrates that there is no proof behind the argument that the best attack is the key exhaustion algorithm. As 2n–1 is the average number of trials required in this attack to break a key, any method that can find key by trials less than 2n–1 is known as short cut attack. The short cut attack if successful on any algorithm, the algorithm is known as break algorithm, even though it may be impracticable to carry 2n–1 operations. The short cut attack is a test on algorithm’s acceptability for encryption. This gives a complete reverse picture that the breaking a key of higher bits may be easier than that of lesser bits. For example, as per[25], an attack if finds that 2200 operations are needed to break the key of 256 bits but requires 2140 operations to break the key of 128 bits; the key of 256 is considered broken whereas the key of 128 bits not, even though attacks of 2200 is impractical and the key of 256 bits is much stronger. Thus the selection of key size is not so straightforward issue, as usually thought of. Frequency Attack It is nothing but cipher only attack. The frequency of letter used in the cipher may be used to find the letter of plain text in comparing the statistical use of letter in message. The attack is directed towards character based message rather than data. Differential cryptanalysis It was in 1991 that Eli Biham and Adi Shamir [25-27] discovered the differential cryptanalysis. The attack works against the secret or private key encryption, the classic example of which is the DES encryption. Actually the attack of this nature is believed to be behind the changes in the S-boxes made in the final version of TripleDES(TDES). The technique behind the attack is to correlate the differences between pairs of plaintexts and that of the corresponding cipher texts.

5. SECRET OR PRIVATE KEY CRYPTOGRAPHY The modern coding techniques are based on one way or unitary functions. Secret key encryption, Public key encryption and Key exchange algorithms techniques employ the one way function.

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A function like y = f(x) is said to be a one way function if : (1) for every x, there is a unique y, (2) given x it is easy to find y, (3) given y, it is quite difficult to find x and (4) f(x) is not equal to f(x/) when x is not equal to x/. Here, x is the original or plain text and y is the coded or encrypted text. A simple analogy is a dictionary, say, from English to Bengali. For a given English word, it will be easy to find its Bengali equivalent using the stated dictionary, but it will be quite hard to find English equivalent of a given Bengali word using the stated dictionary. Substitution or transposition coding is usually character based. In binary data communication with computers, modular arithmetic or bit wise xor ( exclusive or) operation is used as because the alphabets of digital network are bits (0, 1). In the Fig. (1), the classical cipher includes secret key cryptography. However the other way of classification could be to divide present days cryptography, called contemporary cryptography into two categories: (a) Private or Secret Key Cryptography and (b) Public Key Cryptography. Whatever may be the classification, the private key cryptography may be: (1) Stream Ciphers, (2) Block Ciphers, e.g., Vernum Codes, DES, AES., (3) Code-book ciphers.

5.1 Stream Ciphers The cipher is generated by the bit wise XOR[16] between the string of input bits of the plain text or data and kits bits usually generated by a pseudo random generator (Fig. 7). In cryptography application, the conventional random number generators available in programming languages are not suitable as they produce statistical random numbers that may not be so resistive to cryptanalysis to break. The random number generators for cryptography applications are required to be physical processes like noise of active devices, users’ keystrokes etc. Stream cipher is actually block cipher with block of 1 bit. Bit stream of original data/message 011...




Cipher bits 110...

Bit stream of pseudo random generator/key 101...

Fig. 7: Stream Cipher

5.2 Block Ciphers In block ciphers, the original message of m bits is broken into several blocks each of n bits (n < m), and each block so obtained is enciphered or deciphered by the same key, k. In substitution coding, each character is a block. In the transposition coding, each period (period = 2 for pair wise transposition) of characters constitutes a block.

Vernum Codes Vernum showed that if any plain text is made encrypted by a secret key of equal length, it would be difficult to break the encrypted message provided the key is changed with every session. The secret key is nothing but a random number. This is illustrated in Fig. 8. The encryption and the decryption algorithm is bit wise XOR. Both transmitter and receiver can have a prearranged understanding of the secret session key, and hence can use the technique for secure data communication. It operates as below: 1. Assume a secret key(k) of, say n bits 2. Break the message or plain text into block each of n bits. Say the message is made of p and the blocks are m1, m2, … mp,

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3. Generate encrypted blocks as ci = mi ⊕ k, for i = 1 to p. Then transmit the encrypted blocks, 4. Receiver will decrypt the blocks as yi = ci ⊕ k for i = 1 to p. Note that yi = mi for i = 1to p. This is because yi = ci ⊕ k = mi ⊕ k ⊕ k = mi. All the blocks received under deciphering will constitute the plaintext. The technique illustrated above is known as the one time (key is one for all blocks for all sessions) secret key (key must be secret and made known to only transmitter and intended receiver) technique. This technique, however, has one major problem. The secret key would be made known only to the two communicating parties and no one else. If a third party somehow gets a copy of the secret key, the very purpose of coding will be defeated. Shannon proved in his original work[17-19] of 1949 in connecting cryptography with information theory that if Vernum theory is applied, data will be absolutely secured. It is said that in 1967, Fidel Castro of Cuba used the Vernum technique for defense communication. It is believed that the hot line communication between Moscow-Washington was done via Vernum code. For successful communication under this method, the receiver must be informed of the secret key used by the transmitter, every time a block of message is transmitted. The secret key is usually transmitted over conventional channels like the telephone line. Plain text of original message Random number or secret key Encrypted message after XOR

00101101 01010101 01111000

00011111 01010101 01011010

Fig. 8: Illustration of Vernum code

BOX 2 How does Vernum Code Become to Secured with Variable Key ? Consider P number of sessions. Each session takes a time of T seconds for completion on average. Assume key size of N bits. Apply brute force attacks for getting the session keys. Under on time Key The eavesdropper may try on average 2N–1 trials over a period of PT seconds. The required time for analysis of a pattern will be than (PT/2N–1) seconds. Under Vernum variable key The eavesdropper has to try on the average 2N–1 trials over a period of single session, T seconds. This is because the key will change from session to session. Thus the required time of analysis will be (T/2N–1). Now as (PT/2N–1) > (T/2N–1), the attack is more effective in the one time key by an order of P. On the other work, the Vernum code is more secured over one time key by an order of P. See that when P = 1, both are same, which would be the case.

5.3 DES (Data encryption standard) DES belongs to block cipher. Due to industry requirement, different data encryption standards (DES) have been developed. The commonly known DES, Data Encryption Standard was developed by the National Bureau Standards of USA in 1972. It was actually announced as a

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standard algorithm by the NIST (National Institute of Standards and Technology) (NIST’s predecessor was NBS) for Federal Information Processing Standard (FIPS) in USA in the year 1977. DES is meant for cryptography application in all unclassified communications. DES[2024] makes the relationship between the original message and the enciphered message much more complicated. Bits of the original message that go through DES, are shifted, added, multiplied, inverted, etc so that eavesdroppers have to try out an enormous number of possibilities to figure out the code. DES uses the substitution, transposition and XOR operation to provide security. We can recall that earlier we mentioned about this that all new codes make use of previous codes either in this or in that form. The essential functions used in DES are partitioning, iteration, permutation, shifting, selection and XOR operation. The basic operation of DES is shown in Fig. (9) and the enciphering algorithm in Fig. (10). The DES is a block cipher. Each block in DES is of 64 bits or 8 bytes. A plaintext block is encoded with DES 64 bits or 8 bytes key. The 8 bits of the 64 bits key is used as odd parity bits each for a byte of 8 bytes of each block of plain text. This is shown in table (2) where every eighth bit is ignored. Effectively 56 bits are the key size. The DES consists of: (a) two permutations at the first and the last operations of the algorithm, (b) 16 identical round of iterations in between the two permutations-operation at each round is identical taking the output of the previous round as the input of the present round, (c) interchanging of the rightmost 32 bits with left most 32 bits at each round, (d) encryption at each round is done as described below for ith round: Li = Ri–1 Ri = Li–1 ⊕ f(Ri–1, Ki) where function, f is complex function of Ri–1, Ki,(e) the keys for each of the 16 iterations are selected by a process of permutation/contraction as in Fig. (11) in which each key of 48 bits is derived from 56 bits source key, (f) 32 bits swap before last permutation (inverse initial permutation), whereby the most significant 32 bits of the 64-bits output of the last iteration are exchanged with the least significant 32 bits. BOX 3 Solved Problem: The initial permutation table for DES is as below, what will be the final permutation table? Input bit position

1

2

3

4

5

...

60

61

62

63

64

Output bit position

40

8

48

16

56

...

9

49

17

57

25

Answer The final permutation table will be just the inverse of the initial permutation table. So reference to the above table, the final table will look like as below one: Input bit position Output bit position

..

8

9

.

16

17

..

25

..

40

..

48

49

..

56

57

..

2

60

.

4

62

..

64

..

1

..

3

61

.

5

63

..

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Analysis DES initial permutation and final permutation are as follows: Initial permutation Input bit

Final permutation

58 50 42 43 26 18 10

2

60 52 44 36 28 20 12

Output bits

1

7

8

9

Input bits

62 54 46 38 30 22 14

6

64 56 48 40 32 24 16

Output bit

17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32

Input bits

57 49 41 33 25 17

Output bits

33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48

Input bits

61 53 45 37 29 21 13

Output bits

49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64

Input bits

40

8

48 16 56 24 64 32 39

7

Output bits

1

2

3

10 11 12 13 14 15 16

Input bits

38

6

46 14 54 22 62 30 37

Output bits

17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32

Input bits

36

Output bits

33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48

Input bits

34

Output bits

49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64

2

4

2

3

4

4

5

5

6

6

9

7

1

5

8

10 11 12 13 14 15 16

59 51 43 35 27 19 11

63 55 47 39 31 23 15

9

44 12 52 20 60 28 35

42 10 50 18 58 26 33

4

5

3

1

8

3

7

47 15 55 23 63 31

45 13 53 21 61 29

43 11 51 19 59 27

41

9

49 17 57 27

Consider input bit 1 (shown bold in the table). Under initial permutation it is at the output 40th position. So in the final permutation the 40th bit is at the 1st position. Question: Prove that in permutations, the bits of the first byte of the input get spread into 8th bit of each of other bytes of the output. The bits of the second byte of the input get spread into 7th bits of each of other bytes of the output. The process continues at that fashion for all input bits. Original Data Block/Message Block of 64 bits Initial permutation

16 rounds/iterations

Encryption algorithm

32 bit Swap

Key selection

Source ke 64 bits

16 keys each of 48 bits

Inverse initial permutation Output cipher of 64 bits

Fig. 9: Block diagram of DES enciphering and deciphering

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Table 2: DES key permutation to select 56 bits from 64 bits Input bit position

1

2

3

4

5

…...

59

60

61

62

63

Output bit position

8

16

24

56

52

…

17

25

45

37

29

Original Data Block / Message Block of 64 bits Initial permutation 64 bits

L0 = 32 bits

R0 = 32 bits K1

f


Round 1

L1 = R0

R1 = L0
f(R0, K1)

Round i-1

Li + 1

Ri-1 Ki

f


Round i

Round 15

Li = Ri-1

Ri = Li-1
f(Ri-1, K1)

L15

R15 f

K16


L16 = R15

Round 16

R16 = L15
f(R15, K16)

32 bits

32 bits 32 bits swan / Inverse initial 64 bits cipher

Fig. 10: DES enciphering algorithm

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64 bits key Remove 8 parity bits from 8 bytes (Table 2) 56 bits

Ci-1 = 28 bits

Di-1 = 28 bits

Left shift(s) as per table 3 based on round

Left shift(s) as per table 3 based on

Round i

Input to next round as well as for present key generation Selection, contraction and permutation as per Ki Ci

Di

Fig. 11: Illustration of key generation under different rounds

Deriving 48 bits key from 56 bits source The 48 bits key is derived after several steps: (i) every round the 56 bits are divided into two 28 bits parts, and each part is independently rotated left either one or two bits position, depending on the round number. The extend of rotation for each round shall be as per table (3), (ii) The 56 bits obtained after shift as mentioned are used both as input for the next round (i.e., preceding shift is repeated) and to select 48 bits key that will be key for the current round. Table (4) shows how 48 bits are selected from the 56 bits by a process of simultaneous selection and permutation. It is noted that input 8 bits namely, 9, 18, 22, 25, 35, 38, 43, and 54 bits are not selected, to make output key of 48 bits. At each round the selection and permutation is same, but different keys are produced as because of repeated shifting of the keys. Table 3: DES key rotation amount in bits at each round Round Number

1

2

3

4

5

6

7

8

9

10 11 12 13 14 15 16

Rotation amount in bits

1

1

2

2

2

2

2

2

1

2

2

2

2

2

2

1

Table 4: DES Selection, Compression /Permutation to get 48 bits key from 56 bits Input bit position

1

2

3

4

5

6

7

8

10

11

12

13

14

15

16

17

Output bit position

5

24

7

16

6

10

20

18

12

3

15

23

1

9

19

2

Input bit position

19

20

21

23

24

26

27

28

29

30

31

32

33

34

36

37

Output bit position

14

22

11

13

4

17

21

8

47

31

27

48

35

41

46

28

Input bit position

39

40

41

42

44

45

46

47

48

49

50

51

52

53

55

56

Output bit position

39

32

25

44

37

34

43

29

36

38

45

33

26

42

30

40

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BOX 4 Description of Key Generation in Other Way 1. Assume 64 bit original key is made of bits: k1k2k3k4k5 ... k62k63k64 that contains parity bits k8k16 … k56k64 2. Remove 8 parity bits and perform initial permutation (table 2) and divide 56 bits into two 28 bits as follows C0 = k57k49k41k33k25k17k9k1k58k50k42k34k26k18k10k2k59k51k43k35k27k19k11k3k60k52k44k36 (the bits may now bit called c bits) D0 = k63k55k47k39k31k23k15k7k62k54k46k38k30k22k14k6k61k53k45k37k29k21k13k5k28k20k12k4 (the bits now be called d bits) 3. The rounds will have left and right shift. The rounds 1, 2, 9 and 16 have a single bit rotate left; all other rounds have a 2-bits rotate left (Table 3). 4. The 56-bit key at each iteration is generated as: Ki = Ci(left half) and Di(right half). With permutation as shown below (Table 4): Ki = c14c17c11c24c1c5c3c28c15c6c21c10c23c19c12c4c26c8c16c27c20c13c2d41d52d31 d37d47d55d30d40d51d45d33d48d44d49d39d56d34d53d46d42d50d36d29d32 Description of Function, f known as Feistel Cipher As shown in Fig. (10), the function f has two inputs. At ith round the inputs are key Ki of 48 bits and data block of previous round, namely Ri–1 of 32 bits. Data block of 32 bits is expanded to a data block of 48 bits; for the purpose of which the 32 bits are divided into 8 chunks each of 4 bits. The each chunk of 4 bits is expanded to 6 bits by stealing the rightmost bit of from the immediate left chunk and the leftmost bit of the immediate right chunk. This is illustrated in Fig. (12). The expansion is circular in nature in the sense that the first and the last chunk get the stealing bits from each other. The key of 48 bits is now divided into eight chunks each of 6 bits. Each chunk of key is XORed with the corresponding expanded 6 bits chunk of data block. The resulting 6 bits value is sent through a substitution process (S boxes) as in table(V) that reduces chunk to 4 bits. The eight chunks of 4 bits each so obtained constitutes the 32 bits output of the function, f. The decryption in DES is the reverse of encryption, but keys used will be in reverse order, i.e., in the order of K16, K15 … K2 and K1. The K16 will be in first iteration, K15 will be in the second iteration and so on….with received cipher text as input.. Original Data Block of Ri-1 1

2

3

4

First chunk

5

6

7

8

9

10 11 12

25

26

27

28

Seventh Chunk

Second chunk

29

30

31

Eighth chunk

Stealing leftmost bit of immediate right chunk

Expanded first chunk of 6 bits

These bits are stealing bits for seventh chunk

Stealing rightmost bit of left chunk

Fig. 12: Expanded chunks in DES encryption

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Table 5: Substitution of 6 bits expanded chunk to 4 bit output chunk (S boxes) Input

000000

Output

000001

1110 ○

○

○

○

○

000010

0100 ○

○

○

○

○

○

○

○

000011

1101 ○

○

○

○

○

○

○

000100

000101

0010

1111

0001 ○

○

○

○

○

○

○

○

○

○

○

○

○

Input

111010

111011

111100

111101

111110

111111

Ouput

0011

1110

1010

0000

0110

1101

BOX 5 Question: Show that under DES, the S-box Conversion for First 6-bit Chunk will be as follows. Input

Output

Input

Output

Input

Output

Input

Output

000000

1110

010000

0011

100000

0100

110000

1111

000001

0000

010001

1010

100001

1111

110001

0101

000010

0100

010010

1010

100010

0001

110010

1100

000011

1111

010011

0110

100011

1100

110011

1011

000100

1101

010100

0100

100100

1110

110100

1001

000101

0111

010101

1100

100101

1000

110101

0011

000110

0001

010110

1100

100110

1000

110110

0111

000111

0100

010111

1011

100111

0010

110111

1110

001000

0010

011000

0101

101000

1101

111000

0011

001001

1110

011001

1001

101001

0100

111001

1010

001010

1111

011010

1001

101010

0110

111010

1010

001011

0010

011011

0101

101011

1001

111011

0000

001100

1011

011100

0000

101100

0010

111100

0101

001101

1101

011101

0011

101101

0001

111101

0110

001110

1000

011110

0111

101110

1011

111110

0000

001111

0001

011111

1000

101111

0111

111111

1101

6. MODES OF OPERATION OF DES Basic patterns that are used to protect data in block ciphers are called modes of operation. Blocks are made from original data of length equals to block size of an encryption algorithm. In the event of data size being higher than the block size of the encryption algorithm, the data is divided into a number of whole blocks, if required making appended with 0s at the leftmost bit positions of the original data. Then all the blocks are enciphered either in parallel as in ECB mode or in a chin process for example, as in Cipher Block Chaining mode. However, the FIPS (Federal Information Processing Standard of USA) defined four modes[28-30] of DES encryption, namely Electronic Code Block (ECB) mode, Cipher Block Chaining (CBC) mode, Cipher Feedback (CFB) mode and Output Feedback (OFB) mode. The EBC and the CBC modes relevant to DES are illustrated in Fig. (13).The ECB and the CBC are directed towards block

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ciphers whereas the CFB and the OFB are directed towards stream ciphers. A new mode of stream cipher is Counter (CTR) mode that was evolved with AES (Advanced Encryption Standard). The modes of stream ciphers will be discussed with AES.

ECB versus CBC The ECB is the basic operation behind all modes of operation. A block of plain text in this mode is enciphered and deciphered independently, that its enciphering or deciphering does not depend on the other blocks. Thus it has the advantage of enciphering and deciphering all the blocks (if not all for designing problem, but multiple blocks) of a plaintext in parallel, causing operation to speed up. In the ECB, if a same block repeats it will result the same cipher text. So the patterns of the input blocks may be easily understood by the eavesdropper. Thus the mode is having disadvantage of insecurity for many applications. Plain text blocks (Sent) P1 Sender does these functions Encryption algorithm

Cipher text blocks C1 Key Receiver does these functions

Decryption algorithm Plain text blocks (Received) P1

(a) Electronic Code Block (ECB) mode of Block Ciphering: Plain text is a single block [Note: 1. Parallel enciphering/deciphering is possible … Advantage/2. Less secure as same input block always results same cipher text block …Disadvantage] In the CBC mode the cipher text of block, i – 1 is XORed with the plaintext of block, i before it is encrypted. However the first block is XORed with an initialization vector (IV). The vector, IV is a row vector of 64 elements for DES. The vector, IV needs not be a secret[31-33]. In the CBC, as the enciphering of the plaintext of present block is dependent on the cipher text of its predecessor block, the mode does not support parallel operations. This is a disadvantage in comparison with ECB mode. Again a single bit error in a cipher text resulting in communication link, will result corrupting two full blocks of plaintext—the plaintext corresponding to the cipher in error and the next plain text. This is another disadvantage of CBC. But the CBC has the advantage that it hides the input data pattern, in the sense that same input block may not result the same cipher text. Thus it is more secure than ECB mode. In practical cases the CBC mode is widely used in security applications including DES operation.

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INFORMATION TECHNOLOGY, NETWORK AND INTERNET Encryption at the end of sender P1

P2

...........................

IV







......................

K

E

K

E

.........

Pn-1

Pn







E

K

K

E

C1

C2

............................

Cn-1

Cn

C1

C2

..............................

Cn-1

Cn

Ink

.............................. K

D

IV




K

D

.................... K

D




K

D







Pn-1

Pn

................................... P1

P2

............................

Decryption at the end of receiver

Fig. 13: Illustration of ECB and CBC of block ciphers

P1, P2, …. are the blocks of plain text; C1, C2 … are the cipher of blocks of P1, P2 respectively E is the encryption function; D is the decryption function; IV is the Initialization Vector (b) Cipher Block Chaining (CBC) mode of Block Ciphering (Note: Advantage of higher security, but disadvantage of slowness and error propagation) In reference to Fig. (13), it is seen that: Ci = E(Ci – 1 ⊕ Pi) is the equation of enciphering and Ci–1 ⊕ D(Ci) = Ci–1(D(E(Ci–1 ⊕ Pi)) is the equation of deciphering = Ci–1 ⊕ (Ci–1 ⊕ Pi) as D and E are complimentary = Pi that verifies the working principle of CBC.

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This verification leads to suggests[5] that the vector IV be kept secured and be known only to the sender and receiver contrary to the argument given in literatures[31-33] and general belief. If an eavesdropper misleads a receiver with wrong IV vector, the eavesdropper be able to change (invert) selected bits in the first block of plaintext; and the receiver will be unable to notice the change. As per CBC, the receiver computes P1 as: P1 = IV ⊕ D(C1) Thus for a particular bit, say ith bit in 64-bit P1 block: P1[i] = IV[i] ⊕ D(C1[i]) which is equally to state by the properties of XOR that: P1[i’] = IV[i’] ⊕ D(C1[i’]) where i’ is inverse of i. The conclusive evidence that making IV public as suggested in different studies will not hamper the CBC operation is thus not established; and requires further research.

Critically looking in the ECB and CBC The comparative analysis of the ECB and CBC made previously clearly demonstrates that the ECB is superior to the CBC in regard to speed and non-propagation of error. The wide acceptability of CBC over ECB is only due to higher security provided by the CBC. But the higher security of the CBC is associated with keeping two things secret < key and the vector IV; whereas the security of the ECB depends on keeping the key secret only. The vector IV is not a key as such, therefore the burden on keeping it secret is a cost factor and not a security factor. This if logically is considered, the superiority of the CBC to the ECB on security requires investigation. In fact, one of the major consideration of encryption algorithm provides a clear distinction to the EBC, and it is none but the speed of algorithm. If m is the amount of time required in each encryption/decryption, the time required in the ECB for processing a plain text of n blocks is 2m (independent of n), whereas the time required in the CBC mode is approximately 2mn. It is factor of n, by which the ECB has the advantage over the CBC. Thus it is investigated how to remove the disadvantage of the ECB in regard to security threats of repeated patterns so that full advantage of speed may be utilized. A technique known as automatic variable key is illustrated next.

7. AUTOMATIC VARIABLE KEY(AVK) The AVK is illustrated in the table (6) for a session between Alice and Bob whereby they respectively exchange data 345 and 789. The key is now variable and after every transmission it changes dynamically such that: K0 = initial secret data Ki = Ki–1 XOR Di for all i > 0 The variable key as suggested if implemented, the repetition of patterns will not result unlike in the ECB mode. An illustration is given in table (7). The speed will slow down to (m + mn) = m(1 + n) (m time on encryption, but mn time on decryption as next decryption is not possible until and unless present plaintext is obtained-illustration in table 7) but yet it is better than CBC as 2mn > m(1 + n). Further research in experimental simulation studies is required for conclusive establishment of the proposal proposed above.

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Table 6: Illustration of AVK under XOR encryption Session slots

Alice Sends

Bob Receives

Bob Sends

Alice receives

Remarks

1.

A secret key say 2

2

A secret key say 6

6

For next slot, Alice will use 6 as key and Bob 2 as key for transmitting data

2.

Alice sends his first data(3) as: 3 XOR 6

Bob gets back original data (3 XOR 6 XOR 6) = 3

Bob sends Alice gets back first data (7) original data (7 as: 7 XOR 2 XOR 2 XOR 2) =7

Alice will create new key 7 XOR 6 for next slot. Bob will create new key 2 XOR 3 for the purpose of transmission.

3.

Alice sends next data (4) as: 4 XOR 6 XOR 7

Bob recovers original data (4 XOR 6 XOR 7 XOR 6 XOR 7) = 4

Bob sends next data (8) as : 8 XOR 2 XOR 3

Alice computes new key 3 XOR 4 and Bob computes new key 7 XOR 8 for transmitting next data.

Alice recovers original data (8 XOR 2 XOR 3 XOR 2 XOR 3) =8

Table 7: Illustration of Conventional Vs AVK EBC with XOR encryption/decryption Conventional ECB Blocks of plaintext with same pattern

AVK with ECB

Remark

P1 = 1011

P2 = 1011

P1 = 1011

P2 = 1011

Key

1010

1010

1010

0001

First key is XORed with previous plaintext data block to generate key for present block

Cipher text Blocks

0001

0001

0001

1010

As all plain text blocks are available, parallel encryption in both the techniques is possible

Decryption

Repetition in cipher block due to same pattern in the plaintext blocks

No repetition of cipher pattern

Parallel encryption for two cipher blocks is possible.

Parallel deciphering not possible, as for deciphering second cipher block, the first plaintext block is to be recovered for decryption key generation

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8. PROOF OF DES In fact the DES can be seen as 19 steps algorithm (Fig. 14). The first step is fixed transposition, next 16 steps are identical substitution, 18th is the 32 bit swapping that is another simple transposition and last is the reverse of transposition done at the first step. The transposition and the substitution being done in encryption and in decryption by the reverse process will complement each other. Only proof[35] is then is to prove the Feisel cipher that is the function, f involved in the DES. Decryption Direction

64 bits plain text

Encryption Direction

Transportation 1/ Initial permutation

Substitution 1

K1

Substitution 16

K16

Transportation 2/32 bit swap

Transportation 3/ Inverse initial

64 bit cipher text

Fig. 14 : Looking DES in simple form

In order to see how does Feisel cipher works, we write decryption function of DES in reference to Fig. (14): Ri–1 = Li and Li–1 = Ri ⊕ f(Li, Ki) But as Li = Ri–1, the Feisel cipher is invertible in nature. To illustrate assume, Li–1 = 0011, Ri–1 = 1010 and Ki = 1011 Encryption yields the following results: Li = Ri–1 = 1010 Ri = Li–1 ⊕ f(Ri–1, Ki) = 0011 ⊕ (1010 ⊕ 1011) (as f is XOR) = 0010 Decryption yields the following results: Ri–1 = Li = 1010 (see original data is received back) Li–1 = Ri ⊕ (Li ⊕ Ki) = 0010 ⊕ (1010 ⊕ 1011) = 0011 (see original data is received back)

9. MERITS AND DEMERITS /PERFORMANCE ANALYSIS OF DES In general no systematic scheme exists for measuring the performance of any cryptosystem, although several parameters like security, hardware and software suitability, computational

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efficiency(speed), flexibility, and memory requirements are often considered to measure performance. On such general basis, the DES performance is critically analyzed. Implementation The general purpose computers operate on words of 16 1o 64 bits. As DES operates for decryption and decryption on chunks of 4 or 6 bits, the DES algorithm is poorly suited on software implementation as then it will slow down the process. In fact it was designed initially for hardware implementation on which the process is fast. The hardware implementation of DES ensures encryption rate of around several Gbps, whereas software implementation only around a few hundred Mbps. But in today’s perspective, software implementation is very important. It must be mentioned that both hardware and software implementation of DES[3134] have existed for quite long times. About the algorithm The DES has been studied extensively over the years. No fatal weakness has been reported against algorithm. For example the only non linearity present in the algorithm is in S boxes. Therefore the choice of S- boxes was reportedly done to ensure maximum security although there is neither any proof nor any analysis to substantiate this claim. (there is a belief[14] that neither proof nor analysis was given as because the NSA, National Security Agency of USA, kept a security hole in the design; and they did it purposefully to break the system in their needs.). Even then many attempts to find a loop hole in the design have not succeeded so far bringing the confidence and popularity in DES[14]. The NIST stated the goal of DES as “The goal is to completely scramble the data and key so that every bit of the cipher text depends on every bit of the data and every bit of the key…. With a good algorithm, there should be no correlation between the cipher text and either the original data or key.” This goal ensures in the CBC mode of DES discussed earlier. Cost The studies in [31,32] demonstrated that the DES is a low cost encryption. The criticism that the standard is not adequately secured, they believed is not due to technology but on political consideration. Speed of encryption Except S-boxes, the DES algorithm works on linear function of XOR operation. Therefore in general the speed of encryption/decryption is high and higher than other well known public key encryption like RSA (discussed later). IT was earlier reported that the DES encryption/ decryption speed is of few hundred Kbps and a several Mbps respectively for software and hardware implementation. Cryptanalysis of DES Brute Force Attack: With the present day’s computers being thousands of times faster than the computers of 1970s during which the DES was developed, there are several reports [14,33,36,37] of breaking DES using brute force attack in contrary to early works of [38,39] that stated the DES is very hard to break. The practical demonstration of breaking DES by the key exhaustion was demonstrated around 1997-98. It is even argued that the DES, given substantial resources, now be broken even within a few days’ time. The DES is therefore not a force of cryptology to be reckon with.

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Differential Cryptanalysis: The works of Biham and Shamir[26, 27] demonstrated that the differential cryptanalysis of DES. In the work of Agrawal[14], a good example was cited of breaking two rounds DES applying differential cryptanalysis. The example was in general sighted with two plain text blocks having same right halves. We portrayed the problem with an example 4-bit block as in Fig. (15) with simple XOR operation for the function, f. It is found that the XORed of left halves of plaintext is equal to the XORed of ciphers halves at two rounds. The pattern matching of cipher blocks with plain text blocks results in differential attack. It was argued that the strategy be applied to break larger number of rounds in a few seconds. Plain text block 1 Round 0

Plain text block 2

01

01

10

01


Round 1

01

11

K = 11


01

00

...Operation repeats in next round .......................................................................... Round 2

11

10

K = 00

00

01

Fig. 15: An example of differential cryptanalysis for DES

Note:

Rj(i) = right half of ith plain text/ cipher text at jth round Lj(i) = left half of ith plain text/cipher text at jth round

R1(1) ⊕ R1(2) = 11; L0(1) ⊕ L0(2) = 11, R2(1) ⊕ R2(1) = 11; L1(1) ⊕ L1(2) = 11

Although DES has been tested over a long period of time and has proven itself as a sound cryptography, yet its 56-bit key is decisively felt not to be enough to prevent attacks with computers in 1990s. Naturally, as substitute of DES several other secret key systems, namely TDES (Triple DES), IDEA (International Data Encryption Algorithm) and finally AES (Advanced Encryption Standard) have been developed each with longer key size.

10. QUANTIFICATION OF PERFORMANCE Following the failure of DES, there emerges a need to define a quantified performance- measurement technique based on certain parameters and the current pace of the technological progress. Basically with the availability of the secret key encryption, AES and the public key encryption, RSA, it is a reasonable assumption to consider brute force attack as the main form of attack on security (The assumption made in our proposal of selecting the brute force attack under key exhaustion algorithm as the main form of attack is contradictory to the research in [25]. The researcher claimed in [25] that there was no way to prove it. We propose otherwise because the proof to our assumption exists. Till date the reports of breaking DES and RSA as reported in different researches are all due to brute force attack).. Secondly, it will be another reasonable assumption to select the following two as the main parameters of performance measurement: • Security under brute force attack

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• Encryption/Decryption speed under algorithm characteristics, (Here an inherent assumption of software implementation of algorithm is taken)

Quantification of brute force attack Under key exhaustion algorithm, it is certain than the key will be broken, but it is only the time and the cost involved in the exercise that matters. Therefore, over a given time and cost, how a key is secured measures the effectiveness of the algorithm. However, under brute force attack, for a n-bit key size average number of attempts required is 2n–1 for a total key space of 2n. As such over a given time and cost, the key security increase nearly proportional to 2n–1. Hence we define, the probability of key breaking of a cryptosystem with n-bit key as: P(k) = 1/2n–1 ...(1) Equation (1) does not take into account the technological progress. Today even with 56bit key, that is with about P(K) = 0 (What a fantastic figure!), the DES has become forcedobsolete (forced by technological progress) due to the powerful contemporary computers. The famous empirical laws[40-42] that correlate, govern and predict the technological progress and growth of computer and its powers are: # Joy’s law, which states that the computing power, expressed in MIPS (Millions of Instructions Per Second), doubles every 2 years, # Ruge’s law estimates that the communication capacity necessary for each MIPS is 0.31Mbps (Million of Bits Per Second), # Metcalfe’s law which states that if there are ‘n’ computers in a network, the power of the computers in a network like Internet is multiplied by ‘n’ square times, # Moore’s laws state that (a) the number of components on an IC would double every year (this is the original Moore’s law predicted in 1965 for the then next ten years), (b) the doubling of circuit complexity on an IC every 18 months (this is known as revised Moore’s law), (c) the processing power of computer will double every year and a half (Moore’s second law which closely resembles to Joy’s law). # Law of “Price and Power” that states that over the years the computing, processing, storage and speed up power of computers will continue to increase whereas the price of computers will continue to fall. These growth laws have changed the DES fate from P(k) = 0 (full success) in 1970s to P(k) = 1 (full failure) in 1990s. During the gap of about 30 years, the mounted power increase of computers is 264, that gives the breaking power growth rate to 22t, where t is the period in years. These as references (i.e., 64 as the base key length and year 2000 as the base year), the relative probability of subsequent developed cryptosystem with n-bit key where n > 64 in any year y, (y > 2000) may be defined by the following equation: Pr(k) = (263 × 22(y–2000))/2n–1 ...(2) The general form of equation(2) is: Pr(k) = {2rn–1 × 2r2(y–y)}/2n–1 …(3) where n > nr and y > yr; nr and yr are respectively reference base key size and reference base year. For the given nr and n, the equation(3) is valid for only a period, (y – yr) so long Pr(k) <= 1. We will demonstrate this in analyzing, TDES, IDEA and AES. Further based on equation (3), period of survival(POS) of an algorithm be defined as the period, (y – yr) over which Pr(k) <1. Thus POS is obtained by making Pr(k) = 1 in equation(3). If n = ns meets the condition, Pr(k) = 1 we get POS as: POS = (ns – nr)/2 ...(4)

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Quantification of speed of encryption and decryption If the number of operation involved in encryption/decryption process is m, the speed factor, S(k) may be taken as: S(K) = 1/ 2m ...(5) Using the proposed quantification formula, the overall performance be measured by the product of P(k) and S(k) as applicable.

11. TRIPLE DES (TDES)/TDEA/3DES As pointed out earlier, in DES, the key is made up of 56 bits so that about 7.2 × 1016 possibilities exist. This is insufficient to keep key secured under mounted pressure of contemporary computers. To counter this, as a variant of DES, TDES or TDEA (Triple data encryption Algorithm) was proposed[43-45]. The block diagram operation of TDES is shown in Fig. (16). TDES uses three keys (in effect it may be two separate keys at three levels) and three executions of DES algorithm. The important features of TDES are: (1) three distinct keys each of length 56 bits make the effective key size of 168 bits, (2) second stage of encryption/decryption process has no significance (Decryption in place of encryption and vice versa) except that it allows the interoperability between DES with TDES i.e., the users of TDES may decrypt the old DES encrypted cipher: as C = Ek1(Dk1(Ek1(P))) = Ek1(P), (3) use of two keys instead of three keys are allowed by making K1 = K3, in which case the effective key size will be of 112 bits. K1

K2

K3 Encryption Equation C = EK1 (DK2 (EK3 (P)))

P

E

D

E

K3

K2

K1

D

E

D

Link

C

Decryption Equation P = Dk1 (Ek2 (Dk3 (C)))

Fig. 16: Illustration TDES/TDEA

TDES uses the encryption and decryption algorithm of DES, which has been tested over a long period of times of 32 years, without any noticeable attack of cryptanalysis except the brute force attack; although we have shown earlier that the differential cryptanalysis attack on DES is possible. The effect of the brute force attack in TDES has been diminished with 168/ 112-bits key. Thus it has got wide financial applications. But the software implementation sluggishness associated with encryption/decryption of DES remains present in TDES. The 64bit block size is also not desirable for the reason of efficiency and security. Accordingly it is believed that TDES may continue till the AES is deployed on full scale. Of course we will critically examine the in terms of equations(3-5), the cryptosystems later to arrive at a better and decisive conclusion. Question: Why is TDES also called 3DES?

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11.1 IDEA/International Data Encryption Algorithm The IDEA[46] is a secret key block cipher. It was developed with a few aims as: (1) the use of 128-bit key size so that is more resistive to brute force attack than DES, (2) be equally well implemented in both hardware and software unlike DES and TDES. The cryptology IDEA has the features (Fig. 17): • It uses 64 bit block data size like DES/TDES. The block is further subdivided into four blocks each of 16 bits. During encryption, the sub blocks pass through a series of iterations with 8-bits manipulations. • IDEA does not use S boxes in each round, but uses three different invertible operations: XOR, Binary Addition and Binary Multiplication with 16 bits sub blocks. Though these operations are linear, a combination of any two of these results a non-linear operation. Moreover the three linear functions are mixed so that the algorithm has becomes complex that is very difficult to analyze and break. Thus IDEA is more secure against attacks like differential cryptanalysis making a conclusion that IDEA is more secure than DES/IDES. • The 52 sub keys are generated from the 128 bits key source. Each sub key is of 16 bits. Six sub keys are used at each of the iterations (that is each round uses 16 × 6 = 96 bits key). The eight iterations exhaust 48 sub keys. The remaining 4 keys are used in the final transposition round. The 52 sub keys each of 16 bits are generated from 128 bits key as follows: (a) sub keys 1 to 8: From 128 bits, chop of from left 16 bits for each sub keys, (b) Sub keys 9 to 16: these keys are generated at bit 25; (c) Sub keys 17 to 24: these keys are generated at bit 50; (d) the rest of the keys are generated by offset of 25 bits. BOX 6 Illustration of sub keys generation in IDEA: Let the given 128 bits key is; 1100110011001100111100001111000011110000.................0011001100110011001100 Sub key 1

Sub key 2

Sub key 8 Sub key 9

Sub key 25 starts at bit 75 Sub key 33 starts at bit 100 Sub key 42 starts at bit 125 (after 124 + 1) Sub key 50 starts at bit 150 (after 124 + 26) • The multiplication operations are completed by dividing the 32-bits product by (216 + 1). The output is then a 16-bits remainder. • The addition operations ignore carry if generated

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Plain text block of 64 bits k1 – k6

Round/Iteration 1

k7 – k12

Round/Iteration 2

128 bits key source k43 – k48

Round/Iteration 8

k49 – k52

Transposition

Cipher text

(a) Block diagram of IDEA Encryption 16 bits sub block k1

16 bits 16 bits sub block sub block

×

×

16 bits sub block

×

k4

×

k3

k2



k3

× ×

x

16 × 16 multiplier

+

16 × 16 adder




16 × 16 XOR

× k6 ×





Input to second round

(b) First round encryption, Encryption at other rounds is same with different keys as in Fig. (a) and inputs from previous round. Fig. 17: IDEA operations Decryption operation of IDEA is the reverse of encryption with reverse order of key sets.

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12. AES/ADVANCED ENCRYPTION STANDARD Because of drawback mentioned earlier, DES has become obsolete and TDES is not a long standing candidate. The NIST of USA in 1997 announced a proposal[47-53] known as Advanced Encryption Standard as a best candidate for long term replacement of DES. The motivation behind deployment of AES were: 1. to secure strength equal to or greater that TDEA. Accordingly the key options of 128, 196 and 256 bits were proposed 2. the level of security increases in AES not only due increases key size, but due to increases block size of plain text. The block size in basic operation of AES 128 bits. In authentications, when two different messages got the same hash value, the blocks are subjected to collisions. The collisions increases after 2m/2 (m is the block size) authentication fields. In block ciphers, if 2m/2 cipher texts are available to the eavesdropper, the attack becomes easy. Thus higher block size provides higher security confidence. The AES is the block cipher that is having 128, 196 and 256 bits block size unlike 64 bits block size of DES, TDES and IDEA. 3. to achieve improved efficiency. Accordingly, the AES is better targeted towards software implementation under different platforms in order to achieve goal of reducing number of clock cycles required to encrypt a data block[54]. However the hardware implementation of AES coexists under different studies[49,55-56] 4. to achieve higher flexibility. Accordingly, the plain text block size and key size were proposed to be any combination of 128, 196 and 256 bits. But the NIST has restricted plain text block to 128 bits, whereas key size may be any one of the three options. The key operation with 128 bits is defined as basic. It is established that the basic system offers adequate security for civil applications and is a most practical one. 5. The NIST was open to select AES algorithm by an open competition in order to avoid the public suspicion on hidden agenda as in DES. After all AES was proposed to be an unclassified and publicly disclosed cipher to be available free anywhere. Accordingly out of many candidates as many as 16, NIST selected five candidates namely MARS, RC6, Serpent, Twofish and Rijndael. The selection was then based on the performance and the characteristics of the algorithm. The extensive studies and research for the selection were the works of [47,57]. In the work of [57], the speeds of encryption and decryption in Mbps were compared for a 200 MHz Pentium Pro Reference platform. The result of comparison so obtained is partly reproduced in table (VIII). The RC6 algorithm as per the average speed appears winner, but the wide speed difference between encryption and decryption in RC6 among others makes it inferior to Rijndael. In table(IX), the key schedule computations time in clock cycles of Pentium Pro II machine from the same research work is portrayed. The advantage of key schedule computation time for Rijnadael has finally made it the champion. The Rijndael algorithm became winner. The winning the match is not time tested. Brian Gladman [58] in his work argued for multiple winners and final selection based on time tested experience. However the AES Rijndael algorithm has got final node from NIST. The Rijndael AES encryption process is illustrated in Fig. (18) for basic operation with key = 128 bits and Data block = 128 bits.

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Table 8: Speed comparison of AES algorithm competitors (for 128 bits key and 128 bits data block) RC 6

Rijndael

MARS

Twofish

Serpent

Encryption

94.8

68.4

69.4

68.1

26.9

Decryption

113.3

72.7

68.1

68.4

28.0

Average

103.2

70.2

68.7

68.3

27.4

Table 9: The comparison of AES algorithm competitors based on Key computation RC6

Rijndael

MARS

Twofish

Serpent

128 bits key

1632

305 : 1389

4316

8414

2402

192 bits key

1885

277 : 1595

4377

11628

2449

256 bits key

1877

374 : 1960

4340

15457

2345

Plain text block

Output of previous round

Special Round 0

Sub Bytes

K0 Round 1

Key K1

Ten Internal rounds

Shift Rows

Sche duler

K9 Round 9

Mix Columns

K10 Round 10

Add Round Key

Cipher Text

Input to Next round

(a) Block diagram of Rijndael AES Encryption

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(b) Structure of one internal round

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Ki+1

Key Schedule for encryption

Inverse Key Schedule for

Ki+1

Ki

The key for round 0 = Secret Key; During encryption, Ki + 1 = f(Ki); During Decryption Ki = f/(Ki + 1) . The function, f and f/ are invertible. (c) Key generation Fig. 18: Rijndael AES symmetric block cipher

12.1 Rounds of Data Block AES encryption applies a fixed number of iterations / rounds to data block. The number rounds so applied are fixed: 10, 12 and 14 respectively for key of 128, 196 and 256 bits. These are in addition to the special first round called round 0. Each round consists of a fixed sequences of transformations. In basic operation except for the first and last round, for other rounds, known as internal rounds four transformations are done: Sub Bytes Transformation: The data block of 128 bits is divided into 16 bytes, Bi(0 <= i <= 15). The byte sequence is rearranged as a 4 by 4 matrix, S known as state matrix:

Fs Gs S = Gs GH s

0, 0 1, 0

2, 0 3, 0

s0, 1 s1, 1 s2, 1 s3, 1

s0, 2 s1, 2 s2, 2 s3, 2

s0, 3 s1, 3 s2, 3 s3, 3

I JJ JK

where by si, j, the element of ith row and jth column = Bi+4j (i.e., the state, S is organized column wise). The every element, si, j of the state, S is first inverted and is passed through an affine transformation, T : si, j = T( si, j – 1) Due to the inversion, the sub byte transformation is non-linear. The transformation operates independently on each byte, and hence transformations of all sub bytes may be done in parallel. This is how the efficiency of AES is better. Shift Rows: In this operation, the first row is left unchanged; the second, the third and the fourth row are rotated left by one byte position, two bytes position and three bytes position respectively. Shift rows will result the following state matrix, S:

Fs Gs S = Gs GH s

0, 0 1, 1

2, 2 3, 3

s0, 1 s1, 2 s2, 3 s3, 0

s0, 2 s1, 3 s2, 0 s3, 1

The shift rows is a linear transformation.

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Mix Columns: The mix columns is a linear transformation. The elements of the state, S is transformed as below: s0, j = α ⊗ s0, j ⊕ β ⊕ s1, j ⊗ s2, j ⊕ s3, j s1, j = s0, j ⊕ α ⊗ s1, j ⊕ β ⊗ s2, j ⊕ s3, j s2, j = s0, j ⊕ s1, j ⊕ α ⊗ s2, j ⊕ β ⊗ s3, j s3, j = β ⊗ s0, j ⊕ s1, j ⊕ s2, j ⊕ α ⊗ s3, j for every column, 0 <= j <= 3, where α = 02 and β = 03 are fixed coefficients. The transformation is a linear combination of four bytes of the same columns. Add Round Key: The 128 bits key, K is divided into 16 bytes similar to data block. The add round key operation is then performed as below: si, j = si, j ⊕ Ki+4j for 0 <= i, j <= 3 The transformation is equivalent to form a matrix of 16 bytes of key similar to data block, and then adding the two matrices. It is also equivalent to bit wise XOR of 128-bits data block with 128-bits key. The transformation is linear obviously.

12.2 Key Scheduler The key schedule function, f is the combination of basic four transformations: a Right rotation, the Sub bytes operations, and an Addition of a byte constant. The operation that has been described above is for basic operation with 128 bit data block and 128 bits key size. The operation for other options may be generalized for which the works of [56, 57] are quite exhaustive.

13. COMPARISONS OF SECRET KEY CRYPTOSYSTEMS A gross comparison of the private or secret key crypto systems is given in table(X). The critical comparison needs to be done in terms of brute force attack, speed of encryption, period of survival(POS), and collision period. Earlier we defined/introduced these parameters. A critical comparison is given in table(XI). This shows that the AES is the best choice for replacement of DES. Table 10: Gross comparison of different secret crypto systems Secret Key Algorithm

Key size in bits

Number of rounds of operations

Different operations

Major applications

DES

56

16

Shift, XOR, S-boxes

SET, Kerberos

TDES

112 (two keys)

48

Shift, XOR, S-boxes

PGP, Financial applications

168 (three keys) IDEA

128

8

XOR, Addition, Multiplication

PGP

AES

128 (Basic)

10

Sub bytes, Shift rows, Mix columns, Add round key

SET, Financial applications

196

12

256

14

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PGP = Pretty Good Privacy Discussed latter in this chapter), a secure e-mail protocol. Kerberos = an authentication service designed to allow clients to access servers in a secure manner over a network using a KDC (key Distribution center) Table 11: Critical comparison of secret crypto systems Probability of key POS = (ns – nr)/2, breaking by brute where ns is the key force attack P(k) size that will make = 1/2n–1 where n Pr(k) to 1 and nr is key size is the reference year, 2000

Factor of speed of algorithm, S(K) = 1/2m where m is the number of round

Period of collision, 2m/2 (m is the block size)

TDES with 112 bits key

3.85 × 10–34

56 years

5.96 × 10–8

4.29 × 109

TDES with 168 bits key

5.34 × 10–51

84 years

5.96 × 10–8

4.29 × 109

IDEA (128 bits key)

5.87 × 10–39

64 years

3.90 × 10–3

4.29 × 109

AES with 128 bits key with 128 bit data block

5.87 × 10–39

64 years

9.76 × 10–4

1.84 × 1019

AES with 196 bits key and 128 bits data block

1.99 × 10–59

98 years

2.44 × 10–4

1.84 × 1019

AES with 256 bits key and 128 bits data block

1.85 × 10–68

128 years

6.10 × 10–5

1.84 × 1019

COMMENT

AES is the best; Lowest probability of break

AES is the best, Maximum POS

AES is the second best in terms of highest factor of speed

AES is the best ; Highest period of collisions

14. MODES OF OPERATIONS OF AES Earlier the modes of operation that are required to use the block codes for data of arbitrary size was discussed in connection with the application of the DES. We discussed ECB and CBC, and mentioned that other modes will be discussed in connection with AES operation. The two common modes of the stream cipher encryption are CFB (Cipher Feedback) and OFB (Output Feedback). CFB is appropriate to data block of shorter sequences than 64 bits. The error transmissions may propagate in the plaintext output. OFB is not subject to error propagation and

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can use both less or 64 bits data block. Except ECB, none of the three other modes does support parallel encryption of data blocks, making it impossible to improve performance. With the evolution of the AES, a new mode the CTR mode (counter mode) was developed. The CTR mode uses a counter (Fig. 19), that is encrypted to generate a key stream. The key stream is XORed with plaintext blocks to generate cipher texts. The parallel operations, as there is having no feed back or chaining, are possible causing high performance of CTR. The counter values are nonce as they must not repeat for the cause of the security. The error propagation does not expand in the mode, but the enemy can invert the bits in the plaintext. To recommend the appropriate mode for the AES operation, different workshops and studies[58, 59] were made. The runners are the CTR with other common modes, ECB, CBC, CFB and OFB. The recommendation appears to be going in favor of the mode CCM (Counter with CBC)[60]. The CCM combines the confidentiality of the counter mode with the authenticity of the CBC mode. Encryption Counter 1

Counter 2

K

E

K

E

P1




P2




Counter 3 K

E




Pn

C1

C2

Cn

Counter 1

Counter 2

Counter 3

Decryption

K

E

K

E

C1




C2




P1

P2

K

Cn

E




Pn

Fig. 19: Illustration of CTR Mode

15. LIMITATIONS OF AES The major limitation of the AES investigated so far is the error propagation. The data path (the encryption operation) and the control path (the key generation) both involve a number of non linear operation in several rounds. The several studies[49-54,61-62] have established that the error in the single step either in the data or in the control path or in both the paths will propagate and will result in the multiple bit errors in the output cipher text and hence in the

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received plain text. Thus there emerges two goals for error detection: (1) to prevent any attacker to inject error[61] in the data path to break the cipher system, and (2) to prevent the transmission of cipher text with errors[49] as because there is no reason to transmit incorrect and useless data spending bandwidth and cost.. Both the objectives meet with the redundancy based technique. In the technique, a test decryption is performed immediately after encryption to check whether the original data block is obtained. The technique guarantees to detect all and any faults/errors. It is independent of error model. But the cost is twice (100% increase in the hardware implementation) that of the original system as a decryption system similar to the encryption system is required for checking. The authoritative work of [49] suggested a paritychecking scheme for fault detection. In the scheme a parity bit over each byte of 64 bits data block is included for error detection. The simulation studies achieved the excellent performance over the different varieties of fault models. The same work reported about 10-20% increase in the cost in hardware implementation. The error/fault detection latency in the scheme depends on the check patterns, namely checking at the output of each round, checking at the end of every round and the checking at the end of the last round. The last pattern has the maximum detection latency similar to that of the redundancy based technique.

16. LIMITATIONS OF SECRET OR PRIVATE KEY CRYPTOGRAPHY The cryptosystems under conventional encryption have many limitations that were mentioned in the previous paragraph. Besides the techniques are having two major other limitations; 1. they appear to be point-to-point in nature, and therefore, may not be applicable to public communications. In the networked world, the communicating parties may never meet and may never converse except over the network. Therefore how do they share key and communicate. 2. conveying the secret key over channel like the telephone line, which are prone to eavesdropping, may upset the very basic goal of the secret cryptosystem. This problem is known as the key distribution/exchange problem of the secret encryption. The limitation (1) could be solved with public key encryption that has been described in the latter section. The limitations(2) is actually the problems related to key generation and management that can be in general seen with the objectives of security and issues of key management: Objectives of security under a key would be achieved when the followings are made with 1. any one authorized to exchange data gets a key 2. key is transmitted reliably and protected from disclosure 3. key is hard to guess. The issues of key management are therefore paramount importance and these are: 1. key must be secret because data is not secret if key is disclosed 2. the more the key is random, the harder it will be to guess 3. the more the key is used, the easier it will be to crack 4. high quality randomness is key to keeping key secret but randomness does not come so easily particularly to computers.

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16.1 Key Establishment Protocols The key generation / distribution problems are tackled with the key establishment protocols. The different key establishment protocols are: (1) Key transport protocol that refers to the protocol whereby keys are generated and distributed by the third party, (2) Key agreement protocol that refers to the protocol whereby keys are established and agreed by the communicating parties without the help of third party, (3) a combination of distribution and agreement protocol under quantum teletransportation.

17. KEY TRANSPORT PROTOCOL (Solution to Limitation (2) of Secret Key Cryptography) The secret key encryption techniques require every pair of users to share a secret key for every session. Therefore the number of keys (K) becomes square of number of users (U) i.e., K = U2, that becomes infeasible to realize in any system. The problem is solved by the concept of key distribution center (KDC)[63-64] and application of the key transport protocol. Every user has a shared secret key kept with KDC (Fig. 20). Assume, Alice and Bob has shared keys KA and KB respectively with KDC When Alice wants to communicate with Bob, Alice makes a request to KDC for a key to communicate with Bob. The KDC authenticates Alice and if authentication is successful, selects a key say, KAB for requested session. The KDC encrypts KAB with shared keys, KA and KB: E KA,( KAB) for Alice and E KB,(KAB) for Bob where symmetric cipher is used and then transmits the encrypted keys to Alice and Bob. Now Alice communicates to Bob with KAB and confirms a ticket of KAB. Alice and Bob then communicate with the key KAB. Jack

Alice

KDC

Jill

Bob

Fig. 20: Key Distribution Centre Protocol

17.1 Needham-Schroeder Protocol A slight modification of KDC protocol is due to Needham and Schroeder. The NeedhamSchroeder[63, 64] transport protocol is illustrated below for communication between two parties, Alice and Bob through the help of server that acts like a KDC: KA : the secret key shared by Alice with server KB KAB

: the secret key shared by Bob with server : the session key created by server for a session between Alice and Bob

NA, NB : two nonce (random challenges–A number that a protocol uses only once in a lifetime) generated by Alice and Bob respectively NA and NB are the additional requirements in this protocol in respect to KDC protocol.

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The protocols works as illustrated in Fig. (21): 1

Alice

3

Bob

4

Server 2

5

Fig. 21: Needham-Schroeder Protocol

Step 1: Alice makes request for a session key to server for communication with Bob with entry Alice, Bob, and NA. Step 2: Server generates session key KAB and returns NA, Bob, KAB and encrypted” KAB and Alice” with KB (i.e., E KB (KAB, Alice)). All these data are encrypted with secret key KA i.e., E KA (NA, Bob, KAB E KB (KAB, Alice)). By checking returned NA in the server’s message, Alice becomes sure that the session key given is meant for present request and not for earlier any session. Step 3: Alice transmits E KB(KAB, Alice) to Bob. Bob gets the session key for communication with Alice. Step 4: Bob transmits E KAB(NB) to confirm the session key and this is for a session with NB Step 5: Alice confirms to B the session key and the present session by sending E KAB (NB–1). The beauty of the protocol is that the nonce allow a verification of the state of session so that duplication of key for different sessions is avoided.

18. KEY AGREEMENT PROTOCOL 18.1 Diffie-Hellman Protocol Diffie-Hellman Protocol[65-67] is a key agreement protocol. In the protocol a shared secret key is agreed upon for a session. However, the shared secret key is computed. Assume Alice and Bob are the communicating parties. The shared is computed as below: 1. Let p is a large prime number and g is an element of large order modulo p. Then if X >= 1 and Y <= (p – 1), X and Y are in 1 : 1 correspondence: Y = gX mod p and Y = loggY. However the “g” could be “e” (exponential function) also. 2. Alice picks up a random number, say, a and sends ca = ga (or ea) to Bob. 3. Bob picks a random number, say, b and sends cb = gb (or eb) to Alice. Bob computes cab = gab (or eab). Now cab is the shared secret key of Bob. 4. On receiving cb, Alice computes cba = gba (or eba). Now cba is the shared secret key for Alice. When key is gab, an eavesdropper or opponent who has no knowledge of a and b can compute the key only using a discrete logarithm that may be made computationally infeasible using currently known best algorithms[68].

18.2 Station to Station Protocol For Alice and Bob, the shared secret key is the same and one as cab = cba. The Diffie-Hellman algorithm has a drawback. Neither of the communicating party is authenticated, and therefore

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neither party knows who to it shares the secret key. The inclusion of station-to-station protocol substantiates this limitation. The station-to-station protocol uses an encryption and a signature algorithm for the purpose of authentication, and the Diffie-Hellman algorithm for shared secret key. In the protocol no specific algorithm is used either for encryption or for digital signature. It is assumed that SA and SB are the signature keys of Alice and Bob respectively and so are their signatures as SSA and SSB. The protocol then follows the following steps (Fig. 22): 1. Alice send ca as in the Diffie-Hellman algorithm to Bob 2. Bob sends cb to Alice as in the Diffie-Hellman algorithm. So Bob now computes a shared key, k = gab (or eab). In addition to cb, Bob sends encrypted message encrypted with shared key as Ek(SSB, (gb, ga)) to Alice 3. Alice after receiving cb, computes shared key, k = gba(or eba). Alice then decrypts the second part of Bob’s message with shared key and verifies Bob’s signature. Alice then sends Ek(SSA, (ga, gb)) to Bob. 4. Bob now decrypts the encrypted message sent by Alice and verifies the Alice’s signature. b

Step 1 : g ALICE

BOB b

b

a

Step 2 : g , E k(SSB, (g , g )

a

b

Step 3 : E k(SSA, (g , g ))

Fig. 22: Station-to-station protocol with Diffie-Hellman protocol

18.3 Merkles’s Puzzle Technique of key agreement The technique[68-70] is illustrated in the Fig. (23). Step 1: Alice randomly selects n integers, and hides them as solution of n puzzles. Alice sends the puzzles to Bob. Step 2: Bob chooses one of the puzzles at random and solves it. The solution becomes the key, usually known as the associated key. Bob then sends a test message encrypted with the associated key to Alice. Alice determines the associated key by trying all n integers on the test message. Example: Step 1: Alice selected three integers 4194304(the biggest number using three twos), 29, and 48. He hides these three integers as solution of three puzzles[71] as respectively: (a) what is the biggest number using three twos (see solution is 222 = 4194304), (b) A few baskets have eggs in them both chicken and duck. The number of eggs is 5, 6, 12, 14, 23 and 29. Guess a basket which when is sold out, will make as many chicken eggs left as the duck eggs. The solution is 29 as because leaving out 29, there is a combination of 23 + 12 + 5 = 40 (take as chicken eggs) that is double of the combination 14 + 6 = 20 (take as duck eggs)), and (c) A car covers a distance between two cities at a speed of 60 km/hr and 40 km/hr respectively on forward and return journey. What is the average speed? (Solution is not (60 + 40)/2 = 50 but the solution is 48 km/hr as because 2l/x = l/60 + l/40 yields a solution of 48 for x.)

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Step 2: Bob gets the puzzles, and selects randomly one puzzle say, the first one. Then a solution for the puzzle is obtained which is 4194304 that becomes the associated key. With this key, Bob will send to Alice a test message encrypted with the key. Alice will try out with all three keys to decipher the test message. Being the test message, the message is known to both parties.; and hence with the key 4194304 only, Alice will get back the test message, establishing that the session key is now 4194304.

ALICE

Step 1

BOB

Step 2

Fig. 23 : Merkles’s Puzzle Technique of key agreement

If the cost of solving each puzzle is m units, the cost to Alice is proportional to n because she is to generate and store n keys, and generates and transmits puzzles; and try all of them (on average n/2) on deciphering the test message and thus to derive the session key. The cost of Bob is m as because he is to solve one puzzle. The cost to any eavesdropper or enemy is proportional to grow m . (n/2) as he has to solve on average n/2 puzzles each at a cost of m units. He would try solving puzzles at random to find the associated key that deciphers the test message.

19. QUANTUM SECURITY The disadvantage of key distribution can be removed with the aid of quantum technology, as per several recent studies [72-74] are made on this issue. If key distribution problem is solved, the use of Vernum technique will be best technique of security. In order to solve distribution problem, use of quantum channel for sending information about key is being explored. In quantum mechanics one can not measure something without causing noise to other related parameter. For example Hysenberg’s uncertainty principle state that ∆x . ∆m = constant. Thus if ∆x is changed, ∆m is bound to change. An ideal quantum channel supports transportation of the single photon. Thus a single photon can represent a bit–0 (zero) or 1 (one). The phase or state of polarization of photon may be used for identifying the 0 or 1. For example, Photons with 0° and 90° of polarization may therefore be treated as bit 0; and photons with 45° and 135° of polarization may be assumed as bit 1. Data security through quantum channel is under active research in the UK and USA. Some positive breakthroughs have been made by Charles Bennet of IBM Research at Yorktown Heights, New York, and by Gilles Brassard at the University of Montreal. If, in the example discussed earlier, Alice wants to send Bob the secret key as required in the Vernum cipher, she can send the key, say of N bits, through quantum channels. Bob will be instructed by Alice to detect the photons (bits) from the quantum channel starting from a given time. There may be some transmission loss, and Bob may be able to detect some fraction of photons or bits. Bob will have to inform Alice over a telephone as to which photon he has seen. For this, they may share both a common and variable key. For instance, if Alice sends

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11110000 as the key, and Bob replies that he has seen the first, seventh and eighth photons (starting from the leftmost bit), then their common key shall be 100. Eavesdropping can be tackled by sending photons with different phases. For example, the bit 0 may be represented by a photon having a phase of 0° or 180°, and the bit 1 can be denoted by a photon with a 90° or 270° phase. When Bob uses, he will be able to detect the bits correctly. Alice can send data haphazardly using different polarized photons. Bob will haphazardly try to filter out the bits. After the operation, Bob will inform Alice over the telephone of the timings and the state of filter used by him. Alice can then inform him at what instances they have used the same state of filters. Based on this exchange of information. Bob and Alice will get to know their keys. Should any eavesdropper attempt to intercepted photon transmission, there shall be garbage with the key accepted by Alice and Bob. This is because the quantum theory ensures that, without changing the phase of the photon, an intercepted photon cannot be retransmitted. Therefore, a change in the polarity of the photon will let Alice and Bob immediately be known of an interception. The scheme of sending information at the one-photon-per bit level as proposed by IBM research and research of university of Montreal reported that “to send the key, the transmitter (Alice) tells the receiver (Bob) that the plans to send n bits (photons) starting at a given time. Alice than sends the bits by and only switching the phase in the transmitter between 0° to 180°; this switches the output in the receiver between “0” and “1”. Although transmission and detection losses mean that Bob will only see a small classical communication channel (the telephone, for example) to tell Alice which photons he has seen-but not which detector he has seen than in. This allows Alice and Bob to share the same random number. For example, Alice uses ten photons to send the random number 1001011101; Bob replies that he only received the second, fifth and last photon; therefore they have shared the random number 001. However, it is conceivable that an eavesdropper could intercept the signal, copy Alice’s message, and send it on to Bob without either Alice or Bob realizing. One way to overcome this, and ensure absolute security, is for both the transmitter and receiver to use non-orthogonal measurement bases. In other words, Alice sends parts of the message by switching the transmitter phase between 90° and 270°, say, and other part by switching between 0° and 180°. When the Bob and Alice are using the same base, the system works as before. However, if Alice is using 0°/180° and Bob is using 900/2700 (or vice versa), the message is meaningless-a photon that Alice sends as a “0” has a 50% chance of being received as a “1” and vice versa. Therefore when Bob tells Alice which photons he has received, he now also says which base he was using and Alice must tell him if that is a valid photon (i.e., one which was sent and received when they were both using the same base). Paul Townsend of British Telecom, working with the Malvern group, recently demonstrated self-interference of short light pulses, containing on average 0.1 photons, down 10 km of standard communications fiber using the technique”[7273,78-79]. The very success of quantum cryptography necessarily depends on quantum computing. The fundamental concept of the quantum computing is the quantum bits, referred to as qubit. A qubit is actually the phase information of any quantum state. Two possible states of qubit are | 0 > and | 1 >. Like binary bits of the classical computing, all possible superposition of qubits are possible. Therefore, a two qubit system has four computational states, namely | 00 >, | 01 >, | 10 > and | 11 >. This means a qubit can stay in superposition and simultaneously

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represent 0 and 1. The qubit is the member of a two dimensional Hilbert space containing a continuum of quantum states unlike finite dimensional binary data of conventional computers. (Hilbert space, Schrodinger Wave equation, and Heisenberg uncertainty principle are the postulates of the quantum mechanics). Thus quantum computers operate in much richer space. For example, while a conventional memory of three bits can at a time store one of the eight possible data, a quantum memory of same size can any time store all the eight data. Researches have demonstrated the mapping from a set of input quantum registers to a set of output quantum registers. It has also been demonstrated that the quantum computers can find two prime factors of a number in the time proportional to a polynomial in the number of digits. This allows quick determination of keys in RSA algorithm used in public key system. It was revealed that only a few quantum gates will be needed to solve RSA code with 100 bits key. Recent theoretical developments of quantum algorithms are note worthy: Shor factoring algorithm, Grover search algorithm, Quantum information- Schumacher theorem. Grover search algorithm deals with finding the shortest route passing through all the cities of a given map of many cities. If there are N routes, the classical computers take O(N) operations, whereas in Grover quantum algorithm will require O(square root(N)) operations. The Schumacher’s quantum theorem is analogous to the Shannon’s noiseless channel coding theorem of classical information theory. With Moore’s Law being saturated, it is expected that quantum computers will be one of the future solutions for high speed and high power computing. The main idea of quantum computing is the parallel manipulation of the infinite number of superposition of wave function, thereby achieving high speed unlike serial manipulation of the conventional computing. A few theoretical work has been reported but practical implementation is yet to reach. The approaches being followed for quantum computing are: atomic quantum implementations, bulk resonance quantum implementations and solid state quantum implementations. Table 12: Different approaches of quantum computing Implementations

Achievements

Advantages

Molecular Electronics

Molecular switches and memory

Organic FET, Protein 3D Memory, Polymer Display

Supports memory based computation

Quantum Computing

NMR(Nuclear Magnetic Resonance) devices, Spin resonance transistors, Linear optics, cavity quantum electrodynamics, trapped ions, optical lattices, superconductors

Different algorithm, Finding prime factor,

Exponential performance scaling, Parallel operation, Application in cryptography

The quantum computing is a very promising field. Different approaches are being explored (table 12) to realize the quantum computers. It is uncertain which approach will finally make the room. Practical implementation of quantum computers is still a far away. It would be worth remembering that the technology and its applications are both extreme in nature, and therefore it will be hard to crack for practical realization.

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BOX 7 Question: How does the public key (asymmetric) cryptosystems eliminate the need of the key transportation problem so acute in private key (symmetric) cryptosystems? (Hints: In the private cryptosystems, both the sender and the receiver use an identical or the same key for encryption/ decryption. This necessitates the key transportation. But in the public key cryptosystems, the sender and the receiver use separate key for encryption/decryption thereby eliminating the key transportation problem.) both

20. PUBLIC KEY CRYPTOGRAPHY (Solution to limitation(1) of secret key Cryptography) The limitation of the conventional cryptography in regard to its application only in point to point communication is removed by using the concept of public cryptosystems. In public cryptosystems, each communicating party has two keys–the secret key and the public key. The public key is used to encipher the original message, while the secret key is used to decipher the encrypted received message. For example, Bob’s public and secret key are, say, P(B) and S(B) respectively. For another person, Alice, say, her public and secret keys as P(A) and S(A) respectively. While sending data to Bob, Alice encodes the original message with Bob’s public key, P(B). Bob deciphers the received message from Alice with the latter’s secret key, S(B). The secret key is made known only to the individuals concerned. One’s public key and the secret key neutralize each other. In public cryptosystem, any one can lock a message with the public key of an intended receiver, but only the secret key holder of a receiver can unlock the received message. This system can be compared to doors with a number lock facility. From outside, one can look the door just by shutting it specific, but only a person knowing the unlocking number can open it.

20.1 Public key CryptographIc Algorithms There are many public key algorithms. The most popular among them is RSA algorithm. Other public key cryptosystems are Merkle Hellman-Knapsack algorithm[69] and McEliece algorithm[75, 79]. The first one is not symmetric in the sense that “although for all messages there corresponds a unique cipher text y, the converse is not true.” The RSA algorithm does not have this problem. The McEliece is based on algebraic coding. The scheme is better in respect of speed compared to RSA algorithm. But the McEliece algorithm has severe weakness: It does not provide protection to the messages that is encrypted more than once, and 2) it fails to protect the message which has a linear relation to one another.

20.2 RSA Algorithm RSA algorithm is named after Rivest, Shamir and Adleman[80-82] who developed it. The algorithm is based upon the fact that : 1. it is easy to find two large randomly chosen prime numbers p and q, 2. it is easy to multiply them to get m = pq, 3. it is hard to factorize m, 4. if a = 1 mod (p – 1)(p – 1) then for any non-negative integer x, we have x = xα mod m, that is known as Euler- Fermat theorem

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20.3 Key Design under RSA Algorithm The receiver (say Alice) performs the following to design two keys known as private or secret key and public key. The private or secret key will be known only to receiver, whereas the private key will be made known to whole world like telephone numbers (a) The receiver chooses two large prime numbers p and q (may be 1024 bits or about 100 decimal digits long) (b) Let n = pq., and Euler’s function ψ = (p – 1)(q – 1) (i.e., y is the number of integers between 1 and n that have no common factor with n). (c) The receiver now selects randomly an integer e between 3 and y – 1 such that the greatest common divisor is one: GOD [e, ψ] = 1 This amount to say that e has no common factors with y. e and y are relatively prime. The integer e may be defined as the public key of the receiver. (d) the receiver now finds an integer d such that d is ‘inverse of e’ modulo y (By Euclids’ algorithm): d = e–1 mod ψ or ed mod ψ = 1 or ed differs from 1 by a multiple of y The integer d may be the private or secret key of the receiver. (e) the receiver makes the public key pair (e and n) to the whole world; but keep d only known to him or her. It may be Mentioned Here that e and d are exchangeable. If e were taken as secret key, d would have been taken as private key.

20.4 RSA Encryption function Any person (say Bob) may now send data to Alice by using public key pair (e, n) of Alice. Bob to send an original or plain text, X to Alice, computes the cipher text Y by the following encryption function: Y = Xe mod n

20.5 Decryption Function Alice on receiving the cipher text Y, computes the original message by the following decryption function: X = Yd mod n The table (13 and 14) shows the operation of RSA with a few keys, and plain text. The values of p and q are very large in RSA algorithm. Since, only n = p . q and e are made public, so we now introduce a different approach on RSA algorithm where choice of p and q are not necessarily large numbers. Step 1. Choose two primes p and q at random. Step 2. Set n = p . q Step 3. Obtain φ(n) = (p – 1)(q – 1) Step 4. Set Z = [φ(n)]k where k >= 1 and integer. Step 5. Calculate e and d from the relation ed ≡ 1 mod Z. Step 6. Y = Xe mod n where X is the actual text. X = Yd mod n.

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Table 13: Illustration of key generation of RSA Key generation by the receiver

Sender Sender does Receiver wishes to encryption does the send using decryption public keys using secret of receiver key of the receiver

p(secret)

q(secret)

n(public)

y(secret)

e(public)

5

7

35

24

5

11

55

40

7 as GOD (7, 40) = 1

7

17

119

96

7

11

77

7

17

119

d(secret)

Plain text Cipher text Original or as prepared text as original and sent by computes by message the sender the receiver 2

Y = 25 mod 35 = 32

X = 325 mod 35 = 2

23 as 7 × 23 mod 40 = 1

2

Y = 27 mod 55 = 18

X = 1823 mod 55 = 2

7

55

2

Y=9

X=2

60

7

43

9

Y = 37

X=9

96

5

77

S(encoded as 19 being 19th alphabet)

Y = 66

X = 19 to be decoded as S

5 as GOD(5, 5 as 5 × 24) = 1 5 mod 24 =1

Table 14: How Alice sends a message “Above” to Bob. Bob’s public keys pair are e = 5 and n = 35 and his secret key is d = 29: What Alice does Original letter

Numeric representation (X)

Xe

A

1

1

What Bob does Cipher text Y = Xe mod n

Gets cipher text Y

Yd

1

1

1

1

A

32

B

Deciphering Original Yd mod n = letter from X numeric representation (X)

B

2

32

32

32

32 39

O

15

759375

15

15

12783403948858939111232757568359400

15

O

V

22

5153632

22

22

8.516433190865377019561944997211e + 38

22

V

E

5

3125

10

10

1000000000000000000000000000

5

E

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BOX 8 Question If p = 3 and q = 11, prove that under RSA a possible key pair will be 7 and 3. Show RSA encryption and decryption operation using the key pair for transmission of the phrase “I love you” where each character is coded by numerical position in the character set.

20.6 How Does RSA Work/Proof of RSA The question is how the decryption applied on the encrypted message recovers the original message? We had seen earlier that the relation between an original message and its encrypted cipher is: Y = Xe mod n At the receiving end the encrypted message is deciphered as: Yd mod n That must be proved as the original message, X. But Yd mod n = (Xe)d mod n. Now it is said that if p and q are prime and n = pq, then ab mod n is the same as a(b mod(p–1)(q–1)) mod n. When we apply this on the deciphering equation, we find that Yd mod n = (Xe)d mod n = X(ed mod(p–1)(q–1)) mod n = X1 mod n {as per RSA key design ed mod (p – 1)(q – 1) = 1)} = X which is what we want for RSA to work. The wonderful property of RSA algorithm is that as (Xe)d mod n = (Xd)e mod n, the e and d may interchangeably be used as either secret or public key.

20.7 How Secure is RSA The enemy or unauthorized user (say Eve) knows cipher text Y, and the public key pair (e, n). But as eve does not know the private key d, eve can not get back original text. Only way he can do is to factorize n to get p and q, which is very hard for large n. An example may predict the strength of RSA provided such long keys are possible under speed consideration: For example as per[76], one example could be: p = 50776057861427 and q = 8002247613043, whereas e and d could be respectively 190507001373540777338798221 and 368580350763525802896536629. If x = 12345678901234567890, then y = 378946255916848618281537398. However two approaches are there to break the RSA cryptography: • brute force approach whereby all possible key combinations are tried. As the key size increases, the algorithm becomes more secure As the key sizes increases, lots of time are spent on encrypting and decrypting for which there is a need for optimal choice. • the major cryptanalysis of RSA is based on factorizing n as mentioned earlier. But some examples of factorizing times are as below: n = 100 digits, around 1 week n = 150 digits, around 1000 years n > 200 digits, around 1 million years. As per[10] the number of steps, S in machine cycles (Schroepped developed this factoring) required to factor n into p and q is given by: S = exp[(ln n) ln(ln n)]0.5 Where n is related to he number of bits, b in the key as n = 2b. It will be pertaining to mention here that usually b is of 512 or 1024 bits[80]. If b is of 512 bits, p and q each will be roughly of 256 bits. As per[76], if a key of 64 bits is used, an ultra fast special computer which

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could try out one key every msec will take on average 264/2 µsec = 292271 years to find the right key. If the key size is of 80 bits, it would take a time longer than the age of the universe to find the right key. But as the computer becomes faster and more powerful by million folds, cryptanalysis may require reduced times by a factor[67] of 109. Yet then, in 1977, a challenge was issued to break a 129 digit (430 bits) message encrypted under RSA algorithm. It was reported that it was broken in 1994 over eight months period and it requires 5000 MIP years. Another report says a 155 digit (515 bits) n of RSA algorithm was broken in 1999 in 5.2 months. These do not invalidate the RSA algorithm, but suggests for higher key size. The current minimum recommended key size is 768 bits, but with 1024 or 2048 higher security is ensured. The work of Shor[88] has demonstrated that the quantum computers can factor numbers very quickly. The quantum computers may take another several years to come up, but once it is developed, the RSA algorithm may be insecure. The research[83, 85] documented the many other forms of attack on RSA, namely side channel attack, timing attack and fault (error) attack. It was concluded that the RSA is prone to all these attacks. For example, the error occurring at any stage of encryption will make cipher text invalid. The phenomenon is error propagation, and needs measure to curb. Thus although the RSA has advantage of implementation in both hardware and software over the DES, yet practical implementation of RSA requires investigation and art.

20.8 Limitations of RSA Algorithm and Suggested Solutions The algorithm itself is simple[76] but it requires modular exponentiations for encryption and decryption. This has resulted in two major limitations of the algorithm: (1) the requirement of large memory space, and (2) the computational slowness. The research [83] demonstrated that if the message(P) and the key (e or d) are of 1024 bits each, there is a need for 10311 bits to store Pe or Pd. Therefore the practical realization requires investigative approaches. The research reported in [84], portrays the slowness of RSA signature with DSA(Digital Signature Algorithm). There are several suggestions in literature to improve the speed of coding under RSA algorithm. Several proposals are suggested in literature to counter the requirement of the large temporary memory and the slowness of the RSA algorithm. These are: (1) The square and multiply method of [83], (2) CRT (Chinese Remainder Method) and (3) Recursive mod method. The square and multiply method The generation cipher text, Y for a given plain text, X requires to compute Xe mod n that invariably requires e – 1 times to compute Xe in common technique. When e = 14 the common technique will follow sequences as: X –> X2 –> X3 –> ...... X14. A faster method suggests to compute the same following sequences as: X......square......X2......multiply......X3......square......X6......multiply......X7......square......X14 The common technique requires 13 multiplication, and the square and multiply technique requires only 5 multiplications. The Chinese Remainder Technique In the technique applicable to the receiver, say Bob. Bob must be knowing his p and q. He rather than computing plaintext from the cipher text as X = Yd mod n; he does followings: • Xp = Ydp mod p and Xq = Ydq mod q where dp = d mod (p - 1) and dq = d mod (q – 1) • X = (up Xp + uq Xq) mod n where up = 1 mod p and 0 mod q uq = 1 mod p and 1 mod q

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The CRT becomes speedier because doing two exponentiation with moduli half the size of n is quicker that doing exponentiation with modulo n. Combination of the square and multiply method with the CRT If b is the bit length of n, the calculation of X = Yd mod n by the square and multiply method takes time proportional to b3. The factors of n, namely p and q have bit lengths of b/2 bits. Thus exponential of moduli half size of n takes time proportional to (b/2)3. Then receiving x from Xp and Xq will b3/(2(b/2)3) = 4 times faster than direct exponentiation. This is why the CRT has wide application in RSA implementation. Recursive mod method The method implements RSA encryption / decryption as below: FOR ENCRYPTION Y : = 1 begin for I = 1 to e do Y : = mod (Y . X, n) . means multiplication end FOR DECRYPTION Same algorithm with d replacing e and Y and X are replacing each other. For example with X = 9, e = 7 and n = 77 the algorithm for encryption works as: I = 1: Y = mod (1 . 9, 77) = 9 I = 2: Y = mod (9 . 9, 77) = 4 I = 3: Y = mod (4 . 9, 77) = 36 I = 4: Y = mod (36 . 9, 77) = 16 I = 5: Y = mod (16 . 9, 77) = 144 I = 6: Y= mod (144 . 9, 77) = 64 I = 7: Y = mod (64 . 9, 77) = 37 Thus value 37 is the cipher text that the receiver receives. The decryption functions acts as: (for n = 7 with p and q as 7 and 11 respectively, d = 43) I = 1: X = mod (1. 37, 77) = 37 I = 2: X = mod (37. 37, 77) = 60 ……………………. I = 43: X = …………….. = 9, the plaintext will be received back. The problem of RSA is compounded when the original message, x is large. The solution lies in breaking x it into several smaller parts, say, x1, x2,…xn. Each part is processed using other fast and cheap algorithm like DES. Alice and Bob shares a non secret DES key. Now Bob (assuming Bob is the sender) forms the first level encrypted message as: p = f(f(…f(f(x1) + x2) + x3…)) where f(a) is the processed output of DES encryption using key agreed upon by Alice and Bob. The important and crucial observation is that it is hard to modify or change x to say, x′ so that both x and x′ can have same value for p. Now the final encrypted message is made by: P = pd mod m where d is the secret key of Bob. Bob now can send P to Alice. Alice generates p from P using public key of Bob and deciphering by DES the original document x. With the RSA algorithm, data can be sent at a rate of few kilobits per second, whereas with DES data rate could be of a few megabits per second. This suggests to implement a hybrid scheme (a combination of RSA and DES) when the message size is long. For short message size the only RSA be employed (table 15).

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Table 15: Application of RSA Type of message

→

Technique to be employed →

Short Message

Long Message

RSA only

Hybrid : RSA and DES/RSA and AES etc.

In the hybrid scheme, RSA could be used to distribute the secret key. Thereafter using the secret key, the DES algorithm may be used to send data securely. The following steps are used: • two communicating parties exchange a key for DES operation using RSA technique • the communicating parties now used the secret key of DES to communicate message under DES technique. The hybrid operation now be done with RSA and AES following the same steps as in the RSA and DES combination.

21. TRAPDOOR KNAPSACK PROBLEM Merkle and Hellman[69] proposed a public key cryptosystem by making use of the trapdoor knapsack problem. The knapsack problem is illustrated, in general, in Fig. (24). Given the weight of filled knapsack 987 grams, find the items (or a subset) from the given set, S that are contained in the knapsack. The problem becomes computationally infeasible when the number of the elements in the set increases. Merkle and Hellman proposed that (i) for a given message in binary row vector, x, and (ii) for a known row vector of n integers, a (the vector, a is known as trap door knapsack vector); the cipher vector, c is generated as: c=a.x The construction of the vector, a, provides the secret trap door. The trap door is the secret key, and the trap door knapsack vector is the public key. Any one can send message, x, by making cipher, c when, a is known. The person knowing the trap door information can only decipher the c in order to receive back, the original message, x. A simple example is due to [69]: 1. Public key, a = (171, 197, 459, 1191, 2410) 2. Secret key or Private key or trap door information = each component of a is larger than the sum of the preceding components 3. Plain text, x = (0, 1, 0, 1, 1) 4. Cipher text, c = 3798 [Encryption: it is obtained as: c = a . x = (171, 197, 459, 1191, 2410) . (0, 1, 0, 1, 1) = 3798] 5. Applying the secret key on c, one can find from trap door knapsack vector, a that x = (0, 1, 0, 1, 1) [Decryption: x5 = 1, as because if it were 0, the sum of the other elements of a, would become less than 3798; After subtracting the effect of x5 from c, the process be recursively applied to obtain other elements of x.} Knapsack = 987 Set, S A = 270

B = 78

C = 190

D = 67

E = 289

F = 471

G = 137

I = 237

J = 370

K = 198

Fig. 24: Knapsack Problem

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The generalized Merkle Hellman Knapsack[69, 86-87] may be defined as below: • Let Bob and Alice shares three secret integers, A, B and M. These three numbers are the secret keys • Bob generates the modified weights, v(1), v(2) … v(n) of the elements of a given trap door knapsack vector, “a” whose elements have original weights as w(1), w(2) ... w(n) such that w(i) = B . v(i) mod M. • Bob informs Alice the numbers, v(1), v(2) … v(n). • Alice sends the data x(1), x(2) … x(n) to Bob in the form: Y = S x(i) . w(i) • Bob receives back x since he knows A, B and M.

22. MCELIECE’S PUBLIC KEY The McEliece’s public key technique is based on Goppa codes[89]. The Goppa codes are error correcting codes. The error correcting property of the code is destroyed when the bits of the codeword are scrambled. It works as below: • The user randomly selects a Goppa code out of all possible codes. The user then selects a permutation of the codeword bits and computer the generator matrix associated with the scrambled Goppa code. The same is now public key. • The user’s secret key is the permutation and the chosen Goppa code. • The message bits may easily be added with the randomly generated error vector for enciphering • Only the person knowing inverse permutation can correct the error to get back original message.

23. COMPARISON OF RSA AND TRAP DOOR PUBLIC KEY CRYPTO SYSTEMS A comparison is given in table (16). Table 16: A Comparison of RSA with TRAP DOOR Algorithm

Storage requirements

RSA

Several hundred to thousand bits per user but for intermediate storage the space requirement if huge

Merkle and Hellman trap door

80 kbits per user

Error propagation

Encryption/ decryption speed

Cryptanalysis resistance

Effect is A few Kbps prominent

Good (Extensive analysis was done earlier)

A few Kbps

It requires of the order of 2b/2 number of operation to break where b is the size of knapsack vector.

24. PUBLIC KEY CRYPTOGRAPHIC MECHANISMS Although the public key cryptography solves the problems of key distribution of the secret or private key cryptography, but it does so not without opening the new possibility of attacks.

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The attack is often known as the man-in-the middle attack. As the public key is readily available, it can be used by the eavesdropper to send forged documents. The question therefore to answer: How the recipient will be sure about the correct receipt of cipher text and about the legitimate sender? This refers to authentication, integrity and nonrepudiation of cryptography. A full cryptographic mechanism must cover confidentiality, integrity and nonrepudiation and authenticity. The three mechanisms applied for full protections are therefore: • encryption/decryption algorithm basically meant for confidentiality • digital signature for authentication of sender, thereby providing nonrepudiation also • integrity check functions (cryptographic hash functions) for authentication and integrity of message content. The application of hash functions is varyingly called: hash value, message digest and checksum. We have elaborately covered the encryption/decryption algorithms. In the subsequent sections we shall discuss digital signature and integrity check functions (Fig. 25).

24.1 DIGITAL SIGNATURE for authentication of the sender The public key may work in double key system to provide digital signature. The sender can encipher the original message using sender’s secret key and receiver’s public key; while the receiver can decipher the transmitted message using sender’s public key and receiver’s secret key. Alice now can send the message after enciphering the message by her private key and Bob’s public key. Bob can now decipher the received encrypted message using his private key and Alice’s public key. Bob will only get back the recognizable plain text if it was in fact encrypted by Alice private key, otherwise Bob will get garbage. This is how digital signature works (Fig. 26). AUTHENTIACATION

Who are you ?

Live authentication

Is the transmitted data that the sender created and signed intact ? Authentication after receiving message

Recognizing communicating parties at the beginning of communication

Authentication protocols Example : During handshaking of TCP

Recognizing after receipt of message

Digital Signature (1) MDn (Message Digest n, n may be 4 or 5 leading to MD4, MD5 and 2) Secure Hash Algorithm (SHA)

Fig. 25: Tree of authentication techniques

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Alice’s Computer

Encrypt

Plain Tex

Bob’s Computer

Decrypt

Encrypt

Decrypt

Received Message

Encrypted

Alice’s Private Key

Alice’s Private Key message

Fig. 26: Public Crypto (2 key)/Digital Signature

24.2 Digital signature under RSA algorithm Under digital signature scheme, Alice shall use her secret key. Assume that Alice chooses p = 7 and q = 5. And hence m = 35 and (p – 1)(q – 1) = 24. Alice chooses e = 5, and therefore she finds d = 5. She makes m = 35 and e = 5 public and keeps d = 5 secret. If Alice now wants to send a plain text 2 to Bob, she will encrypt the message as below: y = (25 mod 35)7 mod 119 encryption with the secret key of Alice followed by the public key of Bob Bob shall get back the original message as below: x = (y55 mod 119)5 mod 35 decryption with the secret key of Bob followed by the public key of Alice. Public key system only maintains the confidentiality of the message. Digital signature maintains both the confidentiality of the message and the authenticity of the sender. The receiver’s public key ensures confidentiality whereas the sender’s secret key ensures authenticity. Other digital signature, namely digital signature with message digest will be discussed later.

24.3 CHECK FUNCTIONS for Authenticity, Integrity and Norepudiation of the Message Content For message authentication, different approaches are used (Fig. 28): MAC(Message Authentication Code), SHA (Secure Hash Algorithm) and Message Digest (MD 5 is the current one). The principal goals of the techniques are: • to check and verify that the signature of the sender • the received data was not changed since the sender signed the same and sent it. MAC The technique is more appropriate foe symmetric cryptosystem. It is assumed that the communicating parties, Alice and Bob share a secret key, K. When Alice has message. M to send, she creates a MAC for the message as a function of M and K: MAC = f(M, K)

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The MAC so obtained is appended at the message. Bob after receiving the message generates MAC following the same procedure but using received message and the shared key. The generated MAC is compared with received MAC to check the integrity of the message. SHA The original message is assumed as a sequence of n-bit blocks. Each block of n bit is processed in iterative fashion to produce a n-bit hash. The hash value in then sent with message. Assuming bit by bit XOR as the hash function, the hash value is obtained as: Ci = bi1 ⊕ bi2 ⊕ ... ⊕ bip where Ci = ith bit of the has code, 1 <= i <= n p = number of n- bit blocks in the original message bij = ith bit of the jth block BOX 9 Question Assume n = 8 bits, p = 3 block. Find the hash code for the 3 blocks given below 11010101 block1 11110000 block2 11001100 block3 11101001 → This is the hash code To make the hash function more effective, the strategies like the shifting of the current hash code right or left by one bit may be employed. The use of hash in both secret and public cryptography is illustrated in Fig. (28). MD5 The message Digest works like that of the checksums used in error control. In general MDn correspond to message digest version n. MD5 corresponds to version 5. In general MD works as follows: • for a given message, M of any length, it produces a fixed length finger print of data, called digest, MD = f(M). Like those in SHA and MAC, the MD is used to check the integrity of the data, because the design is made such that f(M) is not equal to F(M′) where M is the changed M′ • the message digest provides the signature verification also. In the digital signature, the whole of the message is signed by the sender. In the message digest, the sender will sign the MD only (Fig. 28) by computing f(MD, KS) where KS is the secret key of the sender. This is also the case in application of hash function as we had seen earlier. • The message digest function, f must have the following properties: for any message digest value, MD it will be computationally infeasible to find the corresponding message, M; it will be computationally impossible to find any two messages, M and M′ such that f(M) = f(M′). Algorithm of MD5 Ron Rivest developed the MD5. The MD5 computes 128 bits message digest in a four steps process (Fig. 29). It works in following steps:

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(a) it operates on the block of 512 bits at a time. The 512 bits block is generated by a padding technique. Before padding, 64 bit original plain text block (recall in most of all encryption block size is 64 bits) is added. Then the padding is done adding 1 followed by enough 0s so that block is a multiple of 512 bits. (b) to start with a 128 bits initial digest is assumed. It is known as constant, T. (c) a four round process is then operated using complex transformations to produce digests. New digest is combined with next 512 block to produce next digest. The process continues till the final block when the final digest is produced. (d) the transformation and processing at each of the four rounds is done with 32 bit quantities since modern processors handle 32 bits most efficiently. So each digest is made of four 32 bits words (d3, d2, d1, d0), and each 512 bits padded message block is made of sixteen 32 bits words (m15, m14 …m0). The initial constant T is thought of four 32 bits words (T3, T2, T1, T0) Round operations First round In this round, sixteen steps are used to produce new value of digest from the old value. The first six steps are given below: d0 = (d0 + E(d1, d2, d3) + m0 + T0 ) followed by 7 bits left shift d1 = (d1 + E(d2, d3, d0) + m1 + T1 ) followed by 12 bits left shift d2 = (d2 + E(d3, d0, d1) + m2 + T2 ) followed by 17 bits left shift d3 = (d3 + E(d0, d1, d2) + m3 + T3 ) followed by 22 bits left shift d0 = (d0 + E(d1, d2, d3) + m4 + T4 ) followed by 7 bits left shift d1 = (d1 + E(d2, d3, d0) + m5 + T5 ) followed by 12 bits left shift ............……………………………….. where E is a function of combination of bit wise OR, AND and NOT, T4 and T5 so on are the second constant (initial digest), Second Round It works like first round with following exceptions: # E is replaced by different function, F # the constants T0 to T15 are replaced by other constants T16 to T31 # the amount of left rotations are now 5, 9, 14, 20, 5, 9 ...... (i.e., the pattern from 5 to 20 repeats) # the byte message for the ith stage ( 0 <= i <= 15) is now m(5i + 1) mod 16 Third round The operations are same as in the other rounds with following exceptions: # F is replaced by another function G. the function, G is the bit wise XOR # new set of constants namely T32 to T47 are used # the amount of left rotations are now 4, 11, 16, 23, 4, 11 ...... (i.e., the pattern from 4 to 23 repeats) # the byte message for the ith stage (0 <= i <= 15) is now m(3i+5) mod 16

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Fourth Round The operations are same as in the other rounds with following exceptions: # G is replaced by another function H. the function, H is a combination of bit wise operations of XOR, OR and NOT. # new set of constants namely T48 to T63 are used # the amount of left rotations are now 6, 10, 16, 21, 6, 10 ...... (i.e., the pattern from 6 to 21 repeats) # the byte message for the ith stage (0 <= i <= 15) is now m(7i) mod 16 The digest is made for the first 512 bits/16 bytes block with given constants, the first digest so obtained is used as initial digest for second 512 bits/16 bytes block and the process continues. The digest of last 512/26 bytes block of the message is the final messages digest. The MD5 is a fair message digest technique and there by it is used extensively. The keyed MD5 (Fig. 28) whereby the message digest is sent through encryption has got wide application in cryptographic checksum. Message

Message with MAC Key

MAC function

Link

MAC

MAC

MAC

MAC function

Compare

Key

Pass ; Accept message Reject otherwise

(a) Illustration of MAC

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Message with MAC

HASH function

Link

Computed Hash Value

Encrypted Hash value

Encrypted Hash

HASH function

Decryption

Compare Received Hash value

Hash value Accept if passed, otherwise reject

Secret key

Encryption

Secret key

(b) Illustration of HASH with Secret Key Cryptography Bit 1

Bit 2

......

Bit n

Block l

b11

b12

......

bn1

Block 2

b12

b22

......

bn2

......

......

......

......

Block p

b1p

b2p

......

bnp

Hash Code

C1

C2

......

Cn

(c) Illustration of the simple hash value generation. C may be calculated on bit by bit XOR

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Message

Message with MAC

HASH function

Link

HASH function

Computed Hash Value

Encrypted Hash value

Encrypted Hash

Decryption

Compare Received Hash value

Hash value Accept if passed, otherwise reject

Public key

Encryption

Public key

(d) Illustration of HASH in public cryptosystem Fig. 27: Different techniques of message authentications Signed Message for transmission

Message

Received signed Message for verification and then acceptance

HASH function

Link

Encrypted MD

Encrypted MD

MD function

Decryption

Computed MD

Compare Received Hash value

Message Digest Encryption

MD

Secret key

Accept if passed, otherwise reject

Secret key

Fig. 28: Illustration of digital signature generation/verification with MD (Keyed MD)

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[Operation of Keyed MD may be mathematically described as: Any transmitter, T transmits a message, M as M + E {MD5 (M), Private key of T} where E is the encryption function] Each transfer block does four round operations each of sixteen steps. Padded Message / multiple of 512 bits 512 bits

...........

512 bits

Initial 128 bits digest

512 bits Transform

Transform

Transform

Transform

Final Message Digest

Fig. 29: Message Digest Operation

BOX 10 The author of MD5 claimed that “it is conjectured that the difficulty of coming up with two messages having the same message digest is on the order of 264 operations, and that the difficulty of coming up with any message having a given message digest is on the order of 2128 operation”. What is its justification? The digest is 128 bits. The possible digest codes are then in number = 2128. The original message size in the 512 bits block is 64 bits. Thus the possible message codes are in number = 264. Therefore on average per message there exists (2128/264) = 264 number of digest. This is how two messages need 264 operations to create same digest. As there are 264 patterns of digest, the same number of operations are required to be tried out over a given digest to get back the message.

QUESTIONS 1. Say a checksum technique produces 64 bits checksum over a message block of 128 bits. On average how many messages produce the same check sum? Does it mean that getting the message from a given check sum is easy? Number of checksum codes = 264 Number of message codes = 2128 Thus on average the number of messages that will produce same checksum = (2128/264) 64 =2 .

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No it won’t be so easy. For any given message digest, on average 263 number of trials are required to get the first message. But that message may not be the exact message, because on average 264 messages produce the same check sum, causing the pattern matching requiring an average 263 trials. Thus exactly on average (263 × 263) = 2126 trials are required. 2. Let us define a notation as below: T → R : {M} K that means the sender, T sends a message, M encrypted with a key K to the receiver, R. Using this notation write encryption/decryption processes of all techniques of security described so far. 4. Apply the following property in RSA encryption/decryption, and justify any advantage if you are getting thereby: (a . b) mod n = ((a mod n) . (b mod n)) mod n For example: as 25 mod 35 = 32, then 210 mod 35 = (25 . 25) mod 35 =((25 mod 35) . (25 mod 35)) mod 35 = 32 . 32 mod 35 = 9.

24.4 Illustrate the Non-repudiation by Digital Signature of RSA The operation of banking is taken as an example (Fig. 30). If some day customer, T1 challenges the bank that he did not sent any message, M1; the bank can find out the cipher text, C11 from its computer’s database; and then apply the public key Kp of the customer T1 to create the message, M1. Unless the said customer did really send the message, C11 could not at all exist. However the customer may say that some unauthorized person may have stolen his secret key to send C11 answer to this is that it is the customer’s responsibility to keep his secret key secret and secured. Customer 1

Private key

T1 Customer 2

Ks Private key

C11 = encrypted with secret key/digitally signed C12 = C11 encrypted with receiver’s public key

Customer 3

Private key

Public key of bank C12

–

–

Bank Customer 4

Private key

Fig. 30: Illustration of Non repudiation by digital signature of RSA

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25. STRENGTH OF MECHANISM We have earlier introduced several parameters to evaluate the strength of crypto systems. In terms of those parameters we compared the systems also. Anther way of looking into the strength might be to analyze whether the algorithm is empirically secure, or provably secure or unconditionally secure. DES is the main example of empirically secure algorithm, It is time tested, and had wide acceptance. Provably secure algorithm is difficult to break as solving another problem that is known to be hard. RSA may fall in the group. Unconditionally secure algorithm is the one time key algorithm. Vernum code falls in this class. But that has also flaws: an eavesdropper can see the two plain texts by overlying the two cipher texts. Proof is as below when the algorithm is XOR operation: Basic principle for one time key with C, P and K as cipher text, Plain text and key respectively: C = P ⊕ K; P = C ⊕ K = P ⊕ K ⊕ K Attacks may reveal: C1 ⊕ C2 = P1 ⊕ K ⊕ P2 ⊕ K = P1 ⊕ P2 Making guess or sense out of two overlying plaintexts may make cryptanalysis possible. Only when key changes from cipher to cipher, the security is unconditional; and exactly that is what is Vernum code.

26. PGP (PRETTY GOOD PRIVACY) PGP provide encryption/decryption, authentication and digital signature. Besides it uses the data compression method. In order to do these it uses the techniques RSA, MD5, IDEA, and LZ compression (LZ compression is discussed in the chapter of Multimedia). The working principle of PGP is as follows: A. At the end of transmitter/ PGP encryption: • the sender hashes the original data with MD5 • the sender encrypts the hashed message with his secret key (to provide user’s authentication) • the encrypted message is concatenated with the original message • the concatenated message is compressed by LZ compression • a 128 bit key for IDEA is generated by random process (may be some content of the message and the typing speed) • with generated key of IDEA, the compressed message is encrypted by the process of IDEA encryption. The key of IDEA is also encrypted with receiver’s public key of RSA • the encrypted IDEA message and encrypted key are concatenated • the concatenated message is encoded as ASCII characters using base 64. B. At the end of receiver/PGP decryption • the received message is decoded by reverse coding of ASCII using base 64 • the key of IDEA is obtained by decryption with receiver’s secret key • the key of IDEA is used to perform IDEA decryption to obtain the compressed message • the message id decompresses by UNZIP algorithm

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• the original message is separated. The RSA encrypted the has of MD5 is used for sender’s authentication and message authentication as is done in RSA/MD5 The recommended three keys of RSA that may be used in PGP are 384 bits for common uses, 512 bits for commercial uses and 1024 bits for military uses. It is understood that with 384 bits, the encryption cracking is possible by serious & efficient crackers, with 512 bits key, the cracking is possible by huge computing facilities with huge cost and with 1024 bits key, the cracking is very very difficult even by today’s powerful computers. In future 2048 bits key may be recommended. The limitations of PGP are: 1. it uses other crypto algorithm like MD5, RSA and IDEA. But these algorithms are having patents 2. the source code of PGP is freely available in Internet, causing possible violation of regulation. BOX 11

BASIC OBJECTIVE TYPES QUESTIONS Tick the correct answer : 1.

2.

3.

4.

5.

Security refers to: (a) protection of data against link noise, systems’ noise etc. (b) protection of data against deliberate modification, destruction of data by, and disclosure of data to unauthorized users and/or intruders. (c) Keeping data private (d) None of the above. Privacy refers to : (a) protection of data against link noise, systems’ noise etc. (b) protection of data against deliberate modification, destruction of data by, and disclosure of data to unauthorized users and/or intruders (c) keeping data private (d) none of the above. Original message or uncoded text in reference to data security is known as: (a) encrypted message or text (b) offset message or text (c) transmitted code (d) plain text. Random sequence or substitution method of security is applied to a message in Fig. (1) with offsets as shown in the same figure. The encoded message shall be: (a) D P L R J O U F W B S (b) O N C N U U W W R R (c) WAR CONTINUE (d) None of the above. If the same method of security as in question (4) is applied to a data sequence of Fig. (2) with given random string to have encoded message obtained by bit-wise XOR of data sequence with random string, the encoded message shall be: (a) 1001 0011 1001 (b) 0110 1100 0110 (c) 0101 1010 0000 (e) none of the above.

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Original Message : Offsets :

CONTINUE

WAR

+ 1 + 1 – 2 – 2 + 1 + 10 + 1

0+1+1

Fig. 1 Original data :

1100

1010

1001

Random string :

0101

1001

0000

Fig. 2 6.

7.

8.

9.

10.

11.

12.

13.

14.

Pair-wise transposition (of letters in a word) method of security when applied to a plain message “HOW ARE YOU”, the encoded message shall be: (a) “ARE HOW YOU” (b) “HOW YOU ARE” (c) OHW RAE OYU” (d) None of the above. Pair-wise transposition (of letters in a word) method of security when applied to a original data stream “11011011”, the encoded message shall be: (a) “00100100” (b) “11100111” (c) “00011000” (d) None of the above. The major problem of the substitution and the transposition methods of security is that : (a) The possibility of introducers’ getting the algorithm/scheme in using some statistics of English language like “the” is the most frequent word, “qu” pair is unbreakable etc over the transmitted message. (b) The encoding algorithm is too complex (c) The encoding algorithm is irreversible. (d) None of the above. The complex coding scheme of security uses : (a) one way function for encoding message (b) non-unitary function for encoding the message (c) both unitary and non-unitary functions for encoding the message (d) none of the above. Under cryptography, the encoding of the message at the transmitter is known as: (a) deciphering (b) enciphering (c) either enciphering or deciphering (d) none of the above. Under cryptography, the decoding of the message at the receiver is known as: (a) deciphering (b) enciphering (c) either enciphering or deciphering (d) none of the above. Under a single key cryptography, the key size is 32 bits. The number of possible keys is: (a) 32 (b) 5 (c) 64 (d) 232. If a computer which can try one key every µ sec is used to try all possible keys of question (12), the average time taken to find the right key would be: (a) 32/2 µ sec (b) 5 × 2 µ sec (c) 64 × 2 µ sec (d) 232/2 µ sec = 146,135 yrs. The answer to question (13) if the key size is taken as 80 bits, would have been: (a) (146 135 × 80/32) yrs (b) The time longer than the age of the universe (c) (146 135 × 32/80) yrs (d) none of the above.

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15. The key which enciphers the plain-text is usually known as: (a) operational key (b) second-level key (c) third-level key (d) none of the above. 16. The key which enciphers the transmission of the operational key is usually known as: (a) operational key (b) second-level key/secondary key (c) third-level key (d) none of the above. 17. The file of operational keys and the secondary keys (under three level keying technique) kept in the management node is enciphered by (a) operation key (b) secondary key (c) third-level key/primary key (d) none of the above. 18. In public cryptography, the two keys used for enciphering/deciphering are known as: (a) operational key and primary key (b) primary key and secondary key (c) operation key and private key (d) public key and private key. 19. P is a plain text. SPR and SPU are respectively sender’s private key and sender’s public key. RPR and RPU are respectively receiver’s private key and receiver’s public key. The encrypted message for transmission shall be written as (“()” represents functional operation): (a) (RPU(SPR(P))) (b) (SPR(RPR(P))) (c) both (a) and (b) (d) none of the above. 20. Receiver on receiving the encrypted message of question (19) shall decipher as: (a) (SPU(RPR(RPU(SPR(P))))) = P (b) (SPU(SPR(RPR(P)))) = P (c) either (a) and (b) (d) none of the above. 21. Question (19) and (20) refers to two level coding. Under one-key technique, the encrypted and the decrypted process at the transmitter and the receiver may be respectively: (a) (RPU(P) and (RPR(RPV(P))) = P (b) (SPU(P) and (RPR(SPU(P))) = P (c) either (a) and (b) (d) none of the above. 22. Two important very popular public-key algorithms are : (a) symmetric algorithm and DES algorithm (b) symmetric algorithm and RSA algorithm (c) DES and RSA algorithm (d) None of the above. 23. DES (Data Encryption Standard) algorithm was developed by : (a) IBM (b) Rivest, Shamir and Adleman (c) ISO (d) IEEE 24. DES is a: (a) 80-bit substitution cipher (b) 80-bit transposition cipher (c) 64-bit substitution cipher (d) 64-bit transposition cipher 25. Under RSA algorithm, e and d are respectively the public and the private key. T is the plain text. [n = p . q where p and q are two very large prime numbers, used in RSA algorithm]. The encrypted message (E) is : (a) E = Td mod n (b) E = Te mod p d (d) E = Te mod n (c) E = T mod q 26. The deciphered message (T) is of question (25) is : (b) E = Te mod p (a) E = Td mod n d (c) E = T mod q (d) E = Te mod n 27. The advantage of DES over RSA is (a) RSA is around 100 times slower than DES (b) DES is easier to understand and implement (c) Both (a) and (b) (d) None of the above.

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28. The advantage of RSA over DES (a) DES is around 100 times slower than RSA (b) RSA is easier to understand (c) Both (a) and (b) (d) None of the above. 29. Presently hot and active research is going on : (a) RSA (b) DSA (c) Quantum cryptosystem (d) Dynamic cryptosystem 30. Privacy in network is maintained with the concept of : (a) keying as in security technique (b) digital signature (c) password in Unix (d) none of the above

Answers 1. (b) 9. (a) 17. (c)

2. (c) 10. (b) 18. (d)

3. (d) 11. (a) 19. (a)

4. (a) 12. (d) 20. (a)

5. (a) 13. (d) 21. (a)

6. (c) 14. (b) 22. (c)

25. (d)

26. (a)

27. (a)

28. (b)

29. (c)

30. (c)

7. (b) 15. (a) 23. (a)

8. (a) 16. (b) 24. (c)

27. MODERN CRYPTO SYSTEMS A few very modern concepts of cryptography are found in literatures[90-93]. These are speech cryptography, visual cryptography, quantum cryptography, DNA cryptography and Steganography. The ideas are in conceptual stages and require intensive research and laboratory experimentation for practical implementation. But one thing is common that cryptography is based on only coding, either in this or that form. Accordingly we suggest that an integrated solution/coding for error detection/correction and security measure may be taken up as a new avenue for future research (PROPOSAL 4).

28. INTEGRATED SOLUTION FOR ERROR AND SECURITY It is well understood that measures for error control and security are basically based on some for of coding. The optimal requirement is therefore to devise a single code that would be used both for error control and security. Such a solution will be cost effective and efficient one in terms of achieving high speed and less storage capacity. The required investigation may be illustrated with a simple example. (7, 4) Hamming code is a well known one bit error correcting code where the four information bits say, d0, d1, d2, and d3 are coded with three check bits say, c0, c1 and c2 as: d3 d2 d1 c2 d0 c1 c0 This simple error control coding may be used for security , basically to maintain confidentiality by applying several transposition for example as below: d3 d2 d1 c1 d0 c2 c0 ...... pair wise transposition of check bits from right d3 d2 d1 c2 d0 c0 c1 ...... pair wise transposition of check bits from right d3 d2 d1 d0 c2 c0 c1 ...... pair wise transposition of four bits from left d2 d3 c2 d1 d0 c1 c0 ...... pair wise transposition of four bits from right and so on. The interleaving technique for control burst error is inherently such an integration solution so long depth length is kept secret. These simple works[94] can be extended to high level investigation to develop a single integrated coding for error control and security.

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29. INTERNET SECURITY The security for protecting the owner’s site or network is as serious as protecting the data. Rather the protecting of owner’s site or network has become more essential and serious in view of Internet connectivity and global access. An organization may experience two disadvantages in having an Internet connection: (1) the possible use of Internet for non-productive and non-useful purposes by the internal users or employees and (2) the possible connection or access by non-friendly or unauthorized persons from outside into the organizational local network. For these reasons, the organizational local network needs a protection while it is connected to the Internet.

User’s site Rest of the Internet

Computer Firewall

Internet

Local site/LAN

Computer

Attacker

Fig. 31: Firewall scheme

One of the most important security measures connected with the Internet connection and access is Firewall or security gateway. A firewall is a special programmed router that sits between a site and the public Internet (Fig. 31) in order to protect internal network from outside intrusions. Firewall is a router as because it is connected to two or more physical networks and it forwards packets from one network to another selectively and after due filtering the packets. By the process of filtering, the firewall arrives at a decision either to forward or to throw away the packets. Firewall has the permit or the denial right for forwarding packets. As such it is just not a conventional router, but router with a difference and with a purpose. Firewall may also be defined as a routing computer that isolates intranet from the outside world or the external Internet. The firewall in that respect is a security gateway. An Internet firewall is meant for protecting the users’ sites from the attacks originating at other Internet sites. The internal network or the users’ site or the organization’s own network is relatively safe as its users are internal. But external network made of public Internet and outside networks connected to the Internet is potentially hostile, both being external to the

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organization and being connected to numerous diverged users or different high population. Generally firewall protects the site by access control. The access control may be outbound access control and inbound access control. The need for inbound and the outbound control are due to respectively the disadvantages (1) and (2) listed earlier while connecting local network to the Internet. The inbound access is more serious than outbound access. The access is controlled on permit-or-deny basis for which the three things are checked: the type of service requested, the client host or server host being requested and the authentication of caller. Firewalls use four techniques for access control: Service control, Direction control, User control and Behavior control. Service control checks the type of the service requested under both inbound and outbound access before permitting the access or otherwise. For this actually the port number is checked along with the table of permit and denial. Proxy server for this purpose is often used to interpret each service request and consequently take the appropriate action. Direction control interprets the direction in which the particular service is requested for. Some sites may allow internal clients to access Internet servers blocking outside Internet clients in connecting to the site’s internal servers. User control is implemented by checking the user’s identity by IP address. As the name implies the behavior control is done by collection of information. For example, an e-mail with non-senses may be blocked. Access to a web with inappropriate or irrelevant subject may be blocked. All the controls may be implemented using either proxy software (or server) or filter. The typical tasks of firewalls are: access control based on sender and receiver identities, access control based on the type of service requested, hiding the information of the internal network like its topology, addresses, virus checking on inbound access, authentication based on source traffic, and logging of Internet activities.

Filter Based Firewalls Filter based firewalls are designed by configurations like : 1. all packets that match the description {192.21.54.45, 54, 128.79.86.15, 80} may be filtered out (prohibited to forward), which says that all packets from port 54 on host 192.21.54.45 addressed to port 80 on host 128.79.86.15 are prohibited for forwarding. 2. it is not always possible and practical to name all hosts whose packets are denied for access, in case of which the description may look like {*, *, 128.79.86.15, 80} stating that filter out all packets addressed to port 80 on host 128.79.86.15 3. while the example (1) and (2) above is the description for denial, there is no reason why the same type of description can not be used for permission. For example a description {*, *, 128.79,86.15, 25) may be used to allow access to e-mail (the port for e-mail access is 25) access to host 128.79.86.15 but not to allow any other services. The above stated description-conflict is a matter of implicit or explicit nature and has really nothing to do with firewall design. The examples are hints for designing firewalls. As the filter-based firewalls work on deny or permit rules, the firewalls are characterized as: Firewalls that block traffic and firewalls which permit traffic. Firewall control mechanisms are: Packet filtering (also known as network-level firewalls), Circuit filtering (also known as circuit level firewalls) and Application gateways (also known as application level firewalls). Packet filtering is the simplest, fastest and efficient mechanism. It is based on the content of individual packets. This filtering technique identifies the properties of the

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individual packets and accordingly blocks or passes through the packets. The implementation may be done by examination of the destination address, the source address, the destination port, the source port and so on. For example the permissible outgoing IP (Internet Protocol) addresses could be 202.141.35 . x, 202.142.56 . x and so on; and the permissible incoming IP addresses could be 202.176.67 . x, 202.176.68 . x and so on. These may be implemented by the various architectures of firewalls. One such decision-table based filtering router could be as shown in Fig. 32. The speed is the major advantage of the packet filtering but certainly not the security capabilities. The simplicity of the packet filtering is responsible for the low security capability. The individual packets may not contain the full information always. As such sometimes packet filters may flop. The most modern IP router has the in built packet filtering mechanism. Packet filtering takes decision checking the only individual packet headers, whereas circuit filtering, in addition, collects and checks connection state data associated with the packets; and thereby takes a decision to either forward or block. The most important difference between the packet filters and the circuit filters is that packet filters are essentially permissive devices whereas the circuit filters are restrictive filters. Application gateways in addition to the data used in circuit filtering, checks application-specific information to implement the access right. Application gateways apply true user-based access control and behavior control. The cost of the application gateways is higher than both the circuit filter and the packet filter. This is the exchange for more precise security. The application level gateways provide an extra layer of security (Fig. 33). Three main components constitute an application gateway: a gateway node and two firewalls on either side of the gateway. In reference to Fig. (31), firewall I discards packets not addressed to the gateway and by this the inbound access is controlled. Similarly firewall II only accepts packets to gateway, and thereby controls the outbound access. Thus for outbound access, the users have to do the following operations: log onto the gateway, transfer files onto the gateway, and transfer files from gateway onto the global Internet. For inbound access, the operations will be: Log onto the gateway, transfer file from global network onto gateway node and then transfer files from the gateway onto the local network. In Fig. (34), the three filters are compared in terms of their areas of coverage. The three filters can be analogically compared with the processing of letters done by personal assistants before forwarding them to their bosses. At first level the addresses are checked which is like packet filtering. At the next level the sender’s name is checked along with the addressee and this is like the circuit filtering. At the next higher level the application of the letter may be checked along with the previous two, and this may be taken as application gateway filtering. The relationship between the different controls and the different filters is shown in table (17). Service control is the only control that fully utilizes all of the filters. The other extreme control is the behavior control that fully utilizes only the application gateways.

Proxy Based Firewalls The proxy firewalls scheme is illustrated in Fig. (35). The proxy based firewall may be the application level or application gateway firewall. In general, a proxy is a process that sits between a server process and a client process; and to a client, the proxy appears as a server and to a server the proxy appears as a client. Therefore the proxy responses to the client requests without passing the requests to the server. The proxy server checks the requests before forwarding the requests to the server; and by this the access right is controlled. The proxy software receives and interprets each service request and on checking it forwards to the destination server or rejects. A proxy server acts as an intermediate system between the external public Internet and the local network or local server. The local computer communicates with proxy server, and then the proxy server communicates with the external public Internet. By

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this also the external computer communicates with the proxy and the proxy then communicates with the internal computer. Therefore on each access request a separate link is established for security checking. Thus the external network can never directly communicate with local computer. The proxy server is also known as controlled invocation system. Proxy server hides information about internal network. As an example, suppose a company wants to make some of its web server pages accessible to all outside users and to restrict certain of the pages to employees at one or more sites. A simple filter firewall can not implement this requirement of company. For example, filter to block all external access to HTTP (Hyper Text Transfer Protocol)’s well known port 80 will block all services and pages of port 80.The solution is to use a HTTP proxy server. Outsiders can use HTTP/TCP connection with proxy which after checking URL (Universal Resource Locator) contained in the request message may allow a second connection HTTP/TCP to the company web server (Fig. 34) or prohibits the connection. Accordingly the proxy sends message in the backward direction. IP

TCP/UDP Protocol (TCP/UDP) Source Port Destination port

Source IP Destination IP Incoming

Outgoing

Permitted

Denied

Permitted

Denied

Fig. 32: Typical permit-denial routing table.

Firewall I

Application Gateway

Firewall-I accepts only the packets addressed to the application gateway for inbound access

Internet

Firewall II Firewall-II accepts only the packets addressed to the application gateway for outbound access

Organizational Local Network

Fig. 33: Illustration of working principle of application gateway

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Data link

Internet

Transport

Application

Data field

header

header

header

header

(variable)

421

(a) Packet filtering (bold area and shaded portion is the coverage zone of packet filtering) Data link

Internet

Transport

Application

Data field

header

header

header

header

(variable)

+ Connection state (b) Circuit filtering (bold area and the shaded zone is the coverage area of circuit filtering) Data link

Internet

Transport

Application

Data field

header

header

header

header

(variable)

+ Connection state + Application state ©Application gateways (shaded area shows its coverage) Fig. 34: Illustration of filters

Table 17: Relationship between the types of control and the types of filters. Type of control

Type of filtering Packet filtering

Circuit filtering

Application gateways

Service control

Fully supported

Fully supported

Fully supported

Direction control

Partially supported

Fully supported

Fully supported

User control

Partially supported

Partially supported

Fully supported

Behavior control

Not supported

Not supported

Fully supported

External client

Firewall Proxy

External HTTP/TCP connection

Local server

Internal TCCP/TCP connection

Fig. 35: Illustration of proxy based firewall

For successful e-age, high quality and efficient security measure is essential. For this, an efficient method of key generation, distribution and fast enciphering and deciphering

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techniques are required. At present RSA and DES algorithm are used mostly. In the paper we have proposed some proposition for generation of key under RSA algorithm. We believe the propositions will be useful. Active research is going to solve the problem of key distribution. In this regard, the information age is entering the domain of quantum technology for better services. If the quantum data security technique, described above, proves to be successful for long distance communications, a new revolution may take place. We can then have quantum tele transportation of information, leading to an age of IQ-the age of information and quantum technology. The paper reviews the existing crypto systems in depth, and suggests a number of proposals for improvement of the systems. The wide research scope exist for the suggested proposals. The proposed integrated solution for error control and security measure will be a cost effective, efficient and flexible one. In fact, information technology is pushing the world into a state of chaos. Like an atom in vast nuclear power, a computer virus can cause vast havoc in networks. The security problem that is inevitable due to disorder thrown into surroundings by the information processes is another manifestation of chaos. The second law of thermodynamics teaches us that the physical or life processes that are open systems cannot bring order without making its surroundings disorder. This is how when by networked society we like to achieve order in the business and organizations, we find disorder from the surroundings is increasing in different forms. Security is again a need of bringing further order to the information systems. The cycle will roll. BOX 12 Questions 1. RC4/RC5 is a fast block cipher. In RC4, a pseudo random number generator is used for the key. The output of the pseudo random generator is XORed with the plain text to produce cipher text. It is fast algorithm (why?). The size of the key may be of any length. But the same key can not be used twice (Why?). Compare RC4 with similar other cryptosystems in terms of different parameters. 2. RC5 is a fast block cipher for RSA data security. It uses parameterized algorithm with: a variable block size of 32 or 64 or 128 bits, a variable key size of 0 to 2048 bits and a variable number of rounds from 0 to 255. Analyze the cipher in term of key breaking. 3. Show that the following Fig. (1) resembles the encryption and decryption procedures of the AES algorithm, where Nr = 10, 12, or 14 for Cipher Key of length 128, 192, or 256 bits, respectively. Plaintext

Ciphertext

Encryption Round key

Add Round Key ( )

Decryption Round key

Add Round Key ( )

Inv Sub Bytes ( )

Sub Bytes ( )

Inv Shift Row ( )

Shift Row ( ) Nr-1 round Mix Columns ( )

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Round key

Add Round Key ( )

Inv Round key

Add Round Key ( )

Sub Bytes ( )

Inv Sub Bytes ( ) Final round

Final round Shift Rows ( ) Round key

Add Round key ( )

423

Inv Shift Row ( ) Round key

Add Round key ( )

Plaintext

Cipher text

Fig. 1: For the question 3

REFERENCES 1. Allen Householde et al, Computer Attack Trends Challenge Internet Security, Security and privacy, IEEE Computer Society, 2002, pp. 5-7. 2. C. T. Bhunia, Data Security, IT, Sept. 1997, pp. 69-70. 3. C T Bhunia, Data Security Techniques, CSI Communication, July 2000, pp. 11-14. 4. C T Bhunia, Integrated Solution to Security and Accuracy Problems of Data Communication, Indian Journal of Engineers, Calcutta. 5. William Stallings, Network Security Essentials, Pearson Education Asia, India, 2001. 6. H Beker & F Piper, Cipher System: The Protection of Communication, Northwood Booker, London, 1982. 7. Behrouz Forouzan, Data Communication and Networking, Tata McGraw Hill, India, 1999. 8. Larry L Peterson et al., Computer Networks, Harcourt Asia, India, 2000. 9. Leon Garcia et al, Communication Networks, Tata Mcgraw Hill, India, 2000. 10. C T Bhunia, ARQ with multiple copies on retransmission and integrated solution for error control and security, National Conv on Advances in Computer Communication Networks, IIT, Roorke, India, 2003. 11. William F Friedman, Elements of cryptanalysis, A cryptography Series, Aegean Park Press, California, 1976. 12. C E Veni Madhavan and P K Saxena, Recent Trends in Applied Cryptology, IETE Tech Review, New Delhi, Vol. 20, No. 2, March-April 2003, pp. 119-128. 13. M Agrawall, Cryptography: A Survey, IETE Tech Review, New Delhi, Vol. 16, Nos. 3 & 4, MayAugust, 1999, pp. 287-296. 14. N Bhalla, Information Security: A Technical review, IETE Tech Review, New Delhi, Vol. 19, Nos. 1 & 2, Jan-April, 2002, pp.47-59. 15. Bruce Schneier, Applied Cryptography, John Willey & Sons Inc., New York, 1996. 16. D Carlson, Digital Communication Systems, Tata McGraw Hill, New Delhi, 1998. 17. C E Shannon, Mathematical Theory of communication, The Bell System Tech J, Vol. 27, 1948, pp. 379-423, 623-656. 18. C E Shannon, Communication Theory of Secrecy System, The Bell system Tech J., 1949. 19. D. E. Denning, Cryptography and Data Security, Addison-Wesley, 1982.

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28. 29. 30. 31. 32. 33. 34. 35. 36. 37. 38. 39. 40. 41. 42. 43. 44. 45. 46. 47.

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IBM System J, Special Issue on Cryptography, Vol. 17, No. 2, 1978. C E Veni Madhavan & P K Saxena, Recent trends in Applied Cryptology, IETE Tech Review, Vol. 20, No. 2’ March-April 2003, pp. 119-128. A Menezes, P Oorschot & S Vanstone, Handbook on Applied Cryptography, CRC Press, 1997. Data encryption algorithm, X9.17, American National Standard Institute, American National Standard, 1983. William E Burr, Selecting the Advanced Encryption Standard, IEEE Security and Privacy, Vol. 1, No. 2, March-April, 2003, pp. 43-52. E Biham and A Shamir, Differential Cryptanalysis of DES-like Cryptosystems, Proc. Crypto’ 90, Springer-Verglag, 1991, pp. 2-21. E Biham and A Shamir, Differential Cryptanalysis of the full 16 round DES, Proc. Crypto’ 92, Springer-Verlag, 1992, pp. 487-496. Federal Information Processing Standard 81, DES modes of operations, National Bureau of Standards, USA, 1977, www.itl.mist.gov/fipspubs/fip81.htm . Federal Information Processing Standard 46, Data encryption Standard, National Bureau of standards, USA, 1977. Federal Information Processing standard 46-3, Data Encryption Standard, National Institute of standards and Technology, USA, 1999, http:// csrc.nist.gov/publications/fips/fips46-3/fips46-3.pdf. Martin E Hellman, An Overview of Public Ket Cryptography, IEEE Communication Mag, May 2002, pp. 42-49. W Diffe and M E Hellman, Exhaustive Cryotanalysis of the NBS Data encryption standard, Computer, June 1977, pp. 74-84. E Biham, A fast new DES Implementation in Software, Proc. International Symp. Foundations of Software Engineering, FSE 1997, pp. 260-273. H Eberle, A High Speed DES Implementation for Network application, Proc. International Conf. Cryptology, CRYPTO 1992, 12993, pp. 521-539. Fred Halsall, Multimedia Communications, Pearson Education Asia, 2001. Stinson, D, Cryptography: Theory and Practice, Boca Raton, FL, CRC Press, 1995. Cracking DES, Secrets on Encryption Research, Eiretap Politics and chip design, Electronic Frontier Foundation, 1998. W Diffie and M E Hellman, Exhaustive Cryptanalysis of NBS Data Encryption Standard, Computer, June 1977, pp. 74-84. L Kohnfelder, Towards a Practical Public Key Cryptosystem, MIT Lab for Comp Sc, June 1978. C T Bhunia, Laws and data of Information age, Electronics for You, New Delhi, March 2002, pp. 92-94. C T Bhunia, Introduction to Knowledge Management, Everest publication, Pune, 2002. C T Bhunia, “Tomorrow’s Computers”, Science & Knowledge, Jan. 1995, pp. 7-9. Tuchman W, Hellman Presents No Shortcut Solutions to DES, IEEE Spectrum. July, 1979. Federal Information Processing Standard, Publication 46-3. William F Friedman, Elements of Cryptanalysis, A cryptographic series, Aegean Park Press, California, 1976. X Lai and J Massey, Markov Ciphers and Differential Cryptanalysis, Proc. EUROCRYPT’91, 1991, Springer Verlag. J Nechvata et al., Report on the Development of the Advanced Encryption standard(AES), J research US National institute Standards and Technology, Vol. 106, No. 3, 2001, pp. 511-576. Federal Information Processing Standard 197, The advanced Encryption Standard, National Institute of Standard and Technology, 2001.

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48.

49. 50.

51.

52. 53. 54. 55. 56. 57. 58. 59. 60. 61. 62. 63. 64. 65. 66. 67. 68. 69. 70. 71. 72. 73.

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Guido Bertoni et al, Error Analysis and Detection Procedures for a Hardware Implementation of the Advanced encryption Standard, EEE Transac. On Computers, Vol. 52, No. 4, April 2003, pp. 492-504. G Bertoni et al., Fault Detection in the Advanced Encryption standard, Proc. Conf. Massively Parallel Computing systems, 2002, pp. 92-97. G Bertoni et al., On the Propagation of Faults and Their Detection in a Hardware Implementation of the Advanced Encryption standard, Proc. Intl conf. Application specific Systems, architectures, 2002, pp. 303-312. G Bertoni et al., A parity Code Based Fault Detection for an Implementation of The advanced encryption standard, Proc IEEE Intl Symp Defect and Fault tolerance in VLSI systems, 2002, pp. 51-59. B Gladman, A Specification for rijndael , the AES Algorithm, 2001, htt:/.gladman.plus.com. J Daemen and V Rijmen, The Block cipher Rijndeal, smart Card Research and Applications, J Quisquater and B Schneire Springer Verlag, 2000, pp. 288-296. M McLoone and J McCanny, High performance Single ChipFPGA Rijdeal Algorithm Implementations, Proc Workshop Cryptography Hardware and embedded Systems, 2001, pp. 68-80. E Biham , A Fast New DES Implementation in Software, Proc. Intl Symp. Foundations of Software Engg. 1997, pp. 260-273. Brian Gladman, Implementation Experience with AES Candidate Algorithms,, htttp:// fp.gladman.plus.com/2001. Brian Gladman, The Need for Multiple AES Winners, htttp://fp.gladman.plus.com/2001. M Dworkin, Recommendation for block cipher modes of operations, NIST Special Publication 800-38A, 2001. R Housley et al, Counter with CBC-MAC (CCM) AES mode of operation, . R Karri et al, Fault Based Side Channel Cryptanalysis Tolerant Rijndael Symmetric Block Cipher Architecture, Proc Defect and Fault Tolerance in VLSI Systems, DFT’01, PP. 418-426. D Boneh et al, On the importance of Checking Cryptographic Protocols for Faults, Advances in Cryptology EUROCRYPT’97, Vol. 1233, Springer Verlag, pp. 37-51. John Gorden, Public Key Cryptosystem, Proc. Networks, 1984, London, pp. 245-259. Sung-Ming Yen, Cryptanalysis of an Authentication and Key Distribution Protocol, IEEE Communications Letters, Vol. 3, No. 1, January 1999, pp. 7-8. W Diffie & M E Hellman, Multiuser Cryptographic Techniques, Proc. AFIPS National Computer Conference, 1976, pp. 109-112. W Diffie & M E Helman, New Directions in Cryptography, IEEE Trans Info Theory, Vol. IT 22, Nov’ 1976, pp. 644-654. M E Hellman, An Overview of Public Key Cryptography, IEEE Communication Magazine, May 2002, pp. 42-49. S C Pohling & M E Hellman, An Improved Algorithm for Computing Logarithms over GF(p) and its Cryptographic Significance, IEEE Trans Info Theory, Vol. IT 24, Jan 1978, pp. 106-110. R C Merkle & M E Hellman, Hiding Information and Signatures in Trap door Knapsacks, IEEE Trans Info Theory, Vol. IT 24, Sept. 1978, pp. 525-530. R C Merkle, Secure Communication over an Insecure Channel, Commun Ass Comp Mach, Vol. 21, April 1978, pp. 294-299. Yakov Perelman, Mathematics can be fun, Mir Publishers, Moscow, 1985. John Rarity, Dreams of quiet light, Physics World, June 1994, pp. 46-51. T.P. Spiller, Quantum Information Processing: Cryptography, Computation, and Teleportation, Proc. IEEE, Vol. 84, No. 12, Dec. 1996, pp. 1719-1742. V K Gupta, Quantum to Quantum Computing, IETE Tech Review, Vol. 19, No. 5, Sept.-Oct. 2002, pp. 333-347.

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R J McElliece, A Public Key System Based on Algebraic Coding Theory, JPL DSN Progress Rep. 1978. 75. John Gordon, Public Key Cryptosystems, Proc. Networks, 1984, London, pp. 245-261. 76. Hung-Min Sun, Further Cryptanalysis of the McEliece Public-Key Cryptosystem, IEEE Transactions on Communication Letters, Vol. 4, No. 1, Jan. 2000, pp. 18-19. 77. Justin Mullins, Making Unbreakable Code, IEEE Spectrum, May 2002, pp. 40-45. 78. C T Bhunia, From Classical to Quantum Technology, Information Technology, Sept. 1997, pp. 69-70. 79. R L Rivest, A Shamir and L Adleman, On Digital signature and Public Key Cryptosystems, Communications of ACM, (Commun. Ass. Comp. Mach.) Vol. 21, Feb. 1978, pp. 120-126. 80. Keiser, A Local Area Network, McGrawhill. 81. Martin, Distributed Processing and Networks, McGraw-Hill. 82. Sanjay Burman, A System using Provably Strong Cryptography is Provably not Secure: Some Practical Attacks J IETE Tech Review, Vol. 19, No. 4, 2002, pp. 161-168. 83. J IETE Tech Review, July-August 2000. 84. P Kocher, Timing attacks on implementations of Diffie Hellman, RSA, DSS and other systems, Advances in Cryptology, CRYPTO’96, Vol. 1109 Springer Verlag , pp. 104-113. 85. J Gordon, Use of intractable problem in cryptography, Information Privacy, Vol. 2, No. 5, Sept. 1980, pp. 177-184. 86. G Simmons, Informal Discussions, CRYPTO’81, University Santa Barbara, Aug. 1981. 87. P W Shor, Polynomial time algorithm for prime factorization and discrete logarithms on a quantum computer, SIAM Journal on Computing, Vol. 26, 1997, pp. 1484-1509. 88. E R Berlekamp, Goppa Codes, IEEE Trans Infor. Theory, Vol. IT 24, Sept. 1973, pp. 590-592. 89. M Naor and A Shamir, Visual Cryptography, Lecture notes in Computer Science, Proc Euro Crypto. 90. T Parsons, Voice and Speech Processing, McGrawhill, New York, 1987. 91. H J Beker and F C Piper, Secure Speech Communication, Academic Press, 1985. 92. S Katzenbeizzer and F A Petitcolas , Information hiding technique for Stegnography and Digital Watermarking, Artech House Books, 2000. 93. C T Bhunia, ARQ With Multiple Copies On Retransmission and Integrated Solution for Error Control and Security, National Convention, IIT Roorke’2003.

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5

Reviewing Information, IT and Looking into Future IT

The very definition of information and that of the knowledge are yet to be standardized. Yet there is ample hope of leaving the age IT for the age of knowledge. Let us review the concept of information and knowledge in this section.

1. INFORMATION AND KNOWLEDGE Toni Carbo Bearman described information as the lifeblood of society. Without uninterrupted supply of information, today’s society cannot run without disruption in business, industry, education, research, communication, entertainment and other activities. He graphically described this affair as “…how we make financial transactions, control the supply and movement of good and services, educate people, communicate information entertain... Work or shop from home; communicate from virtually anywhere to virtually anywhere...consult medical experts, sharing patient information, from remote areas…the list goes on”. Fritz Machlup defined information as an intangible thing, “involving either the telling of something or that which was being told.” Michael Buckland described information as tangible thing, “as opposed to knowledge which is inherently intangible.” Buckland pointed out that “in order to communicate knowledge it must be expressed or represented in some physical way as a signal, text or communication. Any such expression would, therefore, constitute informationas-thing !” From the perception of the economic dimension of information, the notion of information as a resource has been accepted in several works. Information as a resource has diverse application in different fields of management and communication etc. Horton saw information as a resource “akin to oil and other raw materials.” There are several critics to this definition of information. Among the critics, Michel Menou argued that the claim of information as a resource “needs to be supported by more than anecdotal evidence and a limited body of empirical research. …information is seldom identified to the level of specificity required to demonstrate its impact on any given situation or problem; while continuing controversies over the size and composition of the information economy, merely reflect, at least to some extent, discrepancy between the present level of understanding of micro level realities and their macro level representation.” But “the notion of information as resource is by now well-established in fact, most evidently in recognition of the related concept of a marketplace of ideas, as reflected in the profusion of national and international laws and policies relating to trade in information and its associated goods and services.” From uses point of view, “the notion of information as commodity has gained considerable currency in the past decade, with commodity in this case, comprising all manner of information 427

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services, and including trans border data flows. The concept of information as commodity is wider than that of information as resource, as it incorporates the exchanges of information among people and related activities, as well as its use.” One of the most durable definitions of information is the mathematical theory of communication developed by C. Shannon in 1948. Information refers to uncertainty of the occurrence of any message. More the probability of occurrence of an event or a message, the less is the amount of information, the event or the message carries, and vice-versa. Shannon, often known as father of the information theory defined information as the “reduction of uncertainty.” His definition of information is for information or communication processes. In information theory and coding of communication engineering; information refers to uncertainty to the occurrence of any message [1-3]. The information(I) associated with a message that has the probability of occurrence as p is: I = log(1/p) ...(1) The unit of “I” is different for different base of log as shown in table(I). If an information refers to a set of messages mi (i = 1 to n) with probability of occurrence pi(i = 1 to n), the average information, known as the entropy (H) the set carries is: n

H = Σ p i log(1/p i ) i=1

...(2)

The entropy is a measure of uncertainty of the occurrence of the set. It has a unit of bits per message The Rate of Information (R) is defined as the average number of bits of information per second. If a message source generates message at the rate of r messages per second: R = rH bits/sec ...(3) where R is the average number of bits of information per message. Table 1: Units of information Base

Unit

Unit Conversion Rules

2

bits

1 bit = 1/log2e = 0.693 nats; 1 bit = 1/log210 = 0.301 decits

E

nats

1 nat = 1/ln 2 = 1.442 bits; 1 nat = 1/ln 10 = 0.434 decits

10

decits

1 decit = 1/log102 = 3.32 bits; 1 decit = 1/log10e = 2.303 nats

The definition of Information as stated above for purely communication engineering point of view may fit to networking or compunication engineering or computer engineering. What is knowledge? As per Deva[4], knowledge = Intelligence + Experience, whereas Intelligence = information-noise and Information = process raw data. Thus knowledge is a derivative of raw data and human experience. What is corporate knowledge or organizational knowledge? The corporate Information base and the experience base together constitute the corporate knowledge. Like IQ (Intelligence Quotient) of a person, the information age is having IQ of the corporate or an organization. Managing and development of corporate knowledge so as to achieve increased efficiency, effectiveness, quality, improvement, growth and speed of production and development is known as knowledge management (KM).

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For age of knowledge, information is best defined as [4]: Raw Data when Processed = Information Information – Noise (unwanted information, misinformation etc.) = intelligence Intelligence + Experience = knowledge Knowledge + Judgment = Wisdom

U| |V || W

...(4)

An example may illustrate the relation (4). Consider the followings: Data Domain: 1, 45, 2, 78, 3, 23, 4, A From the data domain, nothing is clear. It does not signify anything Using the same data we can have information domain, by noting that the data refer to marks obtained by the students preceded by their roll numbers by four students. When the raw data is processed in ascending order we get information as shown below: Roll Number

Marks obtained

4

Absent

3

23

1

45

2

78

From the above information, when we eliminate noise (in this case it is A which actually means absent or in place of A we write Absence) we get intelligent information which is as below: Roll Number

Marks obtained

4

Absent

4

23

1

45

2

78

We now apply experience (rule gained) that the students getting 45 or more marks will be declared passed out of the examination. The knowledge domain in reference to students passing out of examination is now as below: Students bearing roll numbers 1 and 2 only have passed out the examination. Bob Debold [5] defined knowledge as “Data and Information wrapped in application and experience.” He related data, information, knowledge and wisdom as below (Fig. 1): Data

Information

Understanding Relation

Knowledge

Understanding Patterns

Fig. 1: Relation of data with others

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All these definitions are more or less same in character and nature. The different classes of value addition distinguish these data types. Recently Tom Stonier has speculated in his work “Information and Internal Structure of the Universe” that there is an analogy between mass/matter, energy/heat and information/ order of an organization. It has been argued that information (I) resident in any organization is proportional to the order (O) of the organization: I = C.O ...(5) where C is the constant of proportionality. If this relation exists there may be a possibility of interchangeability of information with energy (which otherwise speaking will establish a measurable and quantifiable relation between Industrial based society with Information based society). Tom Stonier established an exchange rate which is: 1 Joule per degree Kelvin = 1023 bits of information ...(6) It may raise many criticism and questions, but there is a direction, which if proved correct in future, may lead to a conclusion that information is not something external to nature, but a fundamental unit of nature. Shannon theory demonstrated the inverse relationship between information and entropy in case of communication or information processes. The inverse relationship holds good for physical or life processes also. By the relation between information and knowledge as demonstrated by several works (for example the equation 4) it will be a reasonably good assumption that knowledge is proportional to information. Hence the entropy and the knowledge hold the inverse relationship to each other. The laws of thermodynamics, particularly the second law of thermodynamics, govern physical or life processes. The second law of thermodynamics is related to entropy - which is a measure of disorder. An orderly system is associated with entropy minimization. Entropy minimization means minimization of energy, space and time for a given amount of effort. Life is an open system, that exchanges energy and information with its surrounding for any effect due to any cause. The second law of thermodynamics confirms that an open system or the life system can be made more knowledgeable (more ordered or reduced entropy) only by increasing the disorder in its surroundings or environment. Thus the knowledge increases order of organization or otherwise speaking minimizes the organizational consumption of energy, space and time. Therefore, the justification of the theory of Tom Stonier may hold good very much. Polayni[6] distinguished two forms of knowledge: tacit (implicit) knowledge and explicit knowledge. Tacit knowledge can be looked upon as inherently built and gained knowledge in human beings. The skills on arts, sports, leadership, and writing are typical examples of tacit knowledge. We, the Indians are rich in tacit knowledge. Explicit knowledge is earned knowledge. Scientific and Technological knowledge are examples of explicit knowledge. Nick Willard in [7] defines Tacit Knowledge as the knowledge which “ is hard to formalize and, therefore, difficult to communicate to others. It is also deeply rooted in action and in an individual’s commitment to a specific context.” He told “ explicit knowledge is formal and systematic. For this reason it can be communicated and shared in product specifications or a scientific formula or a computer program.” Explicit knowledge is easy for documentation and preservation. Explicit knowledge is preserved easily in books, journals and computers. Tacit knowledge vanishes when a person dies. It is hard to preserve. Codification of the tacit knowledge into explicit knowledge is the only means of preserving tacit knowledge. Technology is the means for codification. Therefore it is the technology, which is essence behind preserving the knowledge of any type.

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Taylor[8] gave a beautiful example of the stated two types of knowledge: “Explicit: ‘I like meat, I don’t like fish’; Tacit: ‘I don’t know what I want but I’ll know when I see it.” Taylor classified knowledge as deep or sallow and gave an example as: “Deep: ‘I gave the patient these pills because he had symptoms which indicate a certain condition that the pills are effective against’ (casual explanation of reasoning); Shallow: ‘if you’ve got a cough, try cough linctus’ (a rule-of-thumb without explanation)” Taylor then said sometimes the explicit knowledge might appear as shallow, whereas deep knowledge is tacit. I here give an example, which is more pertaining to management issue: In an organization, the new CEO finds that late coming to factory is a common thing. From manager to workers, everybody comes late in the office mostly everyday. To solve this problem the CEO has two options: (1) apply the rule of the organization to tackle the issue. The rule may be to deduct salary or to issue show cause notice etc those are well documented in the service rule of the employees, or (2) apply some artful technique to solve the problem. Application of rule to tackle the issue is the application of the explicit knowledge of the organization. But there may be several artful techniques to tackle the stated problem. One typical artful solution could be like this. The CEO himself one day comes in this office just at 9 am and keeps on standing at the gate. The latecomers, when reach the gate, on seeing the CEO standing at the gate, obviously will wish him with “good morning”. Then the CEO can reciprocate the same just looking at the watch and telling the employee “Good Morning, Go inside, time is running out.” It is understood that such a role of the CEO will surely improve the punctuality of the employees from the next day. This is a solution coming out of the implicit knowledge of the CEO. There may be several such undocumented implicit knowledge solutions to the problem. Which solution will have more impact that depends on several factors like the then environment of the organization? If the organization is well in order and the CEO is well accepted by the employees, the suggested implicit knowledge based solution is surely the best solution. But if the organization is not in order, the employees have not received salary for the last, say, two months or if the CEO is an inappropriate appointee and thereby is not enjoying the due prestige and confidence of the employees etc, the suggested implicit knowledge based solution will not work well and may be even counter productive too. Explicit knowledge can be static or dynamic. For example, the sun will rise in east. This is a static explicit knowledge. The design rule of IC (Integrated Circuit) is an example of the explicit knowledge. The design rule changes with time (1970s had one rule, today in 2001 there is another rule) and technology (Bipolar Technology to say Field Effect Technology). The design rule is an example of dynamic explicit knowledge. The implicit knowledge is mostly dynamic, and this is how it is true implicit in character.

2. PROOF OF TOM STONIER’S THEOREM The impact of IT application on time and mobility of managers can be defined by an equation: Volume of information or knowledge available on managers’ desks × Volume of mobility of managers = Constant ...(7) Information on manager’s desk is certainly proportional to the information resident in any organization. Hence:

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O × Managers’ mobility = Constant ….(8) This means that the order of organization when is high, the managers’ mobility is low. This is what is practical in nature. Thus Tom Stonier’s theorem is acceptable to this extent at least, and the use of theory may be acceptable.

3. TOM STONIER’S THEOREM WITH SHANNON’S THEOREM The quantification of the organizational KM is a difficult task. There are a few techniques for measurement of intelligence of a person, and these are IQ, EQ and most recently introduced SI (Successful Intelligence)[4]. Besides the parameters used in measurement of IQ or EQ, the two new parameters, namely social intelligence and practicability are used in measurement of SI. Sternberg the founder of SI defines SI as “mental self-management” that takes care of three kind of thinking- analytic, creative and practical. Sternberg in his research claimed that many successful managers having higher SI did not-do-well in schools and colleges where only the analytic ability is measured in the name of examinations and evaluation. Practical and common sense are a great lot in real life situations. Practical sense and creativity are the two major dimensions of today’s business. This being the case the quantification of the organizational KM can be done in terms of SI. For organizational purpose the Shannon’s theorem for information is revised as: I(information resident in the organization) = log (1/p) ...(9) where p is the probability of something that will occur but its occurrence is unknown to the organization. Thus p = 0, when nothing unknown will occur to the organization and then its resident information as per equ. (9) in huge and theoretically infinite. As per Tom then the order of the organization is perfect. When p = 1, everything is unknown to organization, its resident information is zero and order of the organization is nil. Following this it is proposed that the order as calculated in the above process told, may be used as IQ of the organization. The only issue remains how to calculate p. We propose that p be calculated as below: p = input/(input + output) where input and output respectively refer to the total information the organization has received from other organizations over a specified period of time and the total information donated to other organizations over the same period of time. As an example if p = 0.5, as per equ. (9) I = 1 when log base is 2. In that case as per equ. (5) O = 1 unit if the constant of proportionality is assumed as one. In an earlier work[9], some knowledge exchange formula is defined based on relative volume of bits exchanged. Any knowledge organization necessarily will have the knowledge inflow and outflow. Based on information inflow and outflow a measure of index of knowledge wealth of any organization may be done by the following: RKI (Relative Knowledge Index) = (Outflow in bits – Inflow in bits)/Total flow in bits Over a period of assessment, if a knowledge organization (P) inputs information from other organization (Q) over a time of t seconds using a link of c bps (bits per second) and outputs information for that other organization over a time of T seconds using a link of C bps, the said organization’s (P’s) knowledge index with respect to the organization, Q will be: ...(10) RKIpq = (CT – ct)/(CT + ct) RKIpq = – RKIqp . RKI may range from – 1 to + 1. The organization with RKI = 1, is the knowledge organization of highest level, and that with KI = – 1 is of lowest level (or not a knowledge organization). The equation (6) gives RKI of the organization who is the acceptor of ct in comparison with the donor who donates CT bits. The RKI defined in equation (10)

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necessarily assumes that the economic value of bits of all organizations is same. But I perceive that in future the knowledge organizations or industries will be classified in different levels or grades like the present types of ranking of universities and industries. At present in India, universities are ranked as one, two, three, four or five star, and industries are ranked as Naba Ratnas or Mini Ratnas etc. Based on the grading or ranking, the bit value of organizations may change. The equation (10) then be modified as: RKIpq = (CTB – ctb)/(CTB + ctb) ...(11) where B and b are respectively the bit value of donor and acceptor. However, if any organization, say P, wants to find average RKI which would be relative to a number of organizations, the equations (10) and (11) will change respectively to equations (12) and (13): RKI (Average) = (ΣCiTi – Σciti)/(ΣCiTi + Σciti) ...(12) RKI (Average) = (ΣCiTi Bi – Σcitibi)/(ΣCiTi Bi + Σcitibi) ...(13) There may be raised a question that if the knowledge organizations are already grouped or leveled as Nava-Ratnas or Mini-Ratnas, what is the further need of RKI? The answer to this question is this: A nava-Ratna may be small in departments, whereas a mini-Ratna may be large in departments. Then the flow of knowledge exchange may be comparative, and RKI then may be used to exchange knowledge sale or purchase. RKI is not only depends on bit value but also the volume of knowledge exchange and their difference. However the fixing of bit values as required in the proposed formula can be resolved with the proposed IQ of the organizations. With O = 1 unit, the bit values be unity; and for others it will relatively low or high.

4. PROPOSING LAWS OF INFORMATION It is understood that Tom Stonier’s work on information is as a treats information fundamental thing of universe. If so, information must follow fundamental laws of universe, like Newton’s Laws of motion, Mass-Energy equivalency, Theory of Relativity etc. Bringing analogy with Newton’s Laws of motion, we predict that the laws guide the information. Before we establish the Laws, the analogies are defined: 1. a bit information is analogically a mass particle. (Here bit is bit of information) 2. network is the universe of bits 3. bit moves with velocity that has the unit of bit/sec and acceleration with unit of bit/sec2 (Here bit is bit of data). 4. knowledge is analogous to force. The unit of knowledge is proposed as KN such that: 1 KN = 1 bit – bit/sec2 First Law Of Information All bits continue to stay in its files, unless it is compelled to move by processing it by the application of knowledge in order to add value, and vice-versa. Second Law Of Information The acceleration (a) of bits for moving in network derives from application of knowledge (k) by the rule: Knowledge = (bits of information) × a.

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Third Law Of Information To every action created by knowledge has equal and opposite reaction. Computer virus/security are the reactions on the knowledge application on information.

5. MASS ENERGY EQUIVALENCY As per Tom Stenier: 1 joule/°K = 1023 bits of information At room temperature: 300 joule = 1023 bits of information 23 If we assume 10 bits of information that causes 300 joule is equivalent to rest mass, M; then by the rule E = MC2, we have M × (3 × 108)2 = 300 M = 0.33 × 10–11 Kg. So, 1023 bits of information = 0.33 × 10–11 Kg or 1 bits of information = 0.33 × 10–34 Kg or 1 Kg = 3 ×1034 bit of information. Establishing knowledge exchange formula, methods and techniques is paramount essential for the knowledge age. Extensive investigation is required both at academic/university and research & development sector. The methods and formula discussed in the paper and earlier work may be tried for initial and experimental purposes. A fundamental inquiry in regard to the value of C, the constant of proportionality in the Tom Stonier’s theorem remains another subject of investigation. The present work may be read with earlier works [10-11] for gainful application and future research in this direction.

6. PRESENT IMBALANCE IN IT ERA, DIGITAL DIVIDE Scientific research has conclusively established that man is a part of nature. The technology that is an application of science has since thus following the science has now attained to implement animalization of machine and vice versa, the latest manifestation of which is the intelligent computer and its predicted applications. While the aim of science is to find answer to “why nature is like as it is” or “what makes nature as it is”; the technology apparently speaking or as is told aims to address the central social issues of reducing poverty and the gap between “haves” and “haves not.” But critical looks belie the goals of the technology. Neither the poverty has been eliminated on the mother earth nor the gap between the rich and the poor has been reduced in past ages of industrial technology and in the present age of information technology. At present scenario it is widely unexpected to see any reverse trend in the coming up knowledge society, that is supposed to be guided by the weightless economy[15], knowledge workers and knowledge factories. Before we critically look into the issues, it will be undoubtedly desirable to look into the problem in depth. In the present age of the information technology, the problem of disparity has been given a technical name called” Digital Divide” (DD), that defines the division of the society into parts of those having access to information and those are not having “ haves” and “haves not.” In general all the developed countries belong to “Information Rich” and the rest of the world to “Information Poor.” An overview of the facts and figures to understand the gravity of the digital divide may be desirable. The IT amounts to Computer, Networks and Communication. Accordingly, the digital divide will be looked into reference to

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Internet, PCs, and Telecommunication. In the present age of IT, there is a concerted prediction that the future is with BRIC ( Brazil, Russia, India and China)[16]. There is need to investigate how far, the BRIC is from its goal and/or how near they are to achieve their goal. At the same time, several reports and investigation have come to some definite judgment, may not be conclusive judgment that IT adds to prosperity both economically and growth wise. Therefore there is an immediate need to research and investigate the missing link, if any and to suggest remedial measures. The present paper aims in that direction. Tagore once told “we have only one country in this universe, and that is world.” Rabindranath Tagore’s such a powerful philosophy may ultimately realize if to-day’s tenet of “one world one village is implemented in true sense in future and the mindful processes of applications. To achieving this, a trend has already been initiated the world over. Privatization. Liberalization and globalization are replacing liberty, fraternity and equality all over the world including the countries of third world. It does not mean that liberty and fraternity have no relevance in to-day’s society. They are ever alive and their universal appeal shall ever remain for the noble human society, but today they are not all in all. Privatization and universalization will be the other social partners with them. This is a wave brought forward by different emerging technologies that are often interactive, interdependent and diffusive. Information technology, computer, communication, microelectronics, Genetic engineering, Bio-technology, Space technology are a few to name worthy. Developing world in general is far lagging behind the modern technological evolutions and revolutions. Besides the developing countries are hardly having capital to deal with such fast, rapid and perpetual changes. Developing world in general is labor intensive rather than capital intensive. Therefore, debate on the ability, the suitability and the acceptability of liberalization is going on and will continue to go on for some more time in the developing countries. Initial mismatch and inertia are parts of life and the fact is that the society never denies mobility. The society ultimately accepts technological changes, which might be off-touch to the society, even a few years back. And irony is that delayed such acceptance is done in quite haphazard and irregular ways. We can sight a figure to justify this. Telecommunications lines of India are 66% digitized, whereas figures of Brazil and Hungary are respectively 35.7% and 41%. But the faults’ figures are 218 faults per 100 lines in India and 2 faults per 100 lines in USA and Japan. Better is not the sole dimension of competitive advantages; faster is equally another important dimension. Thus it will be a sound strategy for the developing country to take part in the globalization with out any further loss of time, but with intelligent, selective, judicious and strategic applications of globalization process, with innovative applications of emerging technologies of which Internet and IT are the front-runners. But the unmindful application of the technology will be suicidal in nature. With this in mind, a thorough picture of IT in world with different parameters is critically looked into this paper before proposing a few innovative solutions. The paper deals with facts and figures of DD in reference to the world, the regions and the countries. The selection of the regions and the countries are made in relevance with the developing countries in general and India & BRIC in particular. The future of DD has been investigated with empirical formula. This investigation shows that the DD will continue to widen in future. Therefore immediate mindful application of IT is the need of the hour to stop further erosion. We are of the opinion that the process of application must not be copied from the developed nations. We may share the developed nations’ experience but we need to formulate our strategies considering our socio-economic fabric. The paper also reports the significant and productive achievements this world has witnessed with IT revolution. The missing link between DD and IT’s economic achievement has been analyzed. Based on the analysis and investigation, a few possible solutions have been proposed for the developing countries.

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6.1 DD Between the Developed and the Developing The International Telecommunication Union (ITU) used an index known as Digital Access Index (DAI) to rank/classify the countries[17]. The DAI was determined by factors like the affordability to Internet access, education, the percentage of high-speed Internet users and the availability of the raw bandwidth. Table (2) lists the ranking of countries as per DAI calculated by ITU. Table 2: DAI ranking as per ITU RANK

Country

Remark

1

Sweden

First Twenty countries includes:

2

Denmark

3

Iceland

4

South Korea

5

Norway

6

Netherlands

7

Hong Kong

8

Finland

9

Taiwan

10

Canada

11

USA

12

UK

13

Switzerland

14

Singapore

15

Japan

16

Luxembourg

17

Austria

18

Germany

19

Australia

20

Belgium

.......

............

63

Russia

64

Mexico

65

Brazil

178

• most of developed and G-5 countries like USA, UK, Germany, Japan • But prominent missing countries are France, Italy, Russia • a few small but rising countries like South Korea, Hong Kong and Taiwan • a few other small countries like Iceland, Finland, Norway etc.

Most striking issues are: • The developed countries in general belong to top ranking Dai • The developing and Poor countries belong to lower ranking Dai

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...... 78 ...... 84

......

• So far people are expecting BRIC (Brazil, Russia, India and China) as the rising sun. But none of these countries is in even first fifty ! Is there any wrong in our prediction?

South Africa ...... China

......

......

119

India

......

......

153

Nigeria

......

......

174

Guinea Bissau

175

Chad

176

Mali

177

Burkina Faso

Last five Countries

Internet Usage on Different zones world over 800,000,000 700,000,000

Internet usages

600,000,000 500,000,000 400,000,000 300,000,000 200,000,000 100,000,000 0 1 Africa

10,095,200

Asia

229,906,112

Europe

204,557,409

Middle East

14,472,500

USA

215,988,656

Latin America

49,504,287

Oceania

15,654,792

World

740,721,956

(a) of Fig. (2)

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Internet penetration world over 70

66.11

Penetration on % population

60 49.08

50 40 28.01

30 20

9.01

10

6.29

11.5

5.58

1.11 0 1 1.11

Africa Asia

6.29

Europe

28.01

Middle East

5.58

USA

66.11

Latin America

9.01

Oceania

49.08

World

11.5

(b) of Fig. (2) Fig. 2: Internet usages and penetration world over (the data are of latest as in March’2004)

The present scenario in terms of latest Internet Usage data and penetration (% population) of the different world regions[18] is portrayed in Fig. (2). The same picture as that of due to DAI is seen. The wide mismatch both in absolute terms of Internet usage and penetration % is found in between the developed and the developing countries. The picture of BRIC is bleak. The %penetration of Asia, Africa, Middle East and Latin America is below the world’s 11.5%. Only USA, Oceania and Europe are above the world figure of 11.5%. The significant observation is that Oceania beats the Europe. A glimmering disparity is illustrated in table(III). It is found that the world’s only about 20% lucky population living in the developed nations in USA and Europe have about 60% Internet access, and the poor vast population of about 70% mostly living in Asia and Africa have only about 30% world Internet access. A disparity of such a huge volume is surely a cause of concern for the technology, the technologists, and the policy makers. In table(IV), a comparison in terms of plus- minus parameters is made for the top ten countries in the world. The plus axis refers to top ten countries having highest number of Internet users, top ten countries with highest penetration ratio and top ten countries with highest per capita income[19]. The minus axis refers to top ten countries with

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highest population, along with their Internet usage penetration ratio and per capita income. The comparison gives an evidence that the countries with higher per capita income and comparatively (in relative to geographic area, of course the other proper term may be population density) are having higher penetration ratio. The inference columns of the positive axis and the negative axis clearly demonstrate a co-relation between population & per capita income on the one hand with Internet usages & penetration ratio on the other hand. The gloomy picture of BRIC is further established. However, the evidence of co-relation may be explained by followings: (a) with higher per capita income, the number of Internet users may increase, as with higher income, the affordability of Internet access will increase directly. Again with higher per capita income, education level will increase that will result in indirect increase of Internet users, and (b) the higher population of the developing countries and lower average education level of the countries, results in lower penetration ratio. Table (V) portrays the standings of countries in terms of current and future IT tools respectively in form of PCs & Internet Access and Mobile users & Broadband Technologies. It is found that: (a) Africa, Asia, China, India, Russia and Brazil are below the world average in each case except that Russia is marginally above the world average in case of telephone lines per 100 inhabitants only; and Brazil is marginally above the world average in respect to telephone lines and Cellular subscribers per 100 inhabitants. Thus the hope of BRIC is not guaranteed. (b) In the world’s topper list, only USA has frequency 2, and the countries with frequency 1 are South Korea, Iceland, Taiwan and Switzerland. (c) One observation of concern is that Europe is below world average in Internet Hosts per 10,000 inhabitants. (d) as expected, the electricity consumption (this is surely determined by the electricity availability) has a direct relation with IT uses and accesses. The picture of BRIC is still gloom in this case too. Except Russia, all other three countries, namely India, China and Brazil are below the world average. Table 3: DD over Different Regions Country

% of world total Internet usage

% of world total population (Country’s population) as estimated for 2004

REMARK

In the world, USA and Europe together have 16.33 % of total world population, and 56.67% Internet usage. Asia and Africa together have 70.63% of total world population with a meager 32.39% Internet usage. Such a Huge Disparity exists on the year 2004.

Africa

1.36%

14.03% (905954600)

Asia

31.03%

56.6% (3654644000)

Europe

27.61%

11.29% (728857380)

Middle East

1.95%

4.01% (259166000)

USA

29.06%

5.06% (326695500)

Latin America

6.68%

8.46% (546100900)

Oceania

2.11%

0.49% ( 31892487)

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Table 4: DD over different countries Positive Axis Top Ten Top Ten countries in countries terms of with highest highest Internet Internet Users (in penetration 2004) (%population) (in 2004)

Top Ten countries with highest per capita income (in 2002)

Countries with frequency 2 or more in the positive axis

USA

Sweden

China

Luxembourg Frequency 2: Japan, South Korea, Australia Norway Iceland, Denmark, Switzerland NetherSwitzerIn the list lands lands no place is there for Hong Kong Liechtenany develstein oping country. Iceland USA

Japan Germany UK South Korea

Bermuda

Negative Axis

Frequency 3: USA only

USA

No place of Bric too.

Top ten The The Countries countries countries countries with with with with rank frequency 2 highest Internet below 100 or more in population penetration in per the in the world rate (% capita negative (in 2004) population) income (in axis (in 2004) 2002) ... Yes below world or No? average of 11.5%—Yes or NO? China

Yes (6.5%)

Yes (rank 136)

India

Yes (1.5)

Yes (rank 162)

USA

NO (66.11%)

NO (rank 6)

Indonesia

Yes (3.6%)

Yes (rank 146)

Brazil

Yes (11.2%)

No (but rank 96)

Pakistan

Yes (1%)

Yes (rank 168)

Nigeria

Yes (0.35)

Yes (rank 180)

No

No (but Almost all rank 100) world highest Yes populated (rank 172) developing countries No fall in the (rank 7) group. No developed country has any place.

France

Denmark

Japan

Brazil

South Korea

Channel Islands

Russia

Italy

Singapore

Denmark

Bangladesh Yes (0.1%)

Canada

Switzerland

Iceland

Japan

No (44.75)

Frequency 3: China, India, Indonesia, Pakistan, Nigeria, Bangladesh Frequency 2: Brazil, But if the marginal figures are considered for brazil and Russia, the Brazil has frequency 3 and Russia 2.

The picture of bric is gloom.

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The weighted mean of frequency for positive axis = (17 × 1 + 5 × 2 + 1 + 3)/23 = 1.3. So fortune countries are those with frequency > 1.3 and these are USA, Japan, South Korea, Iceland, Denmark, Switzerland. The weighted mean of frequency for negative axis = (3 × 1 + 1 × 2 + 6 × 3)/10 = 2.3. The misfortune countries are with frequency > 2, and these are : India, China, Indonesia, Pakistan, Nigeria and Bangladesh (5 countries of Asia, and 1 country of Africa)

Thus it is established that to minimize DD the developing countries need to ensure (i) increased per capita income, (ii) education level and (iii) higher level of electric power availability. Comparing the Brazil’s way of globalization with that the South Korea’s way, Nobel Laureate Prof Amartya Sen repeatedly urged that in the developing countries the social infrastructure like education must be kept under state sponsorship while the physical infrastructure like electricity (power sector) be left to private sector. We are in dilemma till date on implementing the concluding judgment of Prof Sen.

6.2 DD trend in future The issue may be highlighted with a single parameter of IT penetration rate over population. It is not the absolute number of Internet, PCs etc of a country that will make it “information rich”, but the penetration rate over population. The yearly growth of IT access of a country and that of the population of that country determine the penetration rate of IT. Mathematically speaking the stated relation be given as: (dN/dP) = (dN/dY)/(1/(dP/dY) where: dN/dP = penetration rate, dN/dY = growth rate of IT access over years, and dP/dY = population growth with respect to years. While for the developed countries the reasonable empirical relations between N & Y and between P and Y will be: N = aY2 and P = bY; those for the developing countries will be at best N = cY2 and P = dY2 where a, b, c and d are constants. The reasonableness is based on the Fig. (3) that portrays the growth pattern, population and Internet users, for a few selected countries over years. Growth Rate of Internet Users

Internet users in thousand

250000 200000 150000 India USA UK

100000 50000 0 1997

(a) of Fig. (3)

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2002 Year

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442 4301.14 42.20 56.91 94.25 14.6(2)

128.68 230.38

Europe

953.07

3998.77

Brazil

Canada

16.01

1.853

USA

12.5

812.60

1450.21

85.57

America

6.1(13)

963.66

63.65

Taiwan

55.83

6.09

38.22

Singapore

4488.56

8.10

726.65

0.75

1.22

37.08

3.01

South Korea

Japan

7.58

Cellular Broadband Telephone Mobile services (in lines in Subscrib- 2002) [22] ers (in k (in 2002) 2002) [22] [22]

2169.95

822.41

5128.29

5513.77

2575.99

3814.26

5043.59

5518.91

159.14

460.09

584.69

123.01

21.44

7.48

48.70

65.89

28.95

39.46

62.20

55.58

0.72

2.76

4.45

1.30

41.34

22.32

63.55

64.58

34.73

58.17

46.29

48.86

3.98

16.69

11.99

2.77

51.26

20.06

37.72

48.81

29.90

106.15

79.56

67.95

1.22

16.09

12.42

4.59

11.5 (3)

6.5 (11)

9.4(4)

5.5 (14)

21.3(1)

Internet Internet PCs per Tel- Subscrib- Broadband Hosts per users per 100 ephone ers per subscriber 10,000 10,000 inhabit- lines per per 100 100 inhabitinhabitants 100 inhabit- inhabitants ants ants with inhabitants world rank ants in bracket()

Information Technology (in 2002) [22]

5.73

Hong Kong

India

0.474

5.02

China

Asia

Africa

Countries

0.103

586.72

Electricity Consumption per 10,00,000 persons in KWH

Table 5: IT of different countries

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1022.78

4817.41

3698.06

4230.98

5730.74

3510.38

409.32

5063.29

4119.38

5089.30

3138.32

6479.17

5128.15

3283.17

4093.64

9.91

56.51

42.43

40.57

62.13

70.87

8.87

46.66

43.13

44.17

34.71

45.14

57.68

24.14

36.93

17.90

53.86

40.40

59.06

73.57

74.42

24.22

61.77

65.09

52.35

56.89

65.28

68.86

49.44

48.88

19.07

63.98

84.87

84.07

88.89

78.93

12.01

74.47

72.75

86.74

64.70

90.60

83.32

78.56

78.62

7.7 (8)

6.3 (12)

6.5 (10)

5.3 (15)

8.6 (5)

8.6 (6)

8.4 (7)

6.6 (9)

Africa, Asia, China, India, Russia and Brazil are below the world average in each case except Russia is being marginally above average in case of telephone lines per 100 inhabitants only and Brazil is being marginally above the telephone lines and Cellular subscribers per 100 inhabitants. Thus the hope of BRIC is not guaranteed. The earlier conclusion in respect of BRIC stands well. In the world’s topper lists only USA has frequency 2, and others countries with frequency 1 are South Korea, Iceland, Taiwan and Switzerland. One observation of concern is that Europe is below world average in Internet Hosts per 10,000 inhabitants. As expected, the electricity consumption (this is surely determined by the electricity availability) has a direct relation with IT uses and accesses. The picture of BRIC is still gloom in this case too. Except Russia, all other three countries, namely India, China and Brazil are below world average.

OBSERVATIONS:

Note: The world topper in the columns are marked “RED”. The Countries above the world average are marked “BLUE”

World

Around 3

1304.21

968.42

Oceania Australia

485.03

949.54

770.34

27.92

1937.14

314.32

2343.12

UK

9.38

5.8

Sweden

Switzerland

7.33

15.11

Netherlands Russia

6.16

Germany

6.14

5.37

Finland

14.65

232.86

2370.17

Iceland France

1556.74

Denmark

6.98

325.04

Belgium

7.56

450.95

Austria

6.82

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Thus over the coming years, the penetration rate for the developed and the developing countries will grow at the rate of A . Y and B respectively, where A (= 2aY/b) and B (= c/d) are constants. This amounts to say that in coming years, whereas the growth for developing countries will increase over time, the penetration growth of the developing countries will remain a flat one. This may be called negative avalanche for gap reduction in technology: (Penetration rate of the developed nations ÷ Penetration rate of the developing nations) is proportional to years in numbers from any current reference to any future reference ...(14) This is due to disparity in growth of the population between the developed and the developing. Thus the digital divide will increase over time and as per the model proposed the gap will grow linearly with time until and unless some drastic, innovative, appropriate and corrective measures are taken. Population Growth Curve 1800000 1600000

Population in thousand

1400000 1200000 Saudi Arab India UK USA

1000000 800000 600000 400000 200000 0

2000

2001

2002

2003

2004

Year

(b) of Fig. (3) Fig. 3: Growth pattern for Internet Users and Population[7] for a few selected countries over a few years.

6.3 DD Between India and China The hope of BRIC is not evident at least and as far; the facts and figures sighted earlier are concerned. In the BRIC, India and China both belong to Asia. It is will therefore proper to investigate DD if any in between these two projected giant countries of Asia. Table (6) shows the comparison. Leaving aside all reports and speculation, it is a hard fact that India lags behind China not only in the present context but also in the future wireless/mobile 4G ages.

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Table 6: DD in between China and India India

China

1,088,056,200

1,327,976,727

Comparatively same

Internet Users in 2000

5,000,000

22,500,000

Wide gap; India lags

Internet Users in 2004

16,580,000

79,500,000

Wide gap, India lags

231.6%

253.3%

Comparatively same

Internet Penetration in % (population) 2004

1.5%

6.0%

Wide gap, India lags

PCs per 100 inhabitants in 2002

0.72

2.76

Wide gap; India lags

Compounded Annual Growth Rate (CAGR)* in main telephones lines over 1997-2002

16.4

24.3

Wide gap; India lags

IT per Capita in 1999

15.4

37.9

Wide gap; India lags

23.4%

49.1%

Wide gap; India lags

Population in 2004

Users’ growth over 2000-2004

Cellular subscribers as % of total telephone subscribers in 2002— may be taken as an index to migration to wireless age like 4G.

Remarks

India is found to be lagging behind china in almost all areas of it Regional imbalances exist even within the developing zones. As per ITU, CAGR is calculated by the formula: [{(Vp/V0) (1/n)}-1] × 100; where Vp and Vo are respectively the present value and the beginning value, and n is the number of periods.

6.4 DD Within a Country The picture of the DD at the global level, at the regional/zonal level has been examined. Within a country, the digital divide is a challenging social issue. The issue is not limited to the developing or the developed group. It is a global phenomenon. Therefore we propose to analyze the figures of one country in this context. Hong Kong has been chosen for the purpose. In table (7), a comparison of the positions of PCs in home by household income, the use of PCs by educational attainment and the use of PCs by different age group & sex group is made. As per Thematic House hold Survey[6]the table shows that: (a) the percentage of PCs in home increases with house hold income, (b) the use of PCs increases with educational level, (c) the use of PCs decrease with higher age group and (d) the male users of PC is more than female user, with the exception that in the younger group the females either overtake or remain at par with males (of course this is the global trend for younger generation).

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Table 7: Digital Divide in Hong Kong Monthly house hold income in HK$

Rate (as a percentage of all house holds in the respective monthly house hold income group)

Age group

Rate of computer use by male

Rate of computer use by female

< 10,000

15.3%

10–14

73%

72.8%

10,000–19,9999

45.9%

15–24

76.1%

78.9%

20,000–29,999

62.8%

25–34

63.6%

65%

30,000–39,999

70.7%

35–44

48.6%

44.5%

40,000–49,999

74.2%

45–54

25.2%

20.2%

>= 50,000

82.8%

55–64

9.1%

6.6%

>= 65

1%

0.6%

Educational Attainment

Rate

Secondary

52.3%

Tertiary

89.5%

Besides the parameters used above in studying the digital divide within a country (with Hong Kong as example), in all countries the digital divide exists in the Urban Rural divide axis. The table (8) illustrates a picture of several countries to this context as per[25]. The wide disparity between the urban and the rural does never guarantee the social responsibility and achievement of any technology. The urban-rural divide on economics line is the rich-poor division. The technology must be so applied, so nurtured and so adopted so as to reduce the gap rather than increasing it. Table 8: Telephone lines per 100 inhabitants in 1995 in few countries[24] Rural

Urban

Russia

8

20

40%

Moldova

6

23

25%

Georgia

3

18

16.66%

Ukraine

7

21

33.33%

Slovakia

11

28

39.3%

Albania

2

3

66.66%

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6.5 DD in language zone The people speaking in their own language irrespective of the countries they belong, can share and exchange information for achieving benefits expected to get in the age of Internet, often known as global networked age. So a classification of the people on line in Internet access by language zone may be relevant. Table (9) gives a picture as per study[23]. It gives a picture of digital divide in the language zone, clearly giving an edge to English language. Table 9: DD in language zone Language

Internet access in million

% of world online population

English

287.5

35.8%

Total non-English

516.7

64.2%

Total European non-English

276

37.9%

Total Asian Languages

240.6

33.0%

Total Scandinavian languages (Danish, Norwegian, Swedish, Iceland etc.)

14.6

2.0%

Russian

18.5

2.5%

6.6 Looking Differently The sad disparities in different forms between the “haves” and the “haves not” in the current digital age has been clearly demonstrated in the above sections. The several reports and studies, on the other hand, established the very positive and active role of the IT in economic growth and development. After several studies and researches, the dimensions of the transformations due to the information and the communication technologies have become clearer and widely understood. In broad terms, information and communication technologies advance and enhance all the aspects of economic, cultural, social and education life. The economic, social and education gap between poor & rich, rural & urban and man & woman is believed and understood to be narrowed down with infrastructure developments and accessibility of the informatics and the telematics. The UNESCO’s Mac Bridge Commission[25] stated: “There can be no genuine, effective independence without the communication resources needed to safeguard it”. Maitland Commission reported: “No development provision of any country should be regarded as balanced, properly integrated or effective without a full and appropriate role of telecommunication”. GNP per person in US$ in 1994 was respectively 24,290; 8,814 and 300 in Japan, Taiwan and India, while the Internet users in thousand were respectively 3,500; 480 and 60 for these countries. The TV sets, the PCs and the CATV subscribers per 100 people were respectively 64.1, 12 and 8.3; 5.5, 0.1 and 1.1; and 31.5, 8.1 and 14.1 for these countries. These are enough indicators on international front about the positive role of information and communication technologies in the process of the developments. The same sort of picture has been noticed in India. The relative index of development and the teledensity in terms of % deviation from all India figures were – 1% & – 15%; – 57% & – 78%; + 269% & + 692%; + 14% & + 58% and – 3% & – 36% respectively for the states of Andhra Pradesh, Bihar (including Jharkhand), Delhi,

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Gujrat and West Bengal. The correlation between the development and the teledensity is well established in the noted figures. The deviation from the common trend is noticed only in case of Himachal Pradesh. That has allied figures as – 25% and + 53%. Except a very few stray cases like that of the Himachal Pradesh; the appeal of the information and communication technologies in enhancing and improving developments is widely accepted. For substantial establishing the rosy picture of IT, the example of “grameen phone” of Bangladesh that has brought immense economic benefits to rural inhabitants and the example of NDDB( National Dairy Development Board of India) initiatives to avoid the cheaters/middlemen in milk business so that villagers can get the right price of their produce may be noted. It will be pertaining to mention here the world agencies reports[25,26] that the better communication enhances social and economic growth; the telecommunication provides benefit to cost ratio in the ranges from 5:1 to 100:1. However the rosy picture of IT for developments is clearly titled toward the developed economics, as did the technologies of immediate past industrial age. The developing countries are to borrow and import IT technologies causing flow of resources and money from the third World into the developed World. To this problem, there has been added one more, surely new in nature, in the IT age. Before and in 70s, raw materials were used to export from the developing/third World to the developed World; and the value added or finished products were used to import into the third World from the developed world. But with new technologies in hand, the developed countries are increasing the substitution of raw materials. The atomic/nuclear power generation has in a great extent diminished the importance and the economics of OPEC countries. Whereas in the IT age, as the bits can be transported in thousands and thousands within a second in networks, there is no reason to import raw material in developed countries …. this is becoming the rule of IT business. As such the World business has been gradually becoming more and more unidirectional in favor of the developed nations. Added to the injury of the third world is the present dicted globalization that has a single dimension of business. The current model of globalization is not either whole or full or total. Federio Mayor of UNESCO’s[27] told: “Globalization must never remain confined to only the networks, telecommunication, computers, the media world or markets. It will have to be based on the consolidation of a public democratic space worldwide. It is only on this condition that we will succeed in rendering globalization humane, making it a project with truly universal promise and giving it a meaning”. Tagore’s philosophy of: “ One village and one country that is world” in spirit of true globalization will be realized when any people in the world will answer in a common voice to a question “where do you live”, that “ I live in world.” Till this is reached, three-fourths of world population that live in the developing countries will have to go with one-eight of world’s output; and the developed countries will continue to enjoy and comfort with the rest of the nations. Mute question, then, remains: How do the developing countries cope up with such huge disparity, imbalance and wide gap? The question is a million dollar question in view of the fact that the technology is changing in leaps and bound. “Change” is the only thing that remains constant in the universe, whereas “creation, mastery and utilization of modern science and technology are basically what distinguished the South from the North”[28]. Besides, the political nature of technology works heavily for its social entrepreneurship. Therefore, the developing countries must not just ignore or cannot afford to ignore any new technology, particularly IT in current perspective, but have to adopt it for judicious and intelligent application rather than “hope of rising”—based unplanned uses,

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which Dr. Colombo[28] put in as: “The ability of developing countries to derive all the benefits of the new technologies faces one stumbling block right from the start. Although rapidly and seemingly effortlessly permeating the economic and production systems of the world, these technologies are not available ‘off the peg’. They have to be absorbed, metabolized, mastered and controlled. Their applications call for a pre-existing capability to insert new ideas, new practices, and new elements into a flexible system. This does not simple exist in the vast majority of the developing countries.” The cost of a computer in Bangladesh is fifty times that of the average monthly income of a Bangladeshi, whereas it is in USA that of the average weekly income of an American. In such a sad scenario the developing countries have to design flexible system for IT access and application for developments, for which the governments have to take initiatives and to this there appears no alternative. The fact is that IT adds to economic developments and benefits, but the IT in an appropriate shape has not reached the rural and right people at the right application form resulting the DD to grow in volume and dimension. “Recognizing change as the essence of existence does not mean that we have to accept all changes. Being the master of change, we will know ‘where to exploit change’ and ‘where are the opportunities’.” The several proposals in this context are suggested for possible consideration by policy makers of the developing countries in general and India in particular. This is based on the fact that whatever gap and whatever challenge is to be met revolves around three factors: (i) economic and social gap, (ii) education gap and (iii) status gap between agriculture and industry.

COMMUNITY INFORMATICS There is a fundamental difference between invention and innovation. Innovation is the art of practical application of invention for development and productivity. The developing countries need to apply innovation for IT application, the beginning of which may be the development of Community Information center at the rural areas. The rural development shall be one of the most important applications to derive benefits of IT. The community Information will have to be used for providing the IT access training, education for rural mass to benefit them as consumers, producers and citizen; and particularly for farmers in context of market information (the agriculture producers must have the direct knowledge to get actual price of the produces), weather information and crop information (fertilizer, livestock, local production etc.). The developing countries are agriculture and labor intensive, and that is how, IT application needs innovations like Community Informatics. An environment needs to be created where IT must go to the rural mass en block by the government initiative rather than rural people coming to IT. This may be achieved by engaging educated unemployed youths for imparting mobile IT training in rural India. The areas not covered by electricity must not be left out but the technique of solar powered computer system may be employed. The community informatics may be directed to (i) marketing benefits, (ii) increasing productivity of home made technologies/ appropriate technologies, (iii) mass education & training and (iv) small business initiatives. The deprived group, the disadvantaged communities, and the rural mass must be given state sponsored opportunities to use IT for their economic and developmental benefits, before using computers to replace type writers in offices.

INDUSTRIAL BARGA in line with AGRICULTURAL BARGA The existing economic and social gap between rich and poor is primarily due to two avalanche affects:

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(a) In agriculture sector, negative avalanches is “produce and perish” and (b) In business sector, positive avalanche is “produce and flourish”. The one of the possible solutions may be to bring the balance between the prices of agriculture produce and that of business produces by strict control of governments. Mac Bridge Commission reported that the farmer and the agriculture producers must have the direct market knowledge to get actual price of the produces. This is believed to be possible only with IT. Education is an investment not only in terms of money but also in terms of time and human resources. Parents have noticed that the boys/ girls after getting school level education become useless/worthless/resource less rather than resourceful in terms of earnings in the family. They neither get job nor by that time skillful for laborious jobs including agricultural jobs. Had these boys not been sent to schools rather been engaged from the childhood in agriculture related sectors; they would be more useful for earnings for the family. This clearly demonstrate that the education till not is sure with guaranteed minimum income to family, the poor family does not like to take risk of spending mainly time and money in education.

INDUSTRIAL BARGA The state of West Bengal in India has achieved a considerable amount of rural economic growth in the last two decades. The average income of the rural people has increased and the social security of rural people has been established on the solid footings. The disparity in income among the rural people has decreased considerably. An all around development of rural people and society has been noticed. However this development is due to land reforms and “ barga” system sincerely implemented by the Left-front government of W.B. in their 25 years of rule. By the process of land reforms and barga system, the agricultural workers or farmers are given confidence that they will never be thrown out of work and land they do cultivate. This confidence has led to generate among farmers the more sense of belongingness and sincerity in their work. This has reduced the victimization and the injustice meted out to them in terms of payment or no payment earlier by the Land-Lords; which in other ways has caused the agricultural productivity to increase and loss of agricultural working days to decrease as well as the agricultural disputes between labor and owner to lessen. The barga solution is our own and is not something copied from the developed nations. The economic and productivity failures in all sectors namely agricultural, industrial and banking is mainly due to disputes between labors and owners. Thus if such disputes in agriculture sectors are overcome by the barga system; it is logically extensible for other sectors like industrial and banking too. We proposed an “industrial barga” system for Indian Industries in [29]. We have achieved something unique by our own system of bargas in agriculture sector. Similarly the industrial barga not prevailing elsewhere dose not mean it is inappropriate in India. In Indian environment where economic disparity is huge and where labor is cheap and for which victimization of labor is easy; the industrial barga will be the right solution. The proposed industrial barga aims to provide share of production and profit of industries with labor, management and owner as in agricultural barga. There may be several means of implementation. The Industrial barga will not be easy to implement. A scheme of implementation illustrated in this regard in the literature [29] may be used.

RESHAPING EDUCATION/I-C-I OUR SOLUTION IN OUR WAYS “Without improved human capital, countries will inevitable fall behind and experience intellectual and economic marginalisation and isolation...... In the developed World education is a

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major political priority.”—UNESCO. Education at all levels is imperative for all around development of any nation; and it has become more and more important in the present age of competitive globalised economy guided by emerging technologies[30-33]. Education being the basic and main backbone of a nation undoubtedly is a component of social infrastructure that the government must liberally invest in, besides controlling and monitoring as well as raising the quality of education. The improved human resource generation has been tested over history and proved beyond doubt, as the sound strategy for competitive advantage, yet the poor and developing nations under compulsion and resource crunch fail to fully implement the strategy. The developing countries in general face acute problem of funding the high cost of higher education. It has several dimensions: • due to high cost of higher education, although high cost involved in higher education is a debatable issue on several aspects, the developing countries fail to increase the number of higher educational institutes causing percentage of availability of higher education per people decreasing • the stated decrease in percentage as above is compounded multifold due to high increase in enrolment ratio with growing population. The enrolment in tertiary education in India in 1970, 1985 and 1995 was respectively 4.9%, 6.0% and 6.5%. India ranked 141 out of 173 countries in the overall education index in the year 2000. India ranks below even most developing countries in terms of enrolment per 100,000 inhabitants. • the imposition(?) of globalized economy has several restrictions on investment on higher education forcing the developing and poor countries to further fall behind in higher education scenario. At the same time given the current scenario, no government is able to pull out of globalization. • it is an irony that the developing countries in order to compete with the developed nations, blindly try to copy or follow the higher education system/structure of the developed nations. The natural consequences happen in the form of wide mismatch between the requirement, supply and demand, one component of which is seen in the brain drain or reluctant brain drain. This causes the return on investment to decrease that acts heavily behind the investment by government. • it is most unfortunate that the pattern of investment in the higher educational institutes/universities lack merits causing many great universities/institutes not getting grants as appropriate or productive based, whereas other low productive based universities/institutes getting higher share of budget. This causes the negative avalanche in the productive growth of the higher education. • due to GATT adoption that is next natural steps of globalization, the private initiatives in higher education have been noticed at the developing countries. But several studies have clearly demonstrated the huge imbalance in the process that may be counter productive. Our prolonged exercise in copying/adopting the developed countries, in cheap labor and in the generation of third largest(?) technical manpower has proved wrong. They have neither brought any meaningful results nor any relief in any form. It is undoubtedly true that our achievements in academic and basic scientific research are internationally recognized; yet we need to import telecommunication switches, PCs, and air crafts. We are called software giant, yet till date no system software like DOS, WINDOWS, and Unix we could originate. This clearly speaks for the wide gap in between Academic and Industry that is clearly absent in the developed countries. Our failure to lead “INVENTION TO INNOVATION” is the root cause for the continuous lag. This is not identified today, but long back. Even Kothari

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Commission argued for the Institute Industry interaction, but till date no concrete and meaningful interaction achieved. Now a days the industries are opening universities. We need to explore the possibilities of Universities opening industries. The two-way bridges are more flexible and surely competitive, rather than one-way bridge. Therefore a new solution in our form for a solid collaboration is the need of the hour. In this context, the concept of “I-C-I” (Institute Cum Industry) with FARE and SARE [34-37] is a powerful solution. The implementation of I-C-I must be based on several strategies: (a) It should aim FOAK both for academic programs and industrial production, because we badly need a quantum jump to come ahead in the race in coming days, (b) No failure must be discouraged, as failures add experience to the venture, (c) Short term ROI must be forgotten for ever, wait for 6 to 10 years, (d) The research centers must be taken as profit rather than cost centers, (e) Perceptions should not over rule reality and (f) Doers not makers must be reawarded. There is an urgent need to realize DIGITAL DIVIDEND (DIVIDE + END) in order to utilize the opportunities and scope that the emerging technology has to offer. The DIVIEND is to raise the voices of the poor, the deprived, the oppressed and the disadvantaged society; and then only the technology will justify its due role on social dimension

7. LOOKING INTO FUTURE IT The knowledge age is difficult to see as a new one. From the analysis made on the issues, objectives, approaches and problems of KM, it is established that many of these are common to the information age or information management. Truly the knowledge age is derived from or logically next to the information age. This is no new. New ages always establish itself by making good use of its predecessor. The issues or objectives on which knowledge age or management will be different from other ages or managements are limited in number; and themes of difference from the information age or management are mainly: (1) knowledge sharing rather than individual knowledge and (2) weightless (or knowledge) as wealth rather than physical form of wealth. This part of conclusion is not the overall conclusion over the subjects and substances dealt so far on the topic of knowledge age. I would like to conclude the topic by viewing the relevancy of the subject in philosophical line of two great Indians, Swami Vivekananda and Rishi Aurobindo. Let me list here a few observations [12] Swamiji made over nature, man and knowledge: “Nature with its infinite power is only a machine.” “All our knowledge is based upon experience…. All human knowledge proceeds out of experience; we can not know anything except by experience.” “Man is man so long as he is struggling to rise above nature, and this nature is both internal and external.” These observations of Swami Vevekananda imply that man by earns knowledge from experience, and he applies his knowledge to be creator of nature, which is not impossible so long nature is assumed a machine. It will be pertaining to mention here that Tagore told that everything in nature follows a rule. This supplements my views that the KM is a step of human effort where he attempts to be his known creator. For sometimes, I was trying to speculate what is next to the knowledge age. I remember one incident of history. The great Akbar once asked his naba- ratnas: “what moves fast?” When eights of nine ratnas pointed towards Royal Horse, the ninth ratna, Birbal got an edge over

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others by saying “Our Mind, Sir.” I at least find a technology area where the trend is to achieve something like speed of mind. Yes, I am referring to communication. From the trend of communication I have no hesitation (and I am sure all will agree to it) to conclude that it is the speed of communication that is growing leaps and bound. We have seen the age of kilobits per second, and mega bits per second, and presently in the age of gigabits per second, and are seeing a tomorrow of tera bits per second. This is an indication that after knowledge age, the next age may be the age of mind or the age of conscious. The universe is made of non-living and living things. Their comparison in terms of level of intelligence, conscious and communication power is made in table (10). S Ranade, a great admire of Aurobinda told [13]: “Knowledge by identity will change current science completely. Particularly physics and biology will see radical changes. The wave-particle duality and the mass-energy equivalence will be seen in the light of the more basic substance of consciousness” and then he defined [13]: “ consciousness is awareness, awareness of yourself and of others. In the human being both exist. In the animal, there is only awareness of others, not awareness of itself, it is a more limited awareness. In plants the awareness is even less. In the crystal it is still less, but nevertheless it is there.” If the crystal is having awareness, it is surely possible that “the next century will be the century of consciousness” and “you can focus your body consciousness on a point outside the body.” Will the “Will power” or “Mind Power” of Iswar Patuli depicted by great Bengali Novelist Sarat Chandra prevail upon the society, organization, culture and economy at the fragile end of knowledge age? I end with a quote from Mother’s historical declaration[14] made on April’24 1956: “ The manifestation of the supramental upon earth is no more a promise but a living fact, a reality. It is at work here, and one day will come when the most blind, the most unconscious and even the most unwilling shall be obliged to recognize it.” Table 10: Comparison of different entities in universe in terms of sense and communication Non-living things Living things

Apparently no sense and no communication. Dr Ranade sees otherwise Plants

Limited sense and no communication

Animals

Low level sense and communication

Human beings

High level sense and communication

REFERENCES 1. Taub abd Scihhiling, Principles of Communication, McGrawHill. 2. Wilfried Gappmair, Claude E Shannon: 50th Anniversary of Information Theory, IEEE Communication Magazine, April 1999, pp. 102-105. 3. D Carlson, Communication System, McGrawHill. 4. Yaswant Deva, Information Infrastructure, Journal IETE Tech Review, Nov-Dec. 1996. 5. Bob Debold, Managing Individual and Organizational Knowledge, . 6. M Polanyi, The Tacit Dimension, Soutledge and kegan Paul, London, 1966. 7. Nick Willard, Knowledge Management, Managing Information, UK June 1999, pp. 45-49. 8. Robert M Taylor, Knowledge Management, [email protected]/roberttylor@ compuserve.com.

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9. E Biglieri et al, Digital Transmission in the 21st Century….., IEEE Commn Mag, May 2002, pp. 128-137. 10. C T Bhunia, Introduction to Knowledge Management, Everest Publications, 2003. 11. C T Bhunia, Knowledge Exchange Formula, Proc. 37th National Conference of CSI’2002, Tata McGrawhill, 2002, pp. 271-275. 12. Swami Vivekananda, Pearls of Wisdom, RKM Institute of Culture, Calcutta, India, 1998. 13. S Ranade, The Technology of Consciousness, Dipti Publications, Sri Aurobindo Ashram, Pondichery. 14. Rada in “Introduction to Information Technology” Basil Blackwel, UK, 1985. 15. C T Bhunia, Knowledge Management: Why & How, JIETE, Vol. 21, No. 1, Jan-Feb. 2004, pp. 2537. 16. Goldman Website “dreaming with BRIC: the path to 2050.”. 17. Steven M Cherry, A World divided by A Common Internet, IEEE Spectrum, Feb. 2004, pp. 50-51. 18. . 19. World Development Indicators database, World Bank, April 2004. 20. Hong Kong Computer Society Paper on Digital Divide in Hong Kong. 21. World Development Indicators 2003. 22. World Telecommunication Indicators, ITU, December 2003. 23. . 24. World Telecommunication Development Report, ITU, 1998. 25. Mac Bridge Commission Report of World Bank. 26. Maitland Commission Report, ITU, Geneva, 1984. 27. Tibor Braun, Wolfgang Glanzel and Andras Schubert, A global snapshot of scientific trends UNESCO Courier, May, 1999, pp. 28-29. 28. Dr Colombo, The technological Revolution and the Future of the Third world, IEEE Technology & Society Magazine, Spring, 1991, pp. 25-32. 29. C T Bhunia, IT: An Enginee…, Proc. 19th National Seminar of IE(I), 2004, Patna, pp. 61-70. 30. Gerardo R Ungson & John D Trudel, The Emerging Knowledge-Based Economy, IEEE Spectrum, May, 1999, pp. 60-65. 31. Constantine N Anagnostopolous & Lauren A Williams, Few Gold Stars for Pre college Education, April, 1998, pp18-26. 32. C T Bhunia, Person Power Development - Why and How, CSI Communication, 1996, pp. 7-11. 33. C T Bhunia et al, Technical Education & Training for the Information Age, J Productivity, Vol. 39, No. 4, 1999, pp. 579-587. 34. C T Bhunia, Institute Cum Industry—A new model for technical education, University News, Aug. 29, 1994, pp. 11-14. 35. C T Bhunia et al, Model Structure of I C I of Electronics and Computer Engineering, Proc ISTE National Seminar, Feb. 14-15, 1995, pp. 1-11. 36. C T Bhunia, Higher Education - restructuring, J University News, Jan. 2000, pp. 1-4. 37. C T Bhunia Overcoming High Costs of Higher Education, J University News, June 2003, pp. 612.

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