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Data and Computer Network

Communication, Second Edition

Data and Computer Network
Communication, Second Edition

by Shashi Banzal
Laxmi Publications © 2015 Citation

Recommend?

Data and Computer Network Communication, Second Edition
Prof. Shashi Banzal
Ex Faculty Member

Deptt. of Computer Science P.M.B. Gujarati Science College
Indore, M.P.

All rights reserved including those of translation into other languages. In accordance with the Copyright
(Amendment) Act, 2012, no part of this publication may be reproduced, stored in a retrieval system, or transmitted in
any form or by any means, electronic, mechanical, photocopying, recording or otherwise. Any such act or scanning,
uploading, and or electronic sharing of any part of this book without the permission of the publisher constitutes
unlawful piracy and theft of the copyright holder's intellectual property. If you would like to use material from the
book (other than for review purposes), prior written permission must be obtained from the publishers.

Typeset at ABRO Enterprises
First Edition: 2007; Second Edition: 2015
UDC9737595DATA & COMP NETWORK COMBAN
9789383828395
ISBN 9789383828395

Limits of Liability/Disclaimer of Warranty: The publisher and the author make no representation or warranties with
respect to the accuracy or completeness of the contents of this work and specifically disclaim all warranties. The
advice, strategies, and activities contained herein may not be suitable for every situation. In performing activities
adult supervision must be sought. Likewise, common sense and care are essential to the conduct of any and all
activities, whether described in this book or otherwise. Neither the publisher nor the author shall be liable or
assumes any responsibility for any injuries or damages arising herefrom. The fact that an organization or Website if
referred to in this work as a citation and/or a potential source of further information does not mean that the author or
the publisher endorses the information the organization or Website may provide or recommendations it may make.
Further, readers must be aware that the Internet Websites listed in this work may have changed or disappeared
between when this work was written and when it is read.

All trademarks, logos or any other mark such as Vibgyor, USP, Amanda, Golden Bells, Firewall Media, Mercury,
Trinity, Laxmi appearing in this work are trademarks and intellectual property owned by or licensed to Laxmi
Publications, its subsidiaries or affiliates. Notwithstanding this disclaimer, all other names and marks mentioned in
this work are the trade names, trademarks or service marks of their respective owners.
Published in India by

UNIVERSITY SCIENCE PRESS
(An Imprint of Laxmi Publications Pvt. Ltd.)
113, GOLDEN HOUSE, DARYAGANJ,
NEW DELHI 110002, INDIA
Telephone : 91114353 2500, 4353 2501
Fax : 91112325 2572, 4353 2528
[email protected]

C—

ABOUT THE AUTHOR

Prof. Shashi Banzal has been associated with computer field for the past 22 years. She is a leading author and
consultant in the field of computer science and technology.

She has received her Bachelor's degree in Computer Science and Technology from Bhopal University in 1992. She
had a teaching and course development experience in the Department of Information Technology and Computer
Science at a college under Indore University. Prior to this job she was professor and software developer at New
Delhi. In addition, her articles have been published since 1994 in "Telecommunication" magazine, which is circulated
all over India. She has keen interest in new things in computer and web developing fields and she has been
developing websites in ASP on .NET framework, PHP+Mysql and XML. She has written books on C++, Internet and
networking.

ACKNOWLEDGEMENTS

There are a number of individuals who have helped us with the contents of this book. Specifically, I thank Shri Vivek
Banzal (DGM) for enormous support during writing this book.

We are grateful to all our former teaching assistants who helped us in developing exercises, assignments and
systems. There have been a number of friends and colleagues whose comments have led to improvements in the
text. We are also truly indebted to the anonymous outside for their detailed comments and constructive criticism,
which were extremely useful.

We are grateful to our editors, for their enthusiastic support of this project. The production team at Laxmi
Publications Pvt. Ltd. has been great.

Finally, we would like to warmly thank my parents, husband and both kids for providing advice, encouragement and
support at various stages of the preparation of this book. We also thank them for reminding us that there are things
in life beyond writing books. In end, we thank to all staff of Laxmi Publications Pvt. Ltd. for essential support
throughout the arduous process of getting the book into print.

—AUTHOR
Preface to the Second Edition

Data and Computer Network
Communication, Second Edition

by Shashi Banzal
Laxmi Publications © 2015 Citation

Recommend?

Preface to the Second Edition
This book is a complete guide to data and computer network communication. The book covers all networking layers
in detail. First it places emphasis on the physical layer, which has been the high growth area of hardware and
computer networking and communication.

Also at top layer, indeed, many of the recent revolutions in computer networking including the Web, audio and video
streaming and content distribution have taken place at the application layer. In preparation of first edition of this
book, we believed that the application layer would remain the highest "growth area" in the field, both in terms of
research and actual deployment. The intervening years have unquestionably shown this to be true. An early
emphasis on the application layer issues differ from the approaches taken in most other texts, which have only a
small (or nonexistent) amount of material on network applications, their requirements, application layer paradigms
(e.g., clients/server) and the application programming interfaces.
Second, our experience as instructors has been that teaching networking applications near the beginning of the
course is a powerful motivational tool. Students are thrilled to learn about how networking applications work
applications such as email and the web, which most students use on a daily basis. Once a student understands the
applications, the student can then understand the network services needed to support these applications. The
student can then, in turn, examine the various ways in which such services might be provided and implemented in
the upper layers. Covering applications early thus provides motivation for the remainder of the text.

Third, the bottomup approach enables instructors to introduce network hardware development at an early stage.
Students not only see how popular applications and protocols work, but also learn how easy it is to create their own
network applications and applicationlevel protocols. Other introductory computer networking texts do not cover
application development and socket programming. Thus, with the bottomup approach, students get early exposure
to the notions of hardware programming interfaces (APIs), service models, and protocolsimportant concepts that
resurface in all of the subsequent layers.

The partbypart description of the book follows; it is divided into five parts. Part 1 describes the introduction to data
and computer network communication, explores data communication and focuses on computer network as part of an
effective strategy to solve that problem. This part contains two chapters. Part 2 describes the physical layer
structure and it covers six chapters. Part 3 describes the data link layer structure and it contains three
chapters. Part 4 describes the medium access control layer, it contains two chapters. Part 5 describes the local
area network, it contains ten chapters. Part 6 describes all about network layer right from its working and
routing. Part 7 describes flow and congestion control. Part 8 describes transport layer and contains one chapter on
the topic. The last Part 9 contains one last chapter describing all protocols which are used at presentation and
application layer.

The book contains a large number of examples, exercises and figures, which should enhance its suitability for class
room instructions some of the exercises are theoretical and practical in nature and supplement substantially the
main text.

—AUTHOR
Preface to the First Edition

Data and Computer Network
Communication, Second Edition

by Shashi Banzal
Laxmi Publications © 2015 Citation

Recommend?

Preface to the First Edition
Unlike the other computer networking texts, this text is organized in a bottomup manner—that is, it begins at the
physical layer and works its way down towards the application layer. The bottomup approach has several important
benefits. First, it places emphasis on the physical layer, which has been the high "growth area" of hardware and
computer networking and communication. Also at top layer, indeed, many of the recent revolutions in computer
networking—including the Web, audio and video streaming and content distribution—have taken place at the
application layer. In preparing the first edition of this book, we believed that the application layer would remain the
highest growth area in the field, both in terms of research and actual deployment. The intervening years have
unquestionably shown this to be true! An early emphasis on application layer issues differs from the approaches
taken in most other texts, which have only a small (or nonexistent) amount of material on network applications, their
requirements, application layer paradigms (e.g., client/server), and the application programming interfaces. Second,
our experience as instructors has been that teaching networking applications near the beginning of the course is a
powerful motivational tool. Students are thrilled to learn about how networking applications work—application such
as email and the Web, which most students use on a daily basis. Once a student understands the applications, the
student can then understand the network services needed to support these applications. The student can then, in
turn, examine the various ways in which such services might be provided and implemented in the upper layers.
Covering applications early thus provides motivation for the remainder of the text.

Third, the bottomup approach enables instructors to introduce network hardware development at an early stage.
Students not only see how popular applications and protocols work, but also learn how easy it is to create their own
network applications and applicationlevel protocols. Other introductory computer networking texts do not cover
application development and socket programming. Thus, with the bottomup approach, students get early exposure
to the notions of hardware programming interfaces (APIs), service models, and protocols—important concepts that
resurface in all of the subsequent layers.

The PartbyPart description of the book follows, it is divided into five parts:

Part 1, The part describes the Introduction to Data and Computer Network Communication, explores data
communication and focuses on computer network as part of an effective strategy to solve that problem. The part
contains two chapters:

Chapter 1, This chapter focuses on.

Chapter 2, This chapter describes about.

Part 2, The part describes the Physical layer structure and it covers six chapters:

Chapter 3, This chapter explain about data transmission signals.

Chapter 4, This chapter describe about multiplexing techniques and switching.

Chapter 5, This chapter focuses on all transmission medias in details.

Chapter 6, This chapter describe about telephone networks.

Chapter 7, This chapter explain about cable modem and television.

Chapter 8, This chapter focuses on high speed network access such as SONET.

Part 3, The part describes the data link layer structure and it contains three chapters:

Chapter 9, This chapter describe about data link layer.
Chapter 10, This chapter focuses on error control, detection and correction.

Chapter 11, This chapter explain all about data link layer protocol.

Part 4, The part describes the Medium access control layer, it contains two chapters:

Chapter 12, This chapter describe about channels.
Chapter 13, This chapter focuses on medium access control layer.

Part 5, The part describes the local area network, it contains ten chapters:

Chapter 14, This chapter explain about local area network.

Chapter 15, This chapter describe about token ring.

Chapter 16, This chapter focuses on bluetooth.

Chapter 17, This chapter explain about virtual local area network.

Chapter 18, This chapter focuses on wireless transmission and satellite communication.

Chapter 19, This chapter describe about frame relay.

Chapter 20, This chapter explain about ATM.

Chapter 21, This chapter focuses on FDDI.

Chapter 22, This chapter describe about switched multimegabit data service.

Chapter 23, This chapter focuses on ISDN.
Part 6, The part describes all about network layer right from its working and routing:

Chapter 24, This chapter describe about network layer.

Chapter 25, This chapter explain about internetworking.

Chapter 26, This chapter focuses on basics of routing.

Part 7, The part describes flow and congestion control:

Chapter 27, This chapter explain about flow and congestion control.

Chapter 28, This chapter describe about bridge, hub, repeater and gateway.

Part 8, The part describes transport layer:

Chapter 29, This chapter describe about transport layer.

Part 9, The part describes all protocols which are used at presentation and application layer:

Chapter 30, This chapter focuses on presentation and application layer.

The book contains a large number of examples and exercises, which should enhance its suitability for classroom
instructions. Some of the exercises are theoretical in nature and supplement substantially the main text.

—AUTHOR
Data and
Computer Network
Communication,
Second Edition
by Shashi Banzal
Laxmi Publications ©
2015 (804 pages) Citation
ISBN:9789383828395
Taking a precise,
analytical approach,
this book is organized
in a bottomup
manner, beginning at
the physical layer and
working its way down
towards the application
layer to explain
computer networking
and communication.

Recommend?

Table of Contents
Data and Computer Network Communication, Second Edition
Preface to the Second Edition
Preface to the First Edition
Part 1 Computer Communication and Network Basics
Chapter 1 Overview of Computer Communication and Networking
OVERVIEW
COMMUNICATION SYSTEM AND NETWORKING
NEED FOR COMPUTER NETWORK
EVOLUTION OF NETWORK
FUNCTIONS OF DATA NETWORKS
NETWORK CHALLENGES
IMPORTANCE OF NETWORK
NETWORKING BASICS
CABLING BASICS
DATA COMMUNICATION AND COMPUTER NETWORK
MAIN USES OF COMPUTER NETWORKS
COMMUNICATIONS AND DATA TRANSMISSION MODES
DATA TRANSMISSION MODES
SYNCHRONOUS AND ASYNCHRONOUS MODES
CLASSIFYING NETWORK BY AREA
CLASSIFYING THE NETWORK BY TOPOLOGY
DATA TRANSFER IN DIGITAL CIRCUITS
NETWORK FACILITIES
TYPES OF NETWORK ACCESS
SUBNETWORK (SUBNET)
THE INTERNET, INTRANETS AND EXTRANETS
REVIEW QUESTIONS
Chapter 2 Network Models
OVERVIEW
INTRODUCING THE ISO OSI REFERENCE MODEL
OPEN SYSTEMS INTERCONNECTION (OSI) ARCHITECTURE
INTERACTION BETWEEN OSI MODEL LAYERS
CRITICISMS OF THE ISO OSI REFERENCE MODEL
EXAMPLE OF OSI COMMUNICATION
THE TCP/IP REFERENCE MODEL
STANDARDS ORGANIZATIONS
IEEE HISTORY
AIEE, IRE MERGE TO FORM IEEE
INTRODUCTION TO TCP/IP PROTOCOL ARCHITECTURE
NETWORK PROTOCOLS AND STANDARDS
COMMUNICATIONS PROTOCOLS
A PROTOCOL DATA UNIT
PROTOCOL HIERARCHIES
REVIEW QUESTIONS
Part 2 Physical Layer
Chapter 3 Data Transmission Signals
INTRODUCTION TO TRANSMISSION
SIGNAL
DATA TRANSMISSION SIGNALS
ANALOG SIGNAL TYPES
ANALOG VS. DIGITAL TRANSMISSION
NETWORKS
DATA TRANSMISSION CIRCUITS
PARALLEL AND SERIAL DATA
A/D CONVERSION
BANDWIDTH
FREQUENCY
SIGNAL
CARRIER
SAMPLING THEOREM
QUANTIZATION
MODEM
CODECS
MODULATION AND DEMODULATION
MODULATION
PULSE CODE MODULATION
RS232C INTERFACE
RS232C FUNCTIONS
RS449 INTERFACE STANDARD
RS422 AND RS423 INTERFACE STANDARDS
REVIEW QUESTIONS
Chapter 4 Multiplexing and Switching
MULTIPLEXING BASICS
MULTIPLEXING
INVERSE MULTIPLEXER
MULTIPLEXING METHODS
MEDIA SHARING TECHNIQUES
MULTIPLEXING AND SIGNAL CONDITIONING
MULTIPLEXED CHANNELS
MODULATION AND MULTIPLEXING
MODULATION STANDARDS
REVIEW QUESTIONS
Chapter 5 Transmission Media
OVERVIEW
TIMING OF BITS
TRANSMISSION MEDIA
HISTORY OF FIBER OPTIC TECHNOLOGY
PARTS OF A FIBER OPTIC LINK
FIBER OPTIC NETWORK TOPOLOGIES FOR ITS AND OTHER SYSTEMS
BUS NETWORK
STAR NETWORKS
RING NETWORKS
REVIEW QUESTIONS
Chapter 6 Telecommunication
INTRODUCTION
EXAMPLES OF HUMAN (TELE) COMMUNICATIONS
TELEPHONE NETWORK
PUBLIC SWITCHED TELEPHONE NETWORK
ARCHITECTURE AND CONTEXT
IMPORTANCE OF NETWORK IN TELEPHONES
VOICE CHANNELS
DATA CHANNELS
ADSL—ASYMMETRICAL DIGITAL SUBSCRIBER LINE
DIGITAL SWITCH
TELEPHONE STRUCTURE
FUNCTIONS OF A TELEPHONE SET
LOCAL LOOPS
CENTRAL OFFICE
HIERARCHICAL PHONE NETWORKS
CALL ROUTING
TELEPHONE LINE CHARACTERISTICS
ATTENUATION VERSUS FREQUENCY
PROPAGATION DELAY
ENVELOPE DELAY DISTORTION
LINE IMPAIRMENTS
JITTER: AMPLITUDE AND PHASE
MODEMS
TRANSFER RATE VS. PC BUS SPEED
CARRIER
SWITCHING
NETWORKS
INTRODUCTION
SWITCHING SCENARIO
PACKET SWITCHING VERSUS CIRCUIT SWITCHING
MOBILE TELEPHONE SYSTEM
REVIEW QUESTIONS
Chapter 7 Cable Television
OVERVIEW
SYSTEM ARCHITECTURE
CABLE MODEMS
DSL VS. CABLE MODEM COMPARISON
INTERNET OVER CABLE
CABLE MODEMS
PHYSICAL LAYER
DIGITAL SUBSCRIBER LINE (DSL)
REVIEW QUESTIONS
Chapter 8 High Speed Digital Access—SONET
INTRODUCTION TO SONET
APPLICATIONS
TECHNOLOGY
SONET/SDH
SYNCHRONOUS OPTICAL NETWORK (SONET)
OVERVIEW OF SONET
REVIEW QUESTIONS

Part 3 Data Link Layer
Chapter 9 Data Link Layer
DATA LINK LAYER (DLL)
DATA LINK LAYER FUNCTIONS
DIVISION OF DATA LINK LAYER
SERVICE PROVIDED TO NETWORK LAYER
IEEE 802.2 LLC—LOGICAL LINK CONTROL LAYER
SERVICE ACCESS PORTS (SAPS)
TYPES OF LLC OPERATION
CLASSES OF LLC
FRAMING
FRAMING DATA UNITS
THE CONCEPT OF PARITY
CYCLIC REDUNDANCY CHECKS
HAMMING CODES
HUFFMAN CODING
REVIEW QUESTIONS

Chapter 10 Error Control
OVERVIEW
GOBACKN PROTOCOL
STOP AND WAIT ARQ
SELECTIVE REPEAT (ARQ)
FLOW CONTROL
SLIDING WINDOW PROTOCOLS
ONE BIT SLIDING WINDOW PROTOCOL
HDLC—HIGH LEVEL DATA LINK CONTROL
REVIEW QUESTIONS

Chapter 11 Data Link Layer Protocols
OVERVIEW
SIMPLEX STOPAND WAITPROTOCOL
SIMPLEX PROTOCOL FOR A NOISY CHANNEL
REVIEW QUESTIONS

Chapter 12 Channel
OVERVIEW
COMMUNICATION CHANNEL
CHANNEL CAPACITY
CHANNEL CODING: HAMMING DISTANCE
NOISY CHANNEL MODEL
NYQUIST THEOREM
DATA LINK LAYER IN THE INTERNET
PPP DATA FRAMING
PROTOCOL OPERATION
PROTOCOL EFFICIENCY
REVIEW QUESTIONS

Part 4 The Media Access Control Sublayer (MAC)
Chapter 13 Medium Access Control (MAC)
OVERVIEW
MEDIUM ACCESS CONTROL
MAC FRAME FORMAT
CLASSIFICATION OF MAC PROTOCOLS
ALOHA
RANDOM ACCESS PROTOCOLS
NETWORK ACCESS METHODS
CSMA/CD (COLLISION DETECTION)
OTHER ISSUES ABOUT FRAME
EXCEPTION TO THE RULE
WORKING
FORMAT OF A FRAME
WORKING OF EACH FIELD
CARRIERSENSE MULTIPLE ACCESS WITH COLLISION AVOIDANCE (CSMA/CA)
NETWORK INTERFACE CARD
REVIEW QUESTIONS

Part 5 Local Area Network (LAN)
Chapter 14 Ethernet
ETHERNET TECHNOLOGIES
ETHERNET/IEEE 802.3 COMPARISON
ETHERNET CABLING
TYPES OF ETHERNET
THE BASIC ETHERNET FRAME FORMAT
FRAME TRANSMISSION
FLOW CONTROL
FRAME RECEPTION
ETHERNET TOPOLOGIES AND PROTOCOLS
ETHERNET NETWORK ELEMENTS
ETHERNET NETWORK TOPOLOGIES AND STRUCTURES
THE ETHERNET MAC SUBLAYER
1000 MBPS—GIGABIT ETHERNET
IEEE802. 3 PROTOCOL
ETHERNET/IEEE 802.3
REVIEW QUESTIONS
Chapter 15 Token Ring
TOKEN RING/IEEE 802.5
TOKEN RING/IEEE 802.5 COMPARISON
TOKEN RING
CHARACTERISTICS
TOKEN RING HUBS
PHYSICAL CONNECTIONS
TOKEN RING OPERATION
TOKEN PASSING
REVIEW QUESTIONS
Chapter 16 Bluetooth
OVERVIEW
WIRELESS TECHNOLOGY
BLUETOOTH POWER CLASSES
BLUETOOTH SECURITY MODES
BLUETOOTH SECURITY LEVELS
BLUETOOTH SECURITY WEAKNESSES
REVIEW QUESTIONS
Chapter 17 Virtual LAN (VLAN)
OVERVIEW
TELEPHONE TRAFFIC DEFINITIONS
SWITCHING TECHNOLOGY
REVIEW QUESTIONS

Chapter 18 Wireless Transmission
WIRELESS COMMUNICATION
TRANSMISSION MEDIA—UNGUIDED
USES
SATELLITE
IRIDIUM TELECOM SYSTEM
MOBILE TELEPHONE SYSTEM
REVIEW QUESTIONS

Chapter 19 Frame Relay
INTRODUCTION
BACKGROUND OF FRAME RELAY
PRINCIPLES OF FRAME RELAY
FRAME RELAY PROTOCOL DATA UNIT
CONGESTION CONTROL IN FRAME RELAY NETWORK
TRAFFIC ENFORCEMENT
CONGESTION NOTIFICATION
FRAME RELAY VS. X.25
REVIEW QUESTIONS

Chapter 20 ATM (Asynchronous Transfer Mode)
OVERVIEW
A HISTORY OF ATM
SWITCHING
FRAME RELAY
TRANSMISSION
ATM REFERENCE MODEL
ATM LAYER
ADDRESSING
ATM SWITCHING
VIRTUAL CHANNELS
REVIEW QUESTIONS
Chapter 21 Fiber Distributed Data Interface
FIBER DISTRIBUTED DATA INTERFACE(FDDI)
FDDI STATION TYPES
CDDI
DQDB: DISTRIBUTED QUEUE DUAL BUS DEFINED IN IEEE 802.6
REVIEW QUESTIONS
Chapter 22 Integrated Services Digital Network (ISDN)
OVERVIEW
ISDN SERVICES
ISDN—OSI LAYERS
ISDN—OSI MODEL
ISDN PREMISE CONNECTION
X.25
TECHNOLOGY BASICS OF X.25
REVIEW QUESTIONS

Chapter 23 Switched Multimegabit Data Service
INTRODUCTION
SMDS NETWORK COMPONENTS
DISTRIBUTED QUEUE DUAL BUS
SWITCHED MULTIMEGABIT DATA SERVICE (SMDS) OVERVIEW
REVIEW QUESTIONS

Part 6 Network Layer
Chapter 24 Network Layer Overview
NETWORK LAYER
NETWORK LAYER INTERFACE
OVERVIEW OF OTHER NETWORK LAYER ISSUES
NETWORK PROTOCOLS
NETWORK LAYER OVERVIEW
NETWORK LAYER PROTOCOLS
PHYSICAL ADDRESS DETERMINATION
REVERSE ADDRESS RESOLUTION PROTOCOL(RARP)
INTERNET ROUTING—INTERNAL ROUTING TABLES
THE RIP (RFC 1058) PROTOCOL
THE OSPF (RFC 1247) PROTOCOL
REVIEW QUESTIONS

Chapter 25 Internetworking
INTERNETWORKING BASICS
HISTORY OF INTERNETWORKING
INTERNETWORKING CHALLENGES
INTERNETWORKING DEVICE COMPARISON
ADDRESSING
FRAMES, PACKETS AND MESSAGES
KEY ORGANIZATIONS
INTERNET PROTOCOL AND IP ADDRESSING
IP ADDRESSES: NETWORKS AND HOSTS
SUBNET MASK
NETWORK CLASSES
SUBNETTING
DEFAULT GATEWAYS
THE INTERNETWORK PROTOCOL(IP)
ALLOCATION OF IP ADDRESSES
AUTONOMOUS SYSTEMS
INTERNET PROTOCOL
IP ADDRESSES
REVIEW QUESTIONS

Chapter 26 Routing Basics
BASIC ROUTING PROTOCOLS
ROUTING INFORMATION PROTOCOL
NETWORK ROUTING
NETWORK LAYER AND ROUTING CONCEPTS
DIJKSTRA'S ALGORITHM
FLOYD'S ALGORITHM
EXTERIOR ROUTING
OSPF2 PROTOCOL OVERVIEW
RIP
FORMAL ALGORITHM
INTERNET GROUP MANAGEMENT PROTOCOL
INTERNET ROUTING ARCHITECTURE
REVIEW QUESTIONS

Part 7 Flow and Congestion Control
Chapter 27 Flow Control Basics
OVERVIEW
ERRORCHECKING BASICS
PRINCIPLES OF CONGESTION CONTROL
APPROACHES TOWARD CONGESTION CONTROL
ATM ABR CONGESTION CONTROL
CONGESTION AND DEADLOCK
CONGESTION CONTROL
CONGESTION AVOIDANCE AND CONTROL
ENDEND CONGESTION CONTROL: WINDOWBASED CONGESTION CONTROL
ENDEND CONGESTION CONTROL: TCP
NETWORKINDICATED CONGESTION CONTROL
RATEBASED CONGESTION CONTROL
RATE BASED CONGESTION CONTROL: LEAKY BUCKET
CONGESTION CONTROL BY BUFFER PREALLOCATION
CONGESTION AVOIDANCE MECHANISMS
PREALLOCATION OF RESOURCES
DISCARDING PACKETS(NO PREALLOCATION)
ISARITHMIC CONGESTION CONTROL
CONGESTION CONTROL IN MULTICASTING
RSVP—RESOURCE RESERVATION PROTOCOL
RSVP
ROUTING AND SWITCHING
OPERATION OF A ROUTER
INTRODUCTION TO FORWARDING
ROUTING BASICS
ROUTING COMPONENTS
REPEATERS, HUBS, BRIDGES, SWITCHES, ROUTERS AND GATEWAYS
REVIEW QUESTIONS

Chapter 28 Bridge Basics
BRIDGING BASICS
TECHNOLOGY BASICS
BRIDGE METHODOLOGIES
REASONS TO USE A BRIDGE
BRIDGE ADDRESSING
COLLAPSED BACKBONES
TYPES OF BRIDGES
HUBS
GATEWAYS
ADDRESS CLASSES
SUMMARY OF IP ADDRESS CLASSES
SUBNETTING
GATEWAYS/ROUTERS
GATEWAY OWNERSHIP
REVIEW QUESTIONS

Part 8 Transport Layer
Chapter 29 Transport Layer Protocols
TRANSPORT LAYER
TRANSPORT PROTOCOLS
TCP/IP
SENDER/RECEIVER INTERACTION
GENERAL DESCRIPTION OF THE TCP/IP PROTOCOLS
PROTOCOLS OTHER THAN TCP: UDP AND ICMP
DOMAIN SYSTEM
ADDRESS RESOLUTION PROTOCOL
REVERSE ADDRESS RESOLUTION PROTOCOL(RARP)
TRANSPORT LAYER SERVICES AND PRINCIPLE
RELATIONSHIP BETWEEN TRANSPORT AND NETWORK LAYERS
OVERVIEW OF THE TRANSPORT LAYER IN THE INTERNET
ELEMENTS OF TRANSPORT PROTOCOL
TRANSPORT LAYER: MULTIPLEXING
MULTIPLEXING IN TERMS OF NETWORK SERVICES
FLOW CONTROL BASICS
FLOW CONTROL IN THE TRANSPORT LAYER
CRASH RECOVERY
SEQUENCING PROBLEMS IN A NONSEQUENCING NETWORK
CONNECTIONORIENTED TRANSPORTTCP
SUBNETTING A NETWORK
OVERVIEW OF TCP CONGESTION CONTROL
MODELING LATENCY: STATIC CONGESTION WINDOW
STATIC CONGESTION WINDOW
MODELING LATENCY: DYNAMIC CONGESTION WINDOW
CONNECTIONLESS TRANSPORT: UDP
USER DATAGRAM PROTOCOL(UDP)
UDP SEGMENT STRUCTURE
UDP CHECKSUM
REVIEW QUESTIONS

Part 9 Presentation and Application Layer
Chapter 30 Application Layer
A
B
C
D
E
F
G
H
I
J
L
M
N
P
Q
R
S
T
U
V
W
Z

A
B
C
D
E
F
G
H
I
J
L
M
N
P
Q
R
S
T
U
V
W
Z
Appendices
Appendix A
Appendix B
Appendix C
Appendix D
Appendix E
Network Terminology
AB
CE
FI
LO
PS
TX
ZB

Glossary
A
B
C
D
E
F
G
H
I
J
L
M
N
P
Q
R
S
T
U
V
W
Z

Index
A
B
C
D
E
F
G
H
IK
L
M
N
O
P
Q
R
S
T
U
V
WZ

List of Figures
List of Tables
List of Examples
Data and Computer Network
Communication, Second Edition

Data and Computer Network
Communication, Second Edition

by Shashi Banzal
Laxmi Publications © 2015 Citation

Recommend?

Data and Computer Network Communication, Second Edition
Prof. Shashi Banzal
Ex Faculty Member

Deptt. of Computer Science P.M.B. Gujarati Science College
Indore, M.P.

Typeset at ABRO Enterprises
First Edition: 2007; Second Edition: 2015
UDC9737595DATA & COMP NETWORK COMBAN
9789383828395

ISBN 9789383828395

C—

ABOUT THE AUTHOR

Prof. Shashi Banzal has been associated with computer field for the past 22 years. She is a leading author and
consultant in the field of computer science and technology.

ACKNOWLEDGEMENTS

There are a number of individuals who have helped us with the contents of this book. Specifically, I thank Shri Vivek
Banzal (DGM) for enormous support during writing this book.

We are grateful to our editors, for their enthusiastic support of this project. The production team at Laxmi
Publications Pvt. Ltd. has been great.

—AUTHOR
Preface to the Second Edition
This book is a complete guide to data and computer network communication. The book covers all networking layers in
detail. First it places emphasis on the physical layer, which has been the high growth area of hardware and computer
networking and communication.

Also at top layer, indeed, many of the recent revolutions in computer networking including the Web, audio and video
streaming and content distribution have taken place at the application layer. In preparation of first edition of this book, we
believed that the application layer would remain the highest "growth area" in the field, both in terms of research and
actual deployment. The intervening years have unquestionably shown this to be true. An early emphasis on the
application layer issues differ from the approaches taken in most other texts, which have only a small (or nonexistent)
amount of material on network applications, their requirements, application layer paradigms (e.g., clients/server) and the
application programming interfaces.
Second, our experience as instructors has been that teaching networking applications near the beginning of the course is
a powerful motivational tool. Students are thrilled to learn about how networking applications workapplications such as e
mail and the web, which most students use on a daily basis. Once a student understands the applications, the student
can then understand the network services needed to support these applications. The student can then, in turn, examine
the various ways in which such services might be provided and implemented in the upper layers. Covering applications
early thus provides motivation for the remainder of the text.

Third, the bottomup approach enables instructors to introduce network hardware development at an early stage.
Students not only see how popular applications and protocols work, but also learn how easy it is to create their own
network applications and applicationlevel protocols. Other introductory computer networking texts do not cover
application development and socket programming. Thus, with the bottomup approach, students get early exposure to
the notions of hardware programming interfaces (APIs), service models, and protocolsimportant concepts that resurface
in all of the subsequent layers.

The partbypart description of the book follows; it is divided into five parts. Part 1 describes the introduction to data and
computer network communication, explores data communication and focuses on computer network as part of an
effective strategy to solve that problem. This part contains two chapters. Part 2 describes the physical layer structure
and it covers six chapters. Part 3 describes the data link layer structure and it contains three chapters. Part 4 describes
the medium access control layer, it contains two chapters. Part 5 describes the local area network, it contains ten
chapters. Part 6 describes all about network layer right from its working and routing. Part 7 describes flow and
congestion control. Part 8 describes transport layer and contains one chapter on the topic. The last Part 9 contains
one last chapter describing all protocols which are used at presentation and application layer.
The book contains a large number of examples, exercises and figures, which should enhance its suitability for class
room instructions some of the exercises are theoretical and practical in nature and supplement substantially the main
text.

—AUTHOR
Preface to the First Edition
Unlike the other computer networking texts, this text is organized in a bottomup manner—that is, it begins at the
physical layer and works its way down towards the application layer. The bottomup approach has several important
benefits. First, it places emphasis on the physical layer, which has been the high "growth area" of hardware and
computer networking and communication. Also at top layer, indeed, many of the recent revolutions in computer
networking—including the Web, audio and video streaming and content distribution—have taken place at the application
layer. In preparing the first edition of this book, we believed that the application layer would remain the highest growth
area in the field, both in terms of research and actual deployment. The intervening years have unquestionably shown this
to be true! An early emphasis on application layer issues differs from the approaches taken in most other texts, which
have only a small (or nonexistent) amount of material on network applications, their requirements, application layer
paradigms (e.g., client/server), and the application programming interfaces. Second, our experience as instructors has
been that teaching networking applications near the beginning of the course is a powerful motivational tool. Students are
thrilled to learn about how networking applications work—application such as email and the Web, which most students
use on a daily basis. Once a student understands the applications, the student can then understand the network
services needed to support these applications. The student can then, in turn, examine the various ways in which such
services might be provided and implemented in the upper layers. Covering applications early thus provides motivation for
the remainder of the text.

Third, the bottomup approach enables instructors to introduce network hardware development at an early stage.
Students not only see how popular applications and protocols work, but also learn how easy it is to create their own
network applications and applicationlevel protocols. Other introductory computer networking texts do not cover
application development and socket programming. Thus, with the bottomup approach, students get early exposure to
the notions of hardware programming interfaces (APIs), service models, and protocols—important concepts that
resurface in all of the subsequent layers.

The PartbyPart description of the book follows, it is divided into five parts:

Part 1, The part describes the Introduction to Data and Computer Network Communication, explores data communication
and focuses on computer network as part of an effective strategy to solve that problem. The part contains two chapters:

Chapter 1, This chapter focuses on.
Chapter 2, This chapter describes about.

Part 2, The part describes the Physical layer structure and it covers six chapters:

Chapter 3, This chapter explain about data transmission signals.

Chapter 4, This chapter describe about multiplexing techniques and switching.
Chapter 5, This chapter focuses on all transmission medias in details.

Chapter 6, This chapter describe about telephone networks.

Chapter 7, This chapter explain about cable modem and television.

Chapter 8, This chapter focuses on high speed network access such as SONET.

Part 3, The part describes the data link layer structure and it contains three chapters:

Chapter 9, This chapter describe about data link layer.

Chapter 10, This chapter focuses on error control, detection and correction.

Chapter 11, This chapter explain all about data link layer protocol.

Part 4, The part describes the Medium access control layer, it contains two chapters:

Chapter 12, This chapter describe about channels.

Chapter 13, This chapter focuses on medium access control layer.

Part 5, The part describes the local area network, it contains ten chapters:
Chapter 14, This chapter explain about local area network.

Chapter 15, This chapter describe about token ring.

Chapter 16, This chapter focuses on bluetooth.

Chapter 17, This chapter explain about virtual local area network.
Chapter 18, This chapter focuses on wireless transmission and satellite communication.
Chapter 19, This chapter describe about frame relay.

Chapter 20, This chapter explain about ATM.

Chapter 21, This chapter focuses on FDDI.

Chapter 22, This chapter describe about switched multimegabit data service.

Chapter 23, This chapter focuses on ISDN.

Part 6, The part describes all about network layer right from its working and routing:

Chapter 24, This chapter describe about network layer.

Chapter 25, This chapter explain about internetworking.

Chapter 26, This chapter focuses on basics of routing.

Part 7, The part describes flow and congestion control:

Chapter 27, This chapter explain about flow and congestion control.

Chapter 28, This chapter describe about bridge, hub, repeater and gateway.
Part 8, The part describes transport layer:

Chapter 29, This chapter describe about transport layer.

Part 9, The part describes all protocols which are used at presentation and application layer:

Chapter 30, This chapter focuses on presentation and application layer.

—AUTHOR
Part 1: Computer Communication and Network Basics
CHAPTER LIST
Chapter 1: Overview of Computer Communication and Networking
Chapter 2: Network Models
Chapter 1: Overview of Computer Communication and Networking

OVERVIEW
Today, computer is available in many offices and homes and therefore, there is a need to share data and programs
among various computers with the advancement of data communication facilities.
The communication between computers has increased and thus it has extended the power of computer beyond the
computer room. Now, a user sitting at one place can communicate computers of any remote sites through
communication channel.
COMMUNICATION SYSTEM AND NETWORKING
Nowadays, we are fully absorbed by the information age and spending more time communicating and gathering
information through the Internet, it has become necessary to have a working knowledge of the technology behind the
scenes. We are faced with terms like baud rate, modems, cellular phones, TCP/IP, ATM, ISDN etc., and trying to make
decision about our communications needs involving the systems that these terms apply to. In order to develop a useful
working and understanding of this technology requires us to have a good understanding of the basic and background
technology and fundamentals of data communications.

Basic Elements of Communication System
For any communication system to be useful, three main parts are essential. They are:

1. Sending end (Source)

2. Transmission channel (Transmission channel)

3. Receiving end (Receiver)

These parts are shown in figure 1.1. Without these parts we can not do communication.

Figure 1.1: Basic Parts of a Communication System

Speech is the simplest communication man uses, yet it contains all three main parts. When a person speaks (message
generation or sending end), his thoughts are converted to movement which operate vocal cords. The thoughts are in turn
converted to pressure variations in the air. These variations travel outwards (as a signal) through moving air
(transmission channel) and they end up reaching the listener (receiving end). A more generalized view of a
communication system is shown in figure 1.2.

Figure 1.2: Elements of a Communication System

The urgent need to send information quickly over long distances led to telecommunications. Communication over long
distances is no longer a problem, people can communicate instantly in time using modern telephone networks. These
telecommunication networks have made it possible for people to communicate among themselves and when
appropriately modified they can facilitate communication between people and electronic machines (e.g., computer).

The words "tele", "phon", and "graph" are derived from Greek.

Tele—means ‘at a distance’

Phon—means sound or speech

Graph—means writing or drawing.

Therefore, telecommunication means communication at a distance. This can be done through wires called transmission
lines or through atmosphere by a radio link. Other examples include:

Telephone—speaking at a distance

Television—seeing at a distance

Telegraph—writing at a distance.
NEED FOR COMPUTER NETWORK
If your business has more than one computer, chances are you could benefit from networking them. A Local Area
Network (LAN) connects your company's computers, allowing them to share and exchange a variety of information. While
one computer can be useful on its own, several networked computers can be much more useful.

Here are some of the ways a computer network can help your business:
File sharing: Have you ever needed to access a file stored on another computer? A network makes it
easy for everyone to access the same file and prevents a people from accidentally creating different
versions. There is no need to carry a floppy disk.

Printer sharing: If you use a computer, then there are also chances of use of printer. With a network,
several computers can share the same printer and take printouts. Although you might need a more
expensive printer to handle the added workloads, it's still cheaper to use a network printer than to connect
a separate printer to every computer in your office.

Communicate and collaborate: It's hard for people to work together if no one knows what anyone else is
doing. A network allows employees to share files, view other people's work and exchange ideas more
efficiently and easily. In a larger office, you can use email and instant messaging tools to communicate
quickly and to store messages for future reference.

Get organized: A variety of network scheduling software is available that makes it possible to arrange
meetings without constantly checking everyone's schedules. This software usually includes other helpful
features, such as shared address books and todo lists.

Get online: If you have several computers but just one phone line, a network makes using the internet
much easier. Several computers can share one modem to use the same internet connection. Or you can
even install a dedicated highspeed internet connection (such as a cable modem or DSL) for your network.

Save your data: Nowadays, its very important to backup and update your computer data regularly. A
network makes it easier to back up all of your company's data on one set of tapes, CDs or other backup
systems.

Of course this isn't a complete list—once you have a network, you'll probably find many other uses for it. And once you
get used to the benefits of a network, you'll never look at your computer the same way again.
EVOLUTION OF NETWORK
In the past, networks intended to be designed specifically to carry voice, video, or data signals.

The design of voice, video, or data networks differed because of fundamental differences between voice, video, and data
signals, e.g., voice is analog, data is digital in nature.

The telephone network was the first and is by far the largest network supporting wired telephones, fax machines, cellular
phones, cordless phones, answering machines, and modems for data transmission over the phone network.

The TV network uses a combination of coaxial cable, satellite links, and electromagnetic propagation through air to
transmit video (including voice) signals.

The data network uses a variety of transmission media, including the voice and TV networks, to transmit data in the form
of digital signals.
Today, voice and video are becoming increasingly digitized (digital phones, high resolution digital TV) and are
increasingly transmitted by data networks.

Ultimately all voice, video, and data will be digital and will be transmitted by digital (data) networks. The reason is that it
is far more cost effective to build, operate and manage networks that use digital rather than analog signals. (e.g., if the
telephone network were built today, it would be an all digital network). Future data networks will interconnect multimedia
devices capable of handling voice, video, and data.

Evolution of Computer Network
Computer networking started to evolve in the early and mid 1960's with the advent of timesharing.

By 1970 timesharing machines supported networks of local and remote terminals.

These early networks supported timesharing and remote batch processing.
In the latter part of the 1970's, computer to computer network connections were used for load sharing and data
interchange (e.g., early electronic funds transfer).

In the late 1970's and early 1980's, ARPA (Advanced Research Projects Agency) started to test networks for peerto
peer computer communication and terminal support on a national scale.

By the mid 1980's this network became available to the universities and NSF (National Science Foundation) started
funding university connectivity.
Early applications of these networks included electronic mail, downloading large data sets (ftp) and remote access to
computers (telnet).

By the early 1990's, there were several million workstations connected to the Internet and usage was growing very
rapidly.

By the mid 1990's, the Internet was proven reliable technology and began to be commercialized. Companies such as
CompuServe and America online began to provide a number of services using the Internet, including providing Internet
connectivity to personal computers in people's homes.

Today, about 20 million workstations are connected to the commodity Internet and usage continues to increase, resulting
in congestion at peak times of the day over many parts of the network.

The increasing network congestion reduced the value of the Internet for new applications such as webbased distance
education and low latency, high bandwidth connection to and among supercomputers. These applications require support
for QoS (Quality of Service) not available on the current Internet. e.g., bandwidth reservation

Low variance in latency to prevent jerkiness in video transmission. Consequently, leading research institutions are
leading the development of Internet 2, a new high bandwidth network strictly for universities to avoid congestion due to
commercial traffic and one that supports QoS functions required by multimedia and other applications.
FUNCTIONS OF DATA NETWORKS
In general, the vast majority of today's networks are store and forward networks in that stored data is forwarded from its
source to its destination in a series of hops when it is convenient to do so.

While the store and forward technique has several disadvantages (e.g., it is difficult to transmit voice because the
transmission delays are highly variable), it has the advantage that results in more affordable networks and it allows errors
to be detected and corrected through retransmission.
The store and forward concept makes it convenient for networks to be used for:

Exchanging electronic mail

Reading and posting to electronic bulletin boards
Accessing files and information anywhere in the network (e.g., library and web services)

Accessing unique hardware and software resources

Sharing of information (workgroup collaboration)

Such applications provide users with access to needed information and resources when they are needed from wherever
there is telephone or some other means of network access. This is changing how we play, work, communicate, teach,
learn, and conduct business. They provide both new opportunities and new challenges. These challenges will slow the
deployment of networks to a rush from what would otherwise be a stampede.
NETWORK CHALLENGES
Due to address space shortage, IP addresses are limited to 2**32 and efforts are underway to increase this to 2**128.
Enterprises cannot use all 2**24 addresses allowed by a class a license.

Class C addresses are more effective but they increase the size of routing tables in routers, reducing the efficiency of
packet forwarding. Decentralized control allows the network to easily scale. It also threatens to turn it into anarchy where
reliable service is not guaranteed. Security is also an issue. It also means there is no uniform way to do usage
accounting, which is required to guarantee quality of service for emerging applications (e.g., multimedia).
Multimedia (voice, video, and data) applications need realtime performance guarantees such as a certain minimum
bandwidth or bounded latency with small variance (jitter+). These Quality of Service (QoS) performance parameters are
not well supported with the current Internet technologies.

QoS requires signaling to inform all routers in the path about the quality of service parameters for each class of traffic.
New traffic streams must be allowed or denied entry to the network depending upon the current traffic streams and their
QoS parameters. This is difficult to accomplish under decentralized network control.
IMPORTANCE OF NETWORK
Key technology of 20th century is information gathering, processing, and distribution.

World wide telephone networks were constructed. Radio and television networks have reached every corner of the world.
Communications satellites have been launched. In the past 20 years very powerful affordable standalone computers
have been interconnected to form computer networks. These computer networks are changing the way we teach, learn,
do business, and communicate with each other.
NETWORKING BASICS
A network is a group of computers, printers, and other devices that are connected together with cables. Information
travels over the cables, allowing network users to exchange documents & data with each other, print to the same
printers, and generally share any hardware or software that is connected to the network. Each computer, printer, or other
peripheral device that is connected to the network is called a node. Networks can have tens, thousands, or even millions
of nodes or computers. Like most things, networks are assembled according to certain rules. Cabling, for example, has
to be a certain length; each cabling strand can only support a certain amount of network traffic, etc. The rules that
govern how a network is set up are called its topology. The most popular topology in use today is called Ethernet, which
consists of computers and peripherals cabled together in specific ways. Ethernet is relatively inexpensive, easy to set
up and use, and very, very fast.

Ethernet networks are categorized by how fast they can move information. Speed is expressed in megabits per second
(or Mbps), where one "bit" is equal to 1/8th of a character, letter, or number. There are currently two Ethernet speed
categories. Standard Ethernet operates at a fast 10Mbps, which is quick enough for most networking tasks. Fast
Ethernet, by contrast, races along at a blistering 100Mbps, making it ideal for desktop video, multimedia, and other
speedhungry applications. The new technology behind fast Ethernet, which was introduced in the beginning of 1995, is
not readily compatible with standard Ethernet. Making the two "talk" with each other requires special equipment and
some knowledge of Internet working. If you're building your first network, decide whether to go with standard or fast
Ethernet before you begin shopping around for network hardware and software. Unless you plan on using video,
multimedia, or heavy graphics software, plan on using standard Ethernet.
CABLING BASICS
In network, cabling is very important.

The two most popular types of network cabling are twistedpair (also known as 10BaseT) and thin coax (also known as
10Base2). 10BaseT cabling looks like ordinary telephone wire, except that it has 8 wires inside instead of 4. Thin coax
looks like the copper coaxial cabling that's often used to connect a VCR to a TV set.
Thin coax and 10BaseT can both be used exclusively or together, depending on the type of network that you're putting
together. Small networks, for example, may want to use 10BaseT cabling by it, because it's inexpensive, flexible, and
ideal for going short distances. Larger networks (usually with 10 or more computers) may use a thin coax backbone with
small clusters of 10BaseT cabling that branch off from it at regular intervals.
DATA COMMUNICATION AND COMPUTER NETWORK
Today, computer is available in many offices, factories and homes and therefore, there is a need to share data and
programs among various computers with the advancement of data communication facilities. The communication between
computers has increased and thus it has extended the power of computer beyond the computer room. Now a user sitting
at one place can communicate computers of any remote sites through communication channel.

Data Communication
Data Communications is the transfer of data or information between a source and a receiver. The source transmits the
data and the receiver receives it. The actual generation of the information is not part of data communications nor is the
resulting action of the information at the receiver. Data Communication is interested in the transfer of data, the method of
transfer and the preservation of the data during the transfer process.

In local area networks, we are interested in "connectivity", connecting computers together to share resources. Even
though the computers can have different disk operating systems, languages, cabling and locations, they still can
communicate to one another and share resources.

The purpose of data communications is to provide the rules and regulations that allow computers with different disk
operating systems, languages, cabling and locations to share resources. The rules and regulations are called protocols
and standards in data communications.

We all are acquainted with some sorts of communication in our day to day life. For communication of information and
messages we use telephone and postal communication systems. Similarly, data and information from one computer
system can be transmitted to other systems across geographical areas. Thus, data transmission is the movement of
information using some standard methods. These methods include electrical signals carried along a conductor, optical
signals along an optical fibers and electromagnetic areas.

Suppose a manager has to write several letters to various clients. First he has to use his PC and word processing
package to prepare his letter. If the PC is connected to the entire client's PCs through network, he can send the letters to
all the clients within minutes. Thus, irrespective of geographical areas, if PCs are connected through communication
channel, the data and information, computer files and any other program can be transmitted to other computer systems
within seconds. The modern form of communication like email and Internet is possible only because of computer
networking.

The distance over which data moves within a computer may vary from a few thousandths of an inch, as is the case
within a single IC chip, to as much as several feet along the backplane of the main circuit board. Over such small
distances, digital data may be transmitted as direct, twolevel electrical signals over simple copper conductors. Except
for the fastest computers, circuit designers are not very concerned about the shape of the conductor or the analog
characteristics of signal transmission.

Frequently, however, data must be sent beyond the local circuitry that constitutes a computer. In many cases, the
distances involved may be enormous. Unfortunately, as the distance between the source of a message and its
destination increases, accurate transmission becomes increasingly difficult. This results from the electrical distortion of
signals traveling through long conductors, and from noise added to the signal as it propagates through a transmission
medium. Although some precautions must be taken for data exchange within a computer, the biggest problems occur
when data is transferred to devices outside the computer's circuitry. In this case, distortion and noise can become so
severe that information is lost.

Data communications concerns the transmission of digital messages to devices external to the message source.
"External" devices are generally thought of as being independently powered circuitry that exists beyond the chassis of a
computer or other digital message source. As a rule, the maximum permissible transmission rate of a message is
directly proportional to signal power and inversely proportional to channel noise. It is the aim of any communications
system to provide the highest possible transmission rate at the lowest possible power and with the least possible noise.

Criteria of Data Communication Network
The major criteria that a data communication network must meet are:

1. Performance

2. Consistency

3. Reliability

4. Recovery

5. Security

1 Performance
Performance is defined as the rate of transference of errorfree data. It is measured by the response time. Response
time is the elapsed time between the end of an inquiry and the beginning of a response, e.g., requesting a file transfer
and starting the file transfer. Factors that affect response time are:

Number of Users: The more users are on a network, the slower the network will run.

Transmission Speed: The speed that the data will be transmitted at measured in bits per second (bps).

Media Type: The type of physical connection used to connect nodes together.

Hardware Type: Slow computers such as XT, or fast ones such as Pentiums.

Software Program: How well is the network operating system (NOS) written.

2 Consistency
Consistency is the predictability of response time and accuracy of data.

Users prefer to have consistent response times; they develop a feel for normal operating conditions. For example, if the
"normal" response time is 3 seconds for printing to a network printer but a response time of over 30 seconds occurs, we
know that there is a problem in the system.

Accuracy of data determines if the network is reliable! If a system loses data, then the users will not have confidence in
the information and will often not use the system.

3 Reliability
Reliability is the measure of how often a network is usable. MTBF (Mean Time between Failures) is a measure of the
average time a component is expected to operate between failures, and is normally provided by the manufacturer. A
network failure can be caused by a problem with the hardware, the data carrying medium, or the network operating
system.

4 Recovery
Recovery is the network's ability to return to a prescribed level of operation after a network failure. This level is where the
amount of lost data is nonexistent or at a minimum. Recovery is based on having backup files.

5 Security
Security is the protection of hardware, software and data from unauthorized access. Restricted physical access to
computers, password protection, limiting user privileges and data encryption are common security methods. Antivirus
monitoring programs to defend against computer viruses are also a security measure.

Basic Components of Communication
Three basic components are required for communication. Followings are basic components of the communication:

Source: The transmitter of data. Examples are: Terminals, Computers, Mainframes

Medium: The communications stream through which the data is being transmitted. Examples are: Cabling,
Microwave, Fiber optics, Radio Frequencies (RF), Infrared Wireless.

Receiver: The receiver of the data transmitted. Examples are: Printers, Terminals, Mainframes,
Computers.

Figure 1.3: Basic Components
Figure 1.4: Source Medium Receiver

DCE: The interface between the source and the medium, and the medium and the receiver are called the
DCE (Data Communication Equipment) and are a physical piece of equipment.

DTE: Data terminal equipment is the telecommunications name given to the source and receiver's
equipment.

An example of this would be your PC dialing into a BBS (Bulletin Board System):

Figure 1.5: Bulletin Board System
MAIN USES OF COMPUTER NETWORKS
The main purpose of computers is to enable people to exchange data and information. This can take the form of
electronic mail (email) or Intra and Extranets. It is the use of these resources that can decide whether a computer
network is an asset or a problem for the operators.

On the most basic level computer networks are great to share resources, such as printers and storage space. On the
most advanced level computer networks can carry video data for remote video conferencing for example.

Business Use
In the business context computer networks can be a real asset. This includes storage and retrieval of information,
reducing the need for paper traffic and record keeping in paper format, as well as rationalizing the time that staff spends
producing correspondence and accounts. The big danger is that the real purpose of a computer network is not fully
assessed resulting in an installation and maintenance plan that is not adequate for its use.

In addition, if the installation of a new computer network is not fully supported by the staff, then the network itself is may
not be taken full advantage of, and record keeping becomes a mixture between digital and paper based records.
In a business context the following uses are typical. They are listed in ascending order of complexity:

Share resources such as printer.

Share data on a central server, where all data is collected.

Communicate via email.

Exchange information via an internal network.

Allow all staff access to the Internet.

Use the network and the Internet to allow staff access to business data from anywhere in the world.
Integrate the whole business operation into a networked operation, including sales activity, stock holding, quotations,
ordering raw materials, control the production process, process invoices, process all the accounts, analyze business
performance, quality control.

Each stage introduces an extra layer of complexity to the operation and an extra level of staff expertise and acceptance.
In addition to the sheer network installation each level of functionality is combined with a suite of software that needs to
be configured once and kept running subsequently. One example of how complicated things become on the simplest
level is access and sharing of data. A detailed plan and assessment of who should have access to what kind of data,
and who can change and delete data needs to be drawn up and implemented. From a technical point of view this is very
simple, from an organizational point of view this may be a serious difficulty.

With the adoption of a new procedure within the business and the learning of the new processes by the staff, the network
becomes ever more mission critical. This means, that if a problem occurs, the operation of the business could be
seriously disrupted. To avoid major losses, for example, consider the introduction and purchase of adequate data backup
procedures.

And this also means that careful planning is required to make sure that the necessary maintenance of the system is
always kept up. Usually the cost of the system and ongoing maintenance is offset by the productivity gains of the
improved operation of the business.

It is strongly advisable to make a full assessment of the cost and benefits that the introduction of the new procedure
entails and to argue the introduction of a new process in terms of a strict business case.

Very often such undertakings fail because of a communication gap between the technical personnel and engineers who
carry out the installation, and the management and staff of the company that takes such a new system on.

Overall, especially in small to medium sized businesses, the strict assessment of cost and benefit is not carried leading
to serious problems.

There is also the tendency to overcomplicate the procedure in software terms. Software can in theory do anything you
like (as long as it is an algorithmic process), but different offtheshelf packages may not interact with each other very
well. It is impossible to foresee problems of this kind without the detailed input by experienced engineers.

Scientific Use
Within science, computer networks were used very early on to share resources and exchange data. This goes back to
the beginnings of the Internet, and in fact, the Internet as we know it today was created within a scientific laboratory.
It is standard procedure to use networks to share data and use remote computers to carry out large computations. This
includes the use of supercomputers in various locations. As an example consider the use of a supercomputer located in
Norway by researchers in the UK. To the operator the physical distance is immaterial. Data nowadays can be transferred
very quickly from remote computers.

In addition to the "outsourcing" of computing power, the old system of using one powerful machine within a department to
carry out large computations, and many small hosts for the creation of reports is very common.

Research is underway to examine the way in which networked computers can be used for the parallel processing of large
computations. The ultimate aim is to produce a system that would use all the resources available on a computer network
(and even the Internet) as a computing resource.

The use of computers connected via a network for parallel computations is a nontrivial problem. Many researchers are
developing procedures for data transfer and data handling, as well as the adaptation of algorithms for given tasks to
parallel processing.

Parallel computations are also carried out by large machines that contain more than one processor. There are two types:

Processor with local memory

Processors with shared memory

Conceptually the difference between parallel computers and computers on a network is very small. Both exchange data,
both have separate memory per processor.

Here, we now need to make a distinction between processor and computer:

A computer is the combination of a processor, memory and peripheral devices.

A processor is an integrated circuit chip in which the processing is takes place. Without peripheral devices
a processor cannot work.

For scientific use it is inconceivable nowadays not to be using a computer network. Interestingly, each laboratory has a
technician who is dedicated to the maintenance of the computer network and the safeguarding of data through regular
backups. Businesses ought to take notice of this.
COMMUNICATIONS AND DATA TRANSMISSION MODES

Serial and Parallel Communication Modes
In addition to the different models of communications, there are several methods in which data can be transmitted
between source and destination nodes. So, there are two types of communication modes,

1. Serial communications

2. Parallel communications.
Most digital messages are vastly longer than just a few bits. Because it is neither practical nor economic to transfer all
bits of a long message simultaneously, the message is broken into smaller parts and transmitted sequentially. Bitserial
transmission conveys a message one bit at a time through a channel. Each bit represents a part of the message. The
individual bits are then reassembled at the destination to compose the message. In general, one channel will pass only
one bit at a time. Thus, bitserial transmission is necessary in data communications if only a single channel is available.
Bitserial transmission is normally just called serial transmission and is the chosen communications method in many
computer peripherals. Byteserial transmission conveys eight bits at a time through eight parallel channels. Although the
raw transfer rate is eight times faster than in bitserial transmission, eight channels are needed, and the cost may be as
much as eight times higher to transmit the message. When distances are short, it may nonetheless be both feasible and
economic to use parallel channels in return for high data rates. The popular centronics printer interface is a case where
byteserial transmission is used. As another example, it is common practice to use a 16bitwide data bus to transfer
data between a microprocessor and memory chips; this provides the equivalent of 16 parallel channels. On the other
hand, when communicating with a timesharing system over a modem, only a single channel is available, and bitserial
transmission is required. This figure illustrates these ideas:

Figure 1.6: Communication Modes

The baud rate refers to the signaling rate at which data is sent through a channel and is measured in electrical transitions
per second. In the EIA232 serial interface standard, one signal transition, at most, occurs per bit, and the baud rate and
bit rate are identical. In this case, a rate of 9600 baud corresponds to a transfer of 9,600 data bits per second with a bit
period of 104 microseconds (1/9600 sec.). If two electrical transitions were required for each bit, as is the case in non
returntozero coding, then at a rate of 9600 baud, only 4800 bits per second could be conveyed. The channel efficiency
is the number of bits of useful information passed through the channel per second. It does not include framing,
formatting, and error detecting bits that may be added to the information bits before a message is transmitted, and will
always be less than one.

Figure 1.7: Baud Rate

The data rate of a channel is often specified by its bit rate (often thought erroneously to be the same as baud rate).
However, an equivalent measure channel capacity is bandwidth. In general, the maximum data rate a channel can
support is directly proportional to the channel's bandwidth and inversely proportional to the channel's noise level.

A communications protocol is an agreedupon convention that defines the order and meaning of bits in a serial
transmission. It may also specify a procedure for exchanging messages. A protocol will define how many data bits
compose a message unit, the framing and formatting bits, any errordetecting bits that may be added and other
information that governs control of the communications hardware. Channel efficiency is determined by the protocol
design rather than by digital hardware considerations. Note that there is a tradeoff between channel efficiency and
reliability—protocols that provide greater immunity to noise by adding errordetecting and correcting codes must
necessarily become less efficient.
DATA TRANSMISSION MODES
Three different modes of data transmission are used for communication purposes, each responds to a specific type of
circuit: simplex, halfduplex, and fullduplex. These modes specify the protocols that sending and receiving nodes follow
when transferring data.

1 Simplex
In simplex mode the communication can take place in one direction. The receiver receives the signal from the
transmitting device. In this mode the flow of information is unidirectional. Hence, it is rarely used for data
communication.

Figure 1.8

2 HalfDuplex
In halfduplex mode the communication channel is used in both directions, but only in one direction at a time. Thus, a
halfduplex line can alternately send and receive data.

Figure 1.9

3 FullDuplex
In fullduplex the communication channel is used in both directions at the same time. Use of fullduplex line improves the
efficiency as the line turnaround time required in halfduplex arrangement is eliminated. Example of this mode of
transmission is the telephone line.

Figure 1.10
SYNCHRONOUS AND ASYNCHRONOUS MODES
In serial transmission, a receiver must be informed when a complete unit of data has been transmitted. Obviously,
without some way of identifying the beginning (or ending) of a character, the transmitted data would be indecipherable,
resulting in some sort of communication breakdown.

Two methods can be employed to resolve this problem. They are as:

Asynchronous Serial Transmission
We can synchronize the sending and receiving nodes so that the receiving node always knows when a new character is
being sent or we can insert within the bit stream special "start" and "stop" bits. This technique is called asynchronous
communication.

Because no signal lines are used to convey clock (timing) information, this method group's data together into a sequence
of bits (fiveeight), then prefixes them with a start bit and a stop bit.

The purpose of the start and stop bits was introduced for the old electromechanical teletypewriters. These used motors
driving cams which actuated solenoids which sampled the signal at specific time intervals. The motors took a while to
get up to speed, thus by prefixing the first bit with a start bit, this gave time for the motors to get up to speed, and thus
acted as a reference point for the start of the first bit.

At the end of the character sequence, a stop bit was used to allow the motors/cams etc to get back to normal, in
addition, it was also need to fill in time in case the character was an end of line, when the teletypewriter would need to go
to the beginning of a newline. Without the stop character, the machine could not complete this before the next character
arrived.

Figure 1.11: Asynchronous Signal showing Start and Stop Bits

Asynchronous Protocols
Asynchronous systems send data bytes between the sender and receiver. Each data byte is preceded with a start bit,
and suffixed with a stop bit. These extra bits serve to synchronize the receiver with the sender.

Transmission of these extra bits (2 per byte) reduce data throughput. Synchronization is achieved for each character
only. When the sender has no data to transmit, the line is idle and the sender and receiver are not in synchronization.
Asynchronous protocols are suited for low speed data communications.

Synchronous Serial Transmission
In synchronous transmission, the line idle state is changed to a known character sequence (7E), which is used to
synchronize the receiver to the sender. The start and stop bits are removed, and each character is combined with others
into a data packet. The data packet is prefixed with a header field, and suffixed with a trailer field which includes a
checksum value (used by the receiver to check for errors in sending).

Figure 1.12: Synchronous Data

The header field is used to convey address information (sender and receiver), packet type and control data. The data
field contains the users data (if it can't fit in a single packet, then use multiple packets and number them) or control data.
Generally, it has a fixed size. The tail field contains checksum information which the receiver uses to check whether the
packet was corrupted during transmission.

Synchronous Protocols
Synchronous protocols involve sending timing information along with the data bytes, so that the receiver can remain in
synchronization with the sender. When the sender has no data to transmit, the sender transmits idle flags (a sequence of
alternating 0's and 1's) to maintain sender/receiver synchronization. Data bytes are packaged into small chunks called
packets, with address fields being added at the front (header) and checksums at the rear of the packet. There are two
main types of synchronous protocols used today, character orientated (bisync) and bit orientated (hdlc).

Asynchronous vs. Synchronous Transmission
Serialized data is not generally sent at a uniform rate through a channel. Instead, there is usually a burst of regularly
spaced binary data bits followed by a pause, after which the data flow resumes. Packets of binary data are sent in this
manner, possibly with variablelength pauses between packets, until the message has been fully transmitted. In order for
the receiving end to know the proper moment to read individual binary bits from the channel, it must know exactly when a
packet begins and how much time elapses between bits. When this timing information is known, the receiver is said to
be synchronized with the transmitter, and accurate data transfer becomes possible. Failure to remain synchronized
throughout a transmission will cause data to be corrupted or lost.

Two basic techniques are employed to ensure correct synchronization. In synchronous systems, separate channels are
used to transmit data and timing information. The timing channel transmits clock pulses to the receiver. Upon receipt of
a clock pulse, the receiver reads the data channel and latches the bit value found on the channel at that moment. The
data channel is not read again until the next clock pulse arrives. Because the transmitter originates both the data and the
timing pulses, the receiver will read the data channel only when told to do so by the transmitter (via the clock pulse), and
synchronization is guaranteed.

Techniques exist to merge the timing signal with the data so that only a single channel is required. This is especially
useful when synchronous transmissions are to be sent through a modem. Two methods in which a data signal is self
timed are nonreturntozero and biphase Manchester coding. These both refer to methods for encoding a data stream into
an electrical waveform for transmission.

In asynchronous systems, a separate timing channel is not used. The transmitter and receiver must be preset in
advance to an agreedupon baud rate. A very accurate local oscillator within the receiver will then generate an internal
clock signal that is equal to the transmitters within a fraction of a percent. For the most common serial protocol, data is
sent in small packets of 10 or 11 bits, eight of which constitute message information. When the channel is idle, the
signal voltage corresponds to a continuous logic ‘1’. A data packet always begins with a logic ‘0’ (the start bit) to signal
the receiver that a transmission is starting. The start bit triggers an internal timer in the receiver that generates the
needed clock pulses. Following the start bit, eight bits of message data are sent bit by bit at the agreed upon baud rate.
The packet is concluded with a parity bit and stop bit. One complete packet is illustrated as in figure 1.13.
Figure 1.13: Asynchronous Systems

The packet length is short in asynchronous systems to minimize the risk that the local oscillators in the receiver and
transmitter will drift apart. When highquality crystal oscillators are used, synchronization can be guaranteed over an 11
bit period. Every time a new packet is sent, the start bit resets the synchronization, so the pause between packets can
be arbitrarily long. Note that the EIA232 standard defines electrical, timing, and mechanical characteristics of a serial
interface. However, it does not include the asynchronous serial protocol shown in the previous figure, or the ASCII
alphabet.

Digital and Analog Transmission
Data is transmitted from one point to another point by means of electrical signals that may be in digital and analog form.
So one should know the fundamental difference between analog and digital signals. In analog signal the transmission
power varies over a continuous range with respect to sound, light and radio waves. On the other hand, a digital signal
may assume only discrete set of values within a given range. Examples are computer and computer related equipment.
Analog signal is measured in volts and its frequency in hertz (Hz). A digital signal is a sequence of voltage represented
in binary form. When digital data are to be sent over an analog form the digital signal must be converted to analog form.
So, the technique by which a digital signal is converted to analog form is known as modulation. And the reverse process,
that is the conversion of analog signal to its digital form, is known as demodulation. The device, which converts digital
signal into analog, and the reverse, is known as modem.

Figure 1.14: Analog Signal
CLASSIFYING NETWORK BY AREA
Communication networks are usually defined by their size and complexity. Here, we can distinguish four main types:

Figure 1.15: Digital Signal

1 Small Area Network (SAN)
The small area network is often called "the last mile" and it refers to the last mile of analog phone line that goes from the
Central Office (CO) to your house.

Figure 1.16: Small Area Network

Typical small area network protocols are:

Voice lines

Modem connections 56 kbps

ISDN (Integrated Services Digital Network) 2 × 64 kbps digital lines

ADSL (Asymmetrical Digital Subscriber Line) up to 8 Mbps

Cable Modems up to 30 Mbps

Cable modems are not part of the local loop but do fall in the category of "the last mile" or how to get high speed digital
communication to the premise (home). It would incredibly expensive to replace the existing cabling structure. All of these
protocols are used to overcome the existing cabling limitations in the local loop and provide high speed digital data
transmission. The existing cabling was designed for voice communications and not digital.

2 Local Area Network (LAN)
These networks connect computer equipment and other terminals distributed in a localized area, e.g., a university
campus, factory, office. The connection is usually a cable or fiber, and the extent of the cable defines the LAN (Local
Area Networks).
A network is any collection of independent computers that communicate with one another over a shared network
medium. LANs are networks usually confined to a geographic area, such as a single building or a college campus. LANs
can be small, linking as few as three computers, but often link hundreds of computers used by thousands of people. The
development of standard networking protocols and media has resulted in worldwide proliferation of LANs throughout
business and educational organizations.
Figure 1.17: Local Area Network

The components used by LANs can be divided into cabling standards, hardware and protocols.
Examples of cabling standards used on LANs are:

Cat 3, 4 and 5 cables

IBM type 19 cabling standards

EIA568A and 568B

Ethernet cabling standards: IEEE 802.3 (10Base5), IEEE 802.3a (10Base2), IEEE 802.3i
(10BaseT)

Unshielded Twisted Pair (UTP)

Shielded Twisted Pair (STP)

Connectors: RJ45, RJ11, hermaphroditic connectors, RS232, DB25, BNC, TEE

Examples of hardware devices are:

Network Interface Cards NICs

Repeaters

Ethernet hubs or multiport repeaters

Token ring MultiStation Access Units (MSAUs), Control Access Units (CAUs) and Lobe
Access Modules (LAMs)

Bridges

Brouters

Routers

Gateways

Print servers

File servers

Switches

Examples of LAN protocols are:

Ethernet frame types: Ethernet_II, Ethernet_SNAP, Ethernet_802.2, Ethernet_802.3

Media Access Control layer (MAC layer)

Token ring: IBM and IEEE 802.5

Logical Link Control Layer (LLC) IEEE 802.2

TCP/IP

SMB, NetBIOS and NetBeui

IPX/SPX

Fiber Distributed Data Interchange (FDDI)

Asynchronous Transfer Mode (ATM)
3 Metropolitan Area Network (MAN)
Metropolitan Area Networks (MANs) are used to interconnect LANs that are spread around, say, a town or city. This kind
of network is a high speed network using optical fiber connections.

Figure 1.18: Metropolitan Area Networks

The main criteria for a MAN are that the connection between LANs is through a local exchange carrier (the local phone
company). The protocols that are used for MANs are quite different from LANs except for ATM which can be used for
both under certain conditions.

Examples of MAN protocols are:

RS232, V35

X.25 (56kbps), PADs

Frame relay (up to 45 Mbps), FRADs

Asynchronous Transfer Mode (ATM)

ISDN (Integrated Services Digital Network) PRI and BRI

Dedicated T1 lines (1.544 Mbps) and fractional T1

T3 (45 Mbps) and OC3 lines (155 Mbps)

ADSL (Asymmetrical Digital Subscriber Line) up to 8 Mbps

xDSL (many different types of Digital Subscriber Lines).

4 Wide Area Network (WAN)
These networks connect computers and other terminals over large distances. They often require multiple communication
connections, including microwave radio links and satellite.

Often a network is located in multiple physical places. Wide area networking combines multiple LANs that are
geographically separate. This is accomplished by connecting the different LANs using services such as dedicated
leased phone lines, dialup phone lines (both synchronous and asynchronous), satellite links, and data packet carrier
services. Wide area networking can be as simple as a modem and remote access server for employees to dial into, or it
can be as complex as hundreds of branch offices globally linked using special routing protocols and filters to minimize
the expense of sending data sent over vast distances.

Figure 1.19: Wide Area Network

The main difference between a MAN and a WAN is that the WAN uses long distance carriers. Otherwise the same
protocols and equipment are used as a MAN.
CLASSIFYING THE NETWORK BY TOPOLOGY
A network topology describes the configuration of a network (how the network components are connected together).
There are five main topologies.

1 Star
The star topology uses a central hub through which all components are connected. In a computer network, the central
hub is the host computer, and at the end of each connection is a terminal.

A star network uses a significant amount of cable, each terminal is wired back to the central hub, even if two terminals
are side by side several hundred meters away from the host. All routing decisions are made by the central hub, and all
other workstations can be simple.

An advantage of the star topology is failure in one of the terminals does not affect any other terminal, however; failure of
the central hub affects all terminals.

This type of topology is frequently used to connect terminals to a large timesharing host computer.

Figure 1.20: Star Topology

2 Ring
The ring topology connects workstations in a closed loop. Each terminal is connected to two other terminals (the next
and the previous), with the last terminal being connected to the first. Data is transmitted around the ring in one direction
only; each station passing on the data to the next station till it reaches its destination.

Faulty workstations can be isolated from the ring. When the workstation is powered on, it connects itself into the ring.
When power is off, it disconnects itself from the ring and allows the information to bypass the workstation.

Figure 1.21: Ring Topology

Information travels around the ring from one workstation to the next. Each packet of data sent on the ring is prefixed by
the address of the station to which it is being sent to. When a packet of data arrives, the workstation checks to see if the
packet address is the same as its own. If it is, it grabs the data in the packet. If the packet does not belong to it, it
sends the packet to the next workstation in the ring.

Ring systems use 4 pair cables (separate send/receive). The common implementation of this topology is token ring. A
break in the ring causes the entire network to fail. Individual workstations can be isolated from the ring.
3 Bus
The bus topology connects workstations using a single cable. Each workstation is connected to the next workstation in a
point to point fashion. All workstations connect to the same cable.

If one workstation goes faulty, all workstations are affected. Workstations share the same cable for the sending and
receiving of information. The cabling costs of bus systems are the least of all the different topologies. Each end of the
cable is terminated using a special terminator.

The common implementation of this topology is Ethernet. A message transmitted by one workstation is heard by all the
other workstations.

Figure 1.22: Bus Topology

4 Mesh
The mesh topology connects all computers to each other. The cable requirements are high, but there are redundant
paths built in. Any failure of one computer allows all others to continue, as they have alternative paths to other
computers.

Figure 1.23: Mesh Topology

Mesh topologies are used in critical connection of host computers (typically telephone exchanges). Alternate paths allow
each computer to balance the load to other computer systems in the network by using more than one of the connection
paths available.

5 Tree
A tree topology combines characteristics of linear bus and star topologies. It consists of groups of starconfigured
workstations connected to a linear bus backbone cable. Tree topologies allow for the expansion of an existing network,
and enable schools to configure a network to meet their needs.

Advantages of a Tree Topology
Pointtopoint wiring for individual segments.

Supported by several hardware and software venders.

Disadvantages of a Tree Topology
Overall length of each segment is limited by the type of cabling used.

If the backbone line breaks, the entire segment goes down.

More difficult to configure and wire than other topologies.
DATA TRANSFER IN DIGITAL CIRCUITS
Data is typically grouped into packets that are 8, 16, or 32 bits long, and passed between temporary holding units called
registers. Data within a register is available in parallel because each bit exits the register on a separate conductor. To
transfer data from one register to another, the output conductors of one register are switched onto a channel of parallel
wires referred to as a bus. The input conductors of another register, which is also connected to the bus, capture the
information.

Following a data transaction, the content of the source register is reproduced in the destination register. It is important to
note that after any digital data transfer, the source and destination registers are equal; the source register is not erased
when the data is sent.

Figure 1.24: Digital Data Transfer

The transmit and receive switches shown above are electronic and operate in response to commands from a central
control unit. It is possible that two or more destination registers will be switched on to receive data from a single source.
However, only one source may transmit data onto the bus at any time. If multiple sources were to attempt transmission
simultaneously, an electrical conflict would occur when bits of opposite value are driven onto a single bus conductor.
Such a condition is referred to as a bus contention. Not only will a bus contention result in the loss of information, but it
also may damage the electronic circuitry. As long as all registers in a system are linked to one central control unit, bus
contentions should never occur if the circuit has been designed properly. Note that the data buses within a typical
microprocessor are fundamentally halfduplex channels.

Transmission over Short Distances (<2 Feet)
Whenever the source and destination registers are part of an integrated circuit (within a microprocessor chip, for
example), they are extremely close (thousandths of an inch). Consequently, the bus signals are at very low power levels,
may traverse a distance in very little time, and are not very susceptible to external noise and distortion. This is the ideal
environment for digital communications. However, it is not yet possible to integrate all the necessary circuitry for a
computer (i.e., CPU, memory, disk control, video and display drivers, etc.) on a single chip. When data is sent offchip to
another integrated circuit, the bus signals must be amplified and conductors extended out of the chip through external
pins. Amplifiers may be added to the source register.
Bus signals that exit microprocessor chips and other VLSI circuitry are electrically capable of traversing about one foot
of conductor on a printed circuit board, or less if many devices are connected to it. Special buffer circuits may be added
to boost the bus signals sufficiently for transmission over several additional feet of conductor length, or for distribution to
many other chips such as memory chips.

Figure 1.25: Transmission over Short Distances (<2 feet)
Transmission over Medium Distances (<20 Feet)
Computer peripherals such as a printer or scanner generally includes mechanisms that cannot be situated within the
computer itself. Our first thought might be just to extend the computer's internal buses with a cable of sufficient length to
reach the peripheral. Doing so, however, would expose all bus transactions to external noise and distortion even though
only a very small percentage of these transactions concern the distant peripheral to which the bus is connected.

If a peripheral can be located within 20 feet of the computer, however, relatively simple electronics may be added to
make data transfer through a cable efficient and reliable. To accomplish this, a bus interface circuit is installed in the
computer:

Figure 1.26: Transmission Over Medium Distances (<20 feet)

It consists of a holding register for peripheral data, timing and formatting circuitry for external data transmission, and
signal amplifiers to boost the signal sufficiently for transmission through a cable. When communication with the
peripheral is necessary, data is first deposited in the holding register by the microprocessor. This data will then be
reformatted, sent with errordetecting codes, and transmitted at a relatively slow rate by digital hardware in the bus
interface circuit. In addition, the signal power is greatly boosted before transmission through the cable. These steps
ensure that the data will not be corrupted by noise or distortion during its passage through the cable. In addition, because
only data destined for the peripheral is sent, the partyline transactions taking place on the computer's buses are not
unnecessarily exposed to noise.

Data sent in this manner may be transmitted in byteserial format if the cable has eight parallel channels (at least 10
conductors for halfduplex operation), or in bitserial format if only a single channel is available.

Transmission over Long Distances (<4000 Feet)
When relatively long distances are involved in reaching a peripheral device, driver circuits must be inserted after the bus
interface unit to compensate for the electrical effects of long cables.

Figure 1.27: Transmission Over Long Distances (<4000 feet)

This is the only change needed if a single peripheral is used. However, if many peripherals are connected, or if other
computer stations are to be linked, a Local Area Network (LAN) is required, and it becomes necessary to drastically
change both the electrical drivers and the protocol to send messages through the cable. Because multiconductor cable is
expensive, bitserial transmission is almost always used when the distance exceeds 20 feet.

In either a simple extension cable or a LAN, a balanced electrical system is used for transmitting digital data through the
channel. This type of system involves at least two wires per channel, neither of which is a ground. Note that a common
ground return cannot be shared by multiple channels in the same cable as would be possible in an unbalanced system.

Figure 1.28: Summing Amplifier

The basic idea behind a balanced circuit is that a digital signal is sent on two wires simultaneously, one wire expressing
a positive voltage image of the signal and the other a negative voltage image. When both wires reach the destination, the
signals are subtracted by a summing amplifier, producing a signal swing of twice the value found on either incoming line.
If the cable is exposed to radiate electrical noise, a small voltage of the same polarity is added to both wires in the
cable. When the signals are subtracted by the summing amplifier, the noise cancels and the signal emerges from the
cable without noise.

A great deal of technology has been developed for LAN systems to minimize the amount of cable required and maximize
the throughput. The costs of a LAN have been concentrated in the electrical interface card that would be installed in PCs
or peripherals to drive the cable, and in the communications software, not in the cable itself (whose cost has been
minimized). Thus, the cost and complexity of a LAN are not particularly affected by the distance between stations.

Transmission over Very Long Distances (>4000 Feet)
Data communications through the telephone network can reach any point in the world. The volume of overseas fax
transmissions is increasing constantly, and computer networks that link thousands of businesses, governments, and
universities are pervasive. Transmissions over such distances are not generally accomplished with a directwire digital
link, but rather with digitallymodulated analog carrier signals. This technique makes it possible to use existing analog
telephone voice channels for digital data, although at considerably reduced data rates compared to a direct digital link.

Transmission of data from your personal computer (PC) to a timesharing service over phone lines requires that data
signals be converted to audible tones by a modem. An audio sine wave carrier is used, and, depending on the baud rate
and protocol, will encode data by varying the frequency, phase, or amplitude of the carrier. The receiver's modem
accepts the modulated sine wave and extracts the digital data from it. Several modulation techniques typically used in
encoding digital data for analog transmission are shown below.

Figure 1.29: Encoding Digital Data for Analog Transmission

Similar techniques may be used in digital storage devices such as hard disk drives to encode data for storage using an
analog medium.
NETWORK FACILITIES
This section describes some common network facilities and connections. They are as:

1 Point to Point Circuits
Point to point topologies are simplest and most straight forward.

Point to point circuits are sometimes called twopoint circuits, and are used where the remote terminal has sufficient data
to warrant using a dedicated single connection to the host computer. The star network is as being point to point.

Figure 1.30: Point to Point Circuit

2 Multidrop/Multipoint Circuits
Multiple devices are attached to a common connection. This allows a number of devices to share an expensive line.
Only one device can use the line at any one time, and the line is often managed using polling. Frame relay is the most
common technology to implement this scheme, and it is typically a WAN technology. All the remote points are
connected to a single frame relay switch or router port, and communication between sites is managed by that central
point.

Figure 1.31: Multidrop Circuits

3 Switched Circuits
Switched circuits are offered by all public communications carriers. Dialup telephone circuits are an example of switched
circuits. Connections are made via a central switching center, which connects the two circuits together (like terminal and
host computer).

Figure 1.32: Switched Circuits

4 Packet Switched Circuits
Packet switching is a store and forward data transmission technique in which a message is broken up into small parts,
each called a packet.

Each packet is transmitted through the network (and may take a different route from the previously sent packet),
independent of other transmitted packets. They are reassembled at the destination end into the original message. The
facility which converts a message into packets, and reassembles packets into messages is called a Packet
Assembly/Disassembly (PAD) facility.
Charges are related to either time, the number of packets transmitted, or a combination of both. Distance charges are
relatively nonexistent.

Packet switching is very reliable, and public communications carriers offer packet switching as an alternative to other
forms of connections (like leased lines for example). There is a high degree of error detection and control built into the
packet switching protocol used to transfer packets from point to point in a network.

5 Integrated Service Data Network (ISDN)
ISDN refers to a new form of network which integrates voice, data and image in a digitized form. It offers the customer a
single interface to support integrated voice and data traffic. Rather than using separate lines for voice (PABX system)
and data (leased lines), ISDN uses a single digital line to accommodate these requirements.

The basic circuit provided in ISDN is the 2B+D circuit. This is two B channels (each of 64 Kbits) and one D channel (of
16 Kbits). The D channel is normally used for supervisory and signaling purposes, whilst the B circuits are used for voice
and data.

6 Local Area Network
A LAN is a group of microcomputers or other workstation devices located in the same general area and connected by a
common cable. The characteristics of LAN's are:

Physically limited<2 Km

High bandwidth>1 MHz

Inexpensive cable media

Data and hardware sharing amongst users owned by the user.

A LAN is used to share resources amongst a group of individuals or company employees. These resources are typically
files, application software, printers, computing power, modems, and fax machines

7 Wide Area Network
A wide area network is similar to a LAN, but geographically spread over a wider area. The different segments of the WAN
are interconnected by communication links (high speed serial, X.25 etc). In fact, a WAN often comprises a number of
interconnected LAN's at various sites.

The characteristics of a WAN are:

Geographically expansive.

Large number of computers and multiple host machines.

Sophisticated support devices like routers and gateways for interconnecting the various segments.
TYPES OF NETWORK ACCESS
A LAN connects resources together so they can be shared by users. In today's terms, this often means linking PC's
together. As discussed earlier, the PC's can be linked together using a bus, ring or star topology.

Each PC is equipped with a network interface card, which fits into an available expansion slot. Appropriate driver
software provides an interface between the PC operating system and the network interface card. Once the network
software is loaded, access to provide to other machines on the network. There are three main types of network access
provided.

1 Peer to Peer Services
Each PC in the network can directly communicate and share resources with any other machine in the network. Examples
of this are:

Lantastic

LAN Manager

Windows for workgroups

Windows NT

OS/2

UNIX

Software programs are executed either locally, or on the peer computer. A user can take advantage of a faster computer
by specifying that the program run on the faster machine, rather than their own.

2 File Server Services
A number of machines are designated as HOST servers. Users login to the host machines and access files and
application software stored on them. Users have no access to other non host machines. Examples of this are NOVELL
NETWARE (Also LAN Manager, Windows NT, and LANtastic). This software is executed on the user's machine. The file
server only provides file access; the file is downloaded from the file server into the workstation, where it is then
executed.

3 Client/Server Services
This is used extensively in database applications, where a front end software program sends commands to a server for
execution, with the results being displayed on the user's machine. It utilizes the high speed of the server machine to do
number crunching and complex operations.

The major problems associated with using networks are:

User training
Support (user and system)

Configuration

Table 1.1: Comparison of LAN and WAN Characteristics
Open table as spreadsheet
Characteristic LAN WAN
geographical size 0 – 2 Km 1 – 100 Km
number of nodes 1 – 200 1 – 500
data rate 1 – 100 Mbps 1 – 100 Mbps
error rate <10 – 9 <10 – 6
delays 1 – 100 ms 100 ms – 100 s
routing simple sophisticated
linkage bridges/repeaters gateways/routers
Figure 1.33: Relationship between Network Architectures
SUBNETWORK (SUBNET)
A subnetwork is a portion of a network, or a network that is part of a larger Internet work. This term is also a rather
subjective one; sub networks can in fact be rather large when they are part of a network that is very large. The
abbreviated term "subnet" can refer generically to a subnetwork.
THE INTERNET, INTRANETS AND EXTRANETS
In fact, the Internet is not just a large internet work, but substantially more. The Internet is defined not just as the
computers that are connected to each other around the world, but as the set of services and features that it offers. More
than that, the Internet defines a specific way of doing things, of sharing information and resources between people and
companies. And though it might be a bit melodramatic to say so, to many people the Internet is a way of life. As Internet
use and popularity exploded in the 1990s, many people realized that the techniques and technologies used on the
Internet would be useful if applied to internal company networks as well. The term Intranet was coined to refer to an
internal network that functioned like a "private Internet". It comes from the prefix "intra", which means "within". Of
course, "inter" is the opposite of "intra", so this makes some people think that an "intranet" is the opposite of an
"Internet". In fact, most intranets are Internet works as well.

As if that weren't bad enough from a jargon standpoint, the buzzword then decided to take matters a step further. If an
intranet is "extended" to allow access to it not only strictly from within the organization, but also by people or groups
outside the main company, this is sometimes called an extranet. "Extra" of course, is a prefix that means "outside" or
"beyond".
So, an extranet is a type of internal, private Internet that isn't entirely internal. An extranet is an extended intranet, which
is really a type of Internet that works like the Internet. An extranet isn't public and open to all—it is controlled by a private
organization. At the same time, it isn't entirely private either.

As you can see, the lines between the Internet, intranets and extranets were pretty blurry from the start, and the
concepts are rapidly blending into a diffuse mass of gray, as the whole computing world becomes more tightly integrated.
For example, even if you have an entirely private intranet, you will want to connect it to the Internet to communicate with
the "outside world" and to allow access to Internet resources. And an extranet may be implemented, in part, through the
public Internet infrastructure, using technologies such as Virtual Private Networking (VPN).

The key that binds all of these concepts together is that they all use "Internet technologies", a term that is it somewhat
vague. This usually refers to the use of the TCP/IP protocol suite, which is the defining technology of the Internet, as
well as the set of services that are available on the Internet.
The bottom line is that being told that a company has an "intranet" or an "extranet"— as opposed to a plain old boring
"network"— doesn't tell you much at all. It is best not to rely on the slogans and instead look at the underlying
characteristics of the network or Internet work itself. Furthermore, when designing such a network, focus on using the
technologies and protocols that make sense.
REVIEW QUESTIONS
1. What do you understand by computer network?
2. Write short note on evolution of computer.
3. What are the different functions of data network?
4. What are the different network components of communication?
5. What are the main uses of computer network?
6. Write different modes of communication.
7. What are the differences of asynchronous and synchronous transmissions?
8. What are differences between Internet, intranets and extranets?
9. Define the difference between a LAN and MAN.
10. Specify the amount of territory covered by each of the following network types: LAN, MAN, CAN, WAN, GAN.
11. What are the three main types of network topologies?
12. Encode the following unipolar data stream into a differential Manchesterencoded data stream.
13. What is the purpose of a Network Interface Card (NIC)?
14. What are the NIC's specification tasks?
15. Which IEEE standard addresses metropolitan area networks?
16. What are slots used for?
17. What is an intranet? How does it differ from the Internet?
18. What functions does an extranet provide?
19. The original three network types were LAN, MAN, and WAN. Describe how they differ from one another.
20. What is a topology?
21. List the basic network topologies.
22. Define bus topology.
23. Draw a hybrid topology with a star backbone and three ring networks.
24. Draw a hybrid topology with a ring backbone and two bus networks.
25. Draw a hybrid topology with a bus backbone connecting two ring backbones. Each ring backbone connects three
star networks.
26. Find three standards defined by ISO.
27. Give two instances of how networks are a part of our life today.
28. Assume six devices are arranged in a mesh topology. How many cables are needed? How many ports are needed
for each device?
Chapter 2: Network Models

OVERVIEW
Before discussing network models, we will see network architecture which is a formal, logical structure that defines how
network devices and software interact and function. It defines communication protocols message formats, and standards
required for interoperability. New hardware or software products created within a specific architecture are generally
compatible with other products create within the same architecture. Network architectures are designed by standards
organizations and manufacturers.
For example, IBM designed the Systems Network Architecture (SNA), Digital Equipment Corporation designed the Digital
Network Architecture (DNA), and the International Organization for Standardization (ISO) designed the Open System
Interconnect (OSI) architecture. There are several network models such as:

The internet layered protocol

The TCP/IP 4 layered protocol
The Microsoft networking protocol

But ISO Network model is unique which is described below.
INTRODUCING THE ISO OSI REFERENCE MODEL
Nearly all digital network descriptions start with the OSI (Open Systems Interconnect) model (Figure 2.1). It explains the
various "layers" of network technology. To the casual user, this model is a little abstract, but there is not a better way to
begin to understand what is going on.

Figure 2.1: The Seven Layer OSI Network Model

Sometimes it helps to understand each layer by examining the technology it represents. The application layer is the
most intuitive because it is what the user sees. It represents the problem that the user wants the system to solve.
Internet browsers and email programs are good examples. They allow the user to input and read data while connected
between a client PC and a server somewhere on the Internet. In an industrial application, a program on a Programmable
Logic Controller (PLC) might control a smart valve.
The presentation layer performs formatting on the data going to and from the application. This layer performs such
services such as encryption, compression, and conversion of data from one form to another. For example, an application
(one layer up) might send a time stamp formatted in 12hour time: 01 : 30 : 48 p.m. A more universal representation is 13
: 30 : 48, in 24hour time, which can be accepted or presented at the next node's application in the form that it needs.
One advantage to presentation services is that they help eliminate overhead, or embedded services, in application
programs.

The session layer establishes the connection between applications. It also enforces dialogue rules, which specify the
order and speed of data transfer between a sender and a receiver. For example, the session layer would control the flow
of data between an application and a printer with a fixed buffer, to avoid buffer overflows. In the time stamp example,
once the data is presented in 24hour time format, an identifier and length indicator are appended to the data string.

The transport layer is essentially an interface between the processor and the outside world. It generates addresses for
session entities and ensures all blocks or packets of data have been sent or received. In the time stamp example, an
address for each session entity (sender and receiver) and checksum are appended to the block generated by the session
layer.

The network layer performs accounting, addressing, and routing functions on messages received from the transport
layer. If the message is lengthy, this layer will break it up and sequence it over the network. This layer also uses a
network routing table to find the next node on the way to the destination address. In the time stamp example, a node
address and sequence number are appended to the message received from the session layer.

The data link layer establishes and controls the physical path of communication from one node to the next, with error
detection. This layer performs Media Access Control (MAC) to decide which node can use the media and when. The
rules used to perform these functions also are known as protocols. Ethernet and token ring contention are examples of
protocols. In the time stamp example, a header and trailer are appended to the message received from the transport
layer for flagging the beginning and end of the frame, type of frame (control or data), checksum, and other functions.

The physical layer is perhaps the most conspicuous layer from a cost point of view. It is relatively easy to understand
the labor and material costs of pulling cables, along with a physical infrastructure (conduits, raceways, and duct banks)
for maintaining cable integrity. This layer does not add anything to the message frame. It simply converts the digital
message received from the data link layer into a string of ones and zeroes represented by a signal on the media. One
example is RS485, where a binary 1 is represented by a mark, or off state, and a binary 0 is represented by a space, or
on state. A mark is a negative voltage between terminals on the generator, while a space is a positive voltage on those
terminals.
OPEN SYSTEMS INTERCONNECTION (OSI) ARCHITECTURE
A number of bodies have attempted to define standards to cover all aspects of computer communications, including
LANs. Many standards have been established by default. Though they were developed as products, Ethernet and token
ring networks have been adopted as international standards by such bodies as the Institute of Electrical and Electronic
Engineers (IEEE) and the International Standards Organization (ISO).

Figure 2.2: OSI Seven Layer

OSI refers to communications between computer systems which are enabled to exchange information freely irrespective
of type and manufacturer by virtue of their mutual adherence to a set of standards.

1 The Physical Layer (Layer 1)
The physical layer is the first or the lowest layer in the OSI model. This layer is concerned with transmitting raw bits of
data over a communication channel. The design issues have to do with making sure that when one side sends a 1 bit, it
is received by the other side as a 1, not a 0 bit. Other points which concern this layer include:

How many microseconds a bit occupies?
Whether transmission may proceed simultaneously in both directions?

How the initial connection is established?

How it is torn down when both sides are finished?

How many pins the network connector has and what each pin is used for?

2 The Data Link Layer (Layer 2)
The data link layer provides the nodetonode packet delivery service. Its function includes transmission/reception of
structured streams of bits over the network media. It provides the facility to subdivide the physical layer's raw bit stream
into structured blocks of information commonly called frames or packets.

Data link protocols usually include some means of error detection, typically based upon a simple checksum included at
the end of the frame.

A noise burst on the line can destroy a frame completely. In this case the data link layer's software on the source
machine must retransmit the frame. However, multiple transmissions of the same frame introduces the possibility of
duplicate frames. A duplicate frame could be sent, for example, if the acknowledgement frame from the receiver back to
the sender was destroyed.
It is up to this layer to solve the problems caused by damaged, lost, and duplicate frames so that layer 3 can assume it
is working with an errorfree (virtual) line. Another problem that arises at the data link layer and at most of the higher
layers is how to keep a fast transmitter from drowning a slow receiver in data (flow control).

Some mechanisms must be employed to let the transmitter know how much buffer space the receiver has at the
moment. Typically, this mechanism and error handling are integrated together.

3 The Network Layer (Layer 3)
This unit determines how packets, the units of information exchange in layer 3, are routed within the network. It should
also insure that all packets are correctly received at their destinations and in the proper order. This layer basically
receives messages from the source host, converts them into packets, and sees to it that the packets get directed
towards the destination. A key design issue is how the route is determined. This layer also looks after congestion
control.

4 The Transport Layer (Layer 4)
This layer provides for the transparent transfer of data, e.g., files between systems which might organize their data
somewhat differently. It relieves the transport user from any concern about how in detail data transfers may be affected
and optimises the available communications resources. This layer can be viewed as a bridge between the
communication oriented lower three layers and the application oriented upper three layers.

5 The Session Layer (Layer 5)
This layer supports the establishment, control and termination of dialogues between application processes. It facilitates
full duplex operation and maintains continuity of session connections. It also supports synchronization between user's
equipment and generally manages the data exchanges for the applications task.

6 The Presentation Layer (Layer 6)
The presentation layer is concerned with the syntax and semantics of the information transmitted. A typical example of a
presentation service is encoding data in a standard agreedupon way. Most user programs do not exchange random
binary bit strings. They exchange things such as people's names, dates, amount of money, and invoices. These items
are represented as character strings, integers, floating point numbers, and data structures composed of several simpler
items.

Different computers have different codes for representing character strings (e.g., ASCII and ABCDIC), integers (e.g.,
one's complement and two's complement), and so on. In order to make it possible for computers with different
representations to communicate, the data structures to be exchanged can be defined in an abstract way, along with a
standard encoding to be used ‘on the wire’.

The job of managing these abstract data structures and converting from the representation used inside the computer to
the network standard representation is handled by the presentation layer. The presentation layer is also concerned with
other aspects of information representation. For example, data compression can be used here to reduce the number of
bits that have to be transmitted and cryptography is frequently required for privacy and authentication.

Services provided by the presentation layer include:

a. Data Representation

Different computers have different internal representations for data. All the large IBM mainframes use
EBCDIC as the character code, whereas practically all other computers use ASCII. Some computers also
represent integers in one's complement and other in two's complement. To solve this problem, a conversion
will have to be made and this is carried out by the presentation layer.

b. Data Compression
The organizations that operate computer networks frequently expect to be paid for their efforts. In nearly all
cases, the cost of using a network depends on the amount of data sent. It is clear that the more bytes
sent, the more it costs, so the final bill can be reduced by compressing the data before sending them. Data
compression has been studied in many context for years. It is widely used to save space in memory, on
disk, and on magnetic tape.

c. Network Security and Privacy

Back in the early years when corporations and universities had a single computer centre, achieving
security was easy. All the organization had to do was station a guard at the door to the computer room.
The guard made sure that no one removed any tapes, disks, or card from the room unless explicitly
authorized to do so.
With the advent of networking the situation has changed radically. No one can manually police the millions of bits of data
that daily move between the computers in a network. Moreover, organizations have no way of being sure that their data
are not secretly copied by wiretap or other means on the way to their proper destination. Clearly, some kind of encryption
is needed to make the data unintelligible to all but their intended recipient. Protecting data from prying eyes is not the
only security issue in networking. One can imagine at least four security services:

Protecting data from being read by unauthorized persons preventing unauthorized persons from inserting or
deleting messages, verifying the sender of each message

Making it possible for users to send signed documents electronically. Encryption can be used to achieve
all these goals.

7 The Application Layer (Layer 7)
The application layer holds the user (also known as application) programs that do the actual work for which the
computers were purchased. These programs make use of the services offered by the presentation layer for their
communication needs. However, certain applications, such as file transfer, are so common that standards have been
developed for them to eliminate the need for each company to develop its own, and to make sure that everyone uses the
same protocols.

The transfer and remote file access are two of the most common applications in any computer network. People who are
working together on a project commonly need to share files. One approach is to have a machine where the original of
each file is held, and have copies transferred to other machines as needed. Another approach is to have each file ‘live’
on the machine where it was created (or where it is maintained), and have users on other machines ask for copies when
they need them.

Another situation in which file transfer is used is at a university that has many diskless workstations spread around
campus along with one or more machines with large disks. Students can log into any workstation and access their files
over the network. In an alternative design, the workstations can be equipped with small disks, allowing students to
download their files to their workstations at the start of each terminal session and upload them back to the main machine
when they are done. This arrangement means that a student can log into any workstation on campus (or perhaps even at
the dorm), not just the one specific workstation where his/her files are. Remote file access (as in the case diskless
workstations) is similar to file transfer, except that only pieces of files are read or written, rather than entire files. The
techniques used for file transfer and remote file access are similar.

In contrast to the session and presentation layers, on which there has been very little work done outside of OSI, there
has been a great deal of research about file transfer done at universities and in industry. Many file servers have been
built and experimented with.

The key idea behind most modern file servers is that of a virtual file store, an abstract file server, either freestanding or
running as a process on a timeshared computer. The virtual file store presents a standardized interface to its clients, and
provides a set of standardized operations that the clients can execute. Transfers to and from the virtual file store use
standardized protocols. If the real file server has a different internal structure than that of the virtual file store, it will need
some application layer software to hide the truth from the clients and make only the virtual file store interface visible. By
standardizing on a particular virtual file store interface, as OSI has done, it is possible for application programs to access
and transfer remote files without knowing all the details of numerous incompatible file servers.

Physical Layer Options
There are a number of implementations of the physical layer. Network devices allow a wide range of connectivity options.
Some networks are well defined using the OSI model, where cables, bridges, routers, servers, modems, and PCs are
easily identified. Sometimes only a few devices are linked together in some kind of proprietary network, or where the
network services are bundled in a blackbox fashion with the device.
The most common serial data exchange interfaces are RS232, RS422, and RS485 for connecting two or more devices
together. All three interfaces use Data Terminal Equipment (DTE) and Data Communication Equipment (DCE)
terminology (Figure 2.3). The DTE is the component that wants to communicate with another component somewhere
else, such as a PC communicating with another PC. The DCE is the component actually doing the communicating, or,
performing the functions of the generator and receiver discussed in the standards. A modem is a common example of a
DCE.
Figure 2.3: Serial Data Transmission

The interfaces between DTE and DCE can be categorized by mechanical, electrical, functional, and procedural aspects.
Mechanical specifications define types of connectors and numbers of pins. Electrical specifications define line voltages
and waveforms, as well as failure modes and effects. Functional specifications include timing, data, control and signal
grounds, and which pin(s) the functions are to use. The procedural interface specifies how signals are exchanged.
RS485 is another serial data transmission method. Officially, it is EIA 485, or "standard for electrical characteristics of
generators and receivers for use in balanced digital multipoint systems" by the Electronics Industry Association (EIA).
This standard defines a method for generating ones and zeroes as voltage pulses. Remember, for all the data handling,
framing, packeting, routing and addressing performed by the upper layers, it still comes down to pushing ones and
zeroes over some physical media.

What is important to know about RS485 is that it allows multiple receivers and generators, and it specifies cable
characteristics in terms of signaling speeds and lengths. A typical cable is a shielded twisted copper pair, which is
adequate for the typical signaling rate of 10 million bits per second (Mbps). This standard only defines the electrical
characteristics of the waveforms. Note that RS485 does not specify any media control functionsthat is strictly up to the
device connected to the generator (usually a chip). RS485 is generally good for cable lengths up to 2,000 feet.

One example of a simple serial network might be a series of recorders connected over an RS485 link to a PC that
receives data acquired by each recorder. The manufacturer sells a plugin card that installs in each recorder, with wiring
instructions. Each network card is daisychained to the others over a series of shielded twisted pair cables that
ultimately terminate on a network interface card in the PC. There is no real need to know and understand the network
layers in this arrangement, except to understand the limitations on RS485 (distance, shielding, data rate, etc.).
By title, the RS422 standard is TIA/EIA 422 B, "electrical characteristics of balanced voltage digital interface circuits"
by the telecommunications industry association (in association with the EIA). It is similar to RS485; the main
differences being the rise times and voltage characteristics of the waveform. RS422 generally allows cable lengths up to
1.2 kilometers at up to 100 thousand bits per second (kbps). At 10 million bps (Mbps), cable lengths are limited to around
10 meters (Figure 2.4). In the presence of cable imbalance or high common mode noise levels, cable lengths may be
further reduced in order to maintain a desired signaling rate.

Figure 2.4: Graph of Data Signaling Rate
INTERACTION BETWEEN OSI MODEL LAYERS
A given layer in the OSI model generally communicates with three other OSI layers: the layer directly above it, the layer
directly below it, and its peer layer in other networked computer systems. The data link layer in system A, for example,
communicates with the network layer of system A, the physical layer of system A, and the data link layer in system
B. Figure 2.5 illustrates this example.

Figure 2.5: OSI Model Layers Communicate with other Layers

OSI Layer Services
One OSI layer communicates with another layer to make use of the services provided by the second layer. The services
provided by adjacent layers help a given OSI layer communicate with its peer layer in other computer systems. Three
basic elements are involved in layer services: the service user, the service provider, and the Service Access Point
(SAP).

In this context, the service user is the OSI layer that requests services from an adjacent OSI layer. The service provider
is the OSI layer that provides services to service users. OSI layers can provide services to multiple service users. The
SAP is a conceptual location at which one OSI layer can request the services of another OSI layer.

Figure 2.6: Service Users, Providers, and SAPs Interact at the Network and Data Link Layers
CRITICISMS OF THE ISO OSI REFERENCE MODEL
By the mid 1990's it was clear that the protocols inspired by the International Standards Organisations Open Systems
Interconnection Reference Model (OSIRM), in particular those promulgated by the CCITT and similar bodies had failed to
gain any significant acceptance except at the lowest levels and there was widespread criticism of the OSIRM itself. The
failure of the protocols is related to a number of issues.

The protocols were slow to be finalized
The protocols were difficult to understand and implement

It was difficult to obtain copies of the standards documents

In all respects the contrast with the protocols associated with the Internet is clear and stark. The Internet protocols are
simple, described in plain English, freely available and had the advantage of being early in the field whereas the
OSIRM/CCITT protocols are elaborate and abstruse. The Internet protocols were developed in response to a direct and
immediate need. The Internet protocols were developed in the context of computer networking rather than the highly
regulated world of the national telecommunication bodies who were heavily involved in the OSIRM protocol
developments.

Even so it could be and has been suggested that the OSIRM based protocols could or would eventually replace the
original Internet protocols as the OSIRM based protocols became better understood and the consequences of the
informal design of the Internet protocols became clearer. It now seems most unlikely that this will happen. This is, of
course, partly due to the complete market dominance of the Internet protocols and related products but there are also
fundamental problems with the OSIRM.
The OSIRM prescribes how entities at the various levels should interact with each other and entities at other levels.
Level N entities must exchange data using services provided by level N1 entities. Another was of saying this is that
level N1 entities must be involved in every data transfer between level N entities. Level N1 entities provide their service
by exchanging data units which contain level N1 control information and data from level N entities. Level N control
information is passed to the remote side as level N1 data.

The reference model further specifies seven levels (or layers). This seven layer model has been reproduced in practically
every text on communications, unfortunately the chosen levels bear little relation to the practicalities of communications
protocol implementation and null layers and sublayers are common.

The Internet protocols fit in a much less rigidly defined framework with no formally defined layers but hierarchical use of
"lower" protocols by "higher" protocols being common. To those familiar with the formal framework of the OSIRM the
Internet protocols seem "ad hoc" and pragmatic, as indeed they are. It is a matter of observation that the OSIRM
community has not produced anything like the world wide web or the usenet. On a more technical front the less
prescriptive Internet protocols allow the use of a variety protocol methods that do not fit into the OSIRM scheme of
things.

The use of escape characters within a data stream to allow control information to be included within the data stream to
which it refers. This technique is adopted by the telnet protocol. The use of separate control and data connections at the
same level (i.e., using the same lower level service). This technique allows for different qualities of service for the control
and data connections. This method is used by the Internet FTP. Lower level control information can be used for higher
level functions. For example a low level link closure can imply closure at higher levels without need for explicit closure at
the higher levels.

An entity may use the services of several lower level (or even similar level) entities during a liaison. The use of DNS is a
good example of this kind of thing. A high level entity may bypass intermediate levels requesting direct explicit
encapsulation at the lower levels. This techniques is proposed for the real time extensions to IPng and may also be
useful to allow high level access to facilities such as multicasting and broadcasting.

The OSIRM can be further criticised for inadequate support for the concept of interconnected networks and the various
management functions such as name service. Interconnected networks imply that entities on two networks that are not
directly connected can still communicate even though the networks may be quite different. This has always been
fundamental for the Internet community. Such Internet protocols seem to fit between level 3 and level 4 of the OSIRM.
They rely on network services to transport data from one part of a network to another (i.e., from one gateway to another)
but they are used by endtoend protocols such as TCP which belong at level 4 of the OSIRM.

Management services simply do not seem to fit very well into the hierarchical model. The simplest examples of
management services are various name servers (such as DNS), connectivity tests (such as PING) and more general
network management protocols (such as SNMP). Such things can be seen either as services in their own right or as
facilities to be used by other services.

The Internet protocols place a much stronger emphasis on connectionless services. The basic Internet protocol is
connectionless.
EXAMPLE OF OSI COMMUNICATION
A typical OSI communications scenario can now be sketched. Assume that one end system (local computer) has a
client program which requires the services of a remote server program located at a remote computer connected by a
communications network. A simplified diagram is shown in figure 2.7. This shows the peertopeer communication.

Figure 2.7: Client/Server Interaction Across a Packet Data Network

The diagram provides only a very simplified view of what happens. In fact, a number of primitives are required to perform
even such simple communication. The text below performs a detailed analysis of the primitives and PDUs which are
exchanged.
The user executes the client program and attempts to send one piece of data to the server (for example,. a request to
lookup the address corresponding to a name). The actual data need not be of concern here. Before the client may send
the data, it must first establish a connection to the application layer (i.e., communications software library) on the local
computer.

The client program uses an AAssociate request function call to start the communication session. Once the application
layer has initialized, it contacts the presentation layer, sending a PConnect request primitive. This establishes the
format of the data to be used and the data formats which are to be supported by the system A on the network. This
information is sent to the session layer as an SDU with a SConnect request primitive. The session layer allocates a
session identifier, and selects appropriate protocol options to support the services requested by the application layer.
The session layer also identifies the intended recipient of the communication (i.e., the remote computer).

Figure 2.8: Establishment of a Connection between Client and Server (part 1)

The session layer proceeds to request a transport layer connection to the remote system using a TConnect request. At
this stage, the session layer may choose to not send the SConnect SDU, and instead to wait to see whether the
transport connection request succeeds. The transport request identifies the remote service required (i.e., the port
identifier for the server) and the type of transport protocol to be used. The basic types of transport service are reliable
and besteffort.

At this stage the transport layer requests the network layer to establish a connection to the remote system. The network
layer service will normally have established a link layer connection (and a physical layer connection) to the nearest
intermediate system. In this case however, we assume that all the layers must be connected.

Finally, the network layer connect packet is sent using the link layer service to the remote system. The system
responds by invoking the requested transport layer, and establishing a transport layer connection. If the remote system
is able, it will then confirm the establishment of the transport connection, and pass the confirmation back to the local
client computer. At this time the two systems have a communications path using the transport layer service.

The transport layer may then use this service to request a connection to the server process on the remote computer. It
does this by forwarding the original session layer SDU (which it had previously stored in the session layer of the local
computer) as a transport layer packet. This passes across the network layer service, and at the remote system is
passed to the session layer protocol.

The received SDU identifies the session ID and the application process with which the client wishes to communicate.
The presentation layer is sent a SConnect indication, containing the presentation options requested by the client and the
AAssociate request sent by the client. The application layer now attempts to connect to the server process. (In some
cases, a new server program will be activated to handle this new request).

Figure 2.9: Establishment of a Connection between Client and Server (part 2)
Once this has succeeded a response is generated, carrying the final details of the connection. This information is
passed as an AAssociate response primitive to the application layer. At each layer additional PCI is added by the
remote system, and finally an NData primitive is sent across the network layer service. At the receiver each layer
verifies the information it receives, and confirms the successful connection. The application layer send an AAssociate
confirm message to the client process which is then ready to transmit data.

Transmission of data is accomplished similarly. The client application sends a data request primitive, along with the
data, to the application layer, which adds its header and passes a data request primitive and data unit to the presentation
layer. This process continues, with addition of appropriate headers, until it reaches the data link layer, which adds its
header and passes the individual bits to the physical layer for transmission. The bits are received at the remote
physical layer, and passed to the remote system data link layer. When the data link layer recognizes the end of the data
unit, it strips off the header and sends the remaining PDU to the network layer (by including it in a data indication
primitive).

Data indication primitives, with reduced data units, cascade up the layers until the application data reaches server
application. Client application only learns of successful receipt if the server application returns an acknowledgement (in
this case, possibly with the requested data). The acknowledgement returns across the network. The "association" has
now been made and data may be sent across the network between the client and server.
THE TCP/IP REFERENCE MODEL
The TCP/IP reference model is the network model used in the current Internet architecture. It has its origins back in the
1960's with the grandfather of the Internet, the ARPANET. This was a research network sponsored by the department of
defense in the United States. The following were seen as major design goals:

Ability to connect multiple networks together seamlessly
Ability for connections to remain intact as long as the source and destination machines were functioning

To be built on flexible architecture

The reference model was named after two of its main protocols, TCP (Transmission Control Protocol) and IP (Internet
Protocol).

They choose to build a packetswitched network based on a connectionless Internet work layer.

Figure 2.10: TCP/IP Network Protocol

A detailed description of the reference model is beyond the scope of this document and project. The basic idea of the
networking system is to allow one application on a host computer to talk to another application on a different host
computer.

The application forms its request, then passes the packet down to the lower layers, which add their own control
information, either a header or a footer, onto the packet. Finally the packet reaches the physical layer and is transmitted
through the cable onto the destination host. The packet then travels up through the different layers, with each layer
reading, deciphering, and removing the header or footer that was attached by its counterpart on the originating computer.
Finally the packet arrives at the application it was destined for. Even though technically each layer communicates with
the layer above or below it, the process can be viewed as one layer talking to its partner on the host, as figure shows.

Communication Standards
De jure (meaning "from the low") standards are ratified by an international body such as International Standards
Organization (ISO), CCITT, IEEE. De facto (meaning "from the fact") standards become standards because every
decides they are a good idea.

Example: IBM's System Network Architecture (SNA), which defines how computers and devices communicate together.
DOS 3.1 and Novell's NetWare, all of these have become de facto standards because they hold important positions in
the market.
STANDARDS ORGANIZATIONS
A wide variety of organizations contribute to Internetworking standards by providing forums for discussion, turning
informal discussion into formal specifications, and proliferating specifications after they are standardized.

Most standards organizations create formal standards by using specific processes: organizing ideas, discussing the
approach, developing draft standards, voting on all or certain aspects of the standards, and then formally releasing the
completed standard to the public.
Some of the bestknown standards organizations that contribute to Internetworking standards include these:

1. International Organization for Standardization (ISO). ISO is an international standards organization
responsible for a wide range of standards, including many that are relevant to networking. Its bestknown
contribution is the development of the OSI reference model and the OSI protocol suite.

2. American National Standards Institute (ANSI). ANSI, which is also a member of the ISO, is the
coordinating body for voluntary standards groups within the United States. ANSI developed the Fiber
Distributed Data Interface (FDDI) and other communications standards. Electronic Industries Association
(EIA)— EIA specifies electrical transmission standards, including those used in networking. The EIA
developed the widely used EIA/TIA232 standard (formerly known as RS232). Institute of Electrical and
Electronic Engineers (IEEE)— IEEE is a professional organization that defines networking and other
standards. The IEEE developed the widely used LAN standards IEEE 802.3 and IEEE 802.5.

3. International Telecommunication Union Telecommunication Standardization Sector (ITUT). Formerly
called the Committee for International Telegraph and Telephone (CCITT), ITUT is now an international
organization that develops communication standards. The ITUT developed X.25 and other
communications standards.

4. Internet Activities Board (IAB). IAB is a group of internet work researchers who discuss issues pertinent
to the Internet and set Internet policies through decisions and task forces. The IAB designates some
Request for Comments (RFC) documents as Internet standards, including Transmission Control
Protocol/Internet Protocol (TCP/IP) and the Simple Network Management Protocol (SNMP).
IEEE HISTORY
The IEEE and its predecessors, the AIEE (American Institute of Electrical Engineers) and the IRE (Institute of Radio
Engineers), date to 1884.

From its earliest origins, the IEEE has advanced the theory and application of electro technology and allied sciences,
served as a catalyst for technological innovation and supported the needs of its members through a wide variety of
programs and services. The last quarter of the nineteenth century was marked by a tremendous growth in electrical
technology. By the early 1880s, telegraph wires crisscrossed the United States. Europe and America were connected by
underwater cable.
This growth in the technology and the planning for an international electrical exhibition to be held by the Franklin Institute
in Philadelphia prompted twentyfive of America's most prominent electrical engineers, including Thomas Edison, Elihu
Thomson and Edwin Houston, to issue a call for the formation of a society to promote their burgeoning discipline. On 13
May 1884, the AIEE was born in New York and quickly gained recognition as a representative for American electrical
engineers.

AIEE—Wire Communications, Light and Power
From the beginning, the major interests of the AIEE was wire communications and light and power systems.

An early and active participant in the development of electrical industry standards, the institute laid the foundations for all
work on electrical standards done in the United States. During its first three decades, the AIEE confronted and resolved
such internal concerns as locating permanent headquarters for the organization providing mechanisms for contact with a
farflung membership and with students, and fostering new technical interests through committees that were established
to meet the challenge of increasing specialization.

By 1912, however, the interests and needs of those specializing in the expanding field of radio could no longer be
satisfied by periodic technical committee meetings in their local areas.

The IRE—Wireless Communications
Two largely local organizations—the Society of wireless and telegraph engineers and the wireless institute—merged to
form an international society for scientists and engineers involved in the development of wireless communications—the
institute of radio engineers. Many of the original members of the IRE were members of the AIEE and both organizations
continued to have members in common until they merged to form the IEEE in 1963. The structural development and
general activities of the IRE were similar to those of the AIEE.

Specialized segments were gathered into professional groups under a central governing body. Geographical units and
student branches were formed. Meetings and publications facilitated the creation of an extensive literature and the
exchange of knowledge. Membership grades were established. Standards development became a major effort.

The nature of radio technology meant that the interests of the IRE went beyond national boundaries. Therefore, the new
organization sought and attracted members from many countries and eventually established units in several areas
throughout the world. From the beginning the ‘Proceedings of the IRE’ regularly published papers from authors outside
the United States. In the 1930's, electronics became part of the vocabulary of electrical engineering. Electronics
engineers tended to become members of the IRE, but the applications of electron tube technology became so extensive
that the technical boundaries differentiating the IRE and the AIEE became difficult to distinguish.
AIEE, IRE MERGE TO FORM IEEE
In 1961, the leadership of both the IRE and the AIEE resolved to seek an end to these difficulties through consolidation.
The next year a merger plan was formulated and approved and became effective on 1 January 1963.

IEEE Ethernet Standards
Here is the listing of some IEEE standards.

IEEE 802.1

802.1D (1993) MAC Layer Bridges(ISO 10038)
802.1p Quality of Service and Multicast Support

802.1Q VLAN processing

802.1G Remote Bridges
List of most useful Supplements to the IEEE 802 Standards:

802.3 (1985) Base Standard (10B5)

802.3a (1992) 10B2 Ethernet over thin coaxial cable
802.3b (1985) Broadband Ethernet (using coaxial TV cable, now seldom used)

802.3c (1985) Improved definition of a Repeater 802.3d (1987) Definition of Ethernet for Fiber (10BFOIRL)
(now seldom used)

802.3e (1987) 1Base5 or StarLAN (now seldom used)
802.3h (1991) Layer Management

802.3i (1990) 10BaseT, Ethernet over CAT5 Unshielded Twisted Pair(UTP)

802.3j (1993) defines Ethernet over Fiber (10BF)

802.3p/q (1993) Definition of Managed Objects

802.3u (1995) Definition of Fast Ethernet (100BTX, 100BT4, 100BFX)

802.3x (1998) Definition of Full Duplex Operation in a switched LAN

802.3y (1998) Definition of Fast Ethernet (100BT2 over low quality UTP)
802.3z Definition of Gigabit Ethernet (over Fiber)

802.3aa Definition of Gigabit Ethernet Maintenance

802.3ab Definition of Gigabit Ethernet (over UTP CAT5)

802.3ac Definition of Ethernet VLANs

802.3ad Definition of Ethernet VLAN Trunking
INTRODUCTION TO TCP/IP PROTOCOL ARCHITECTURE
TCP/IP protocols map to a fourlayer conceptual model known as the DARPA model, named after the U.S. government
agency that initially developed TCP/IP. The four layers of the DARPA model are: Application, transport, Internet, and
network interface. Each layer in the DARPA model corresponds to one or more layers of the sevenlayer Open Systems
Interconnection (OSI) model. Figure 2.11 shows the TCP/IP protocol architecture.

Figure 2.11: TCP/IP Protocol Architecture

While there is no universal agreement about how to describe TCP/IP with a layered model, it is generally viewed as being
composed of fewer layers than the seven used in the OSI model. Most descriptions of TCP/IP define three to five
functional levels in the protocol architecture. The fourlevel model illustrated in figure 2.12 is based on the three layers
(application, hosttohost, and network access) shown in the DOD protocol model, with the addition of a separate Internet
layer. This model provides a reasonable pictorial representation of the layers in the TCP/IP protocol hierarchy.

Figure 2.12: Layers in the TCP/IP Protocol Architecture

As in the OSI model, data is passed down the stack when it is being sent to the network, and up the stack when it is
being received from the network. The fourlayered structure of TCP/IP is seen in the way data is handled as it passes
down the protocol stack from the application layer to the underlying physical network. Each layer in the stack adds
control information to ensure proper delivery. This control information is called a header because it is placed in front of
the data to be transmitted. Each layer treats all of the information it receives from the layer above as data and places its
own header in front of that information. The addition of delivery information at every layer is called encapsulation.
(See figure 2.13 for an illustration of this.) When data is received, the opposite happens. Each layer strips off its header
before passing the data on to the layer above. As information flows back up the stack, information received from a lower
layer is interpreted as both a header and data.
Figure 2.13: Data Encapsulation

Each layer has its own independent data structures. Conceptually, a layer is unaware of the data structures used by the
layers above and below it. In reality, the data structures of a layer are designed to be compatible with the structures used
by the surrounding layers for the sake of more efficient data transmission. Still, each layer has its own data structure and
its own terminology to describe that structure.

Figure 2.14 shows the terms used by different layers of TCP/IP to refer to the data being transmitted. Applications using
TCP refer to data as a stream, while applications using the User Datagram Protocol (UDP) refer to data as a message.
TCP calls data a segment, and UDP calls its data structure a packet. The Internet layer views all data as blocks
called datagrams. TCP/IP uses many different types of underlying networks, each of which may have a different
terminology for the data it transmits. Most networks refer to transmitted data as packets or frames. In figure we show a
network that transmits pieces of data it calls frames.

Figure 2.14: Data Structures

Let's look more closely at the function of each layer, working our way up from the network access layer to the application
layer.
NETWORK PROTOCOLS AND STANDARDS
Network operating systems provide access to network resources for user processes, but much more is needed to
realize the goal of interoperability between processes running on separate systems. Interoperability means that
distinct processes, each running on its own processor, cooperate in accomplishing some goal. They cooperate by
exchanging information, by notifying each other as various tasks are completed, by requesting services from and
granting services to each other. The processes run in separate and perhaps very different operating environments;
they are related only in the cooperation. Their cooperative operation depends on a common understanding of how
their interaction will proceed. They must agree on the format and the meaning of messages exchanged, and under
what conditions the exchange will take place. The agreement about what messages will be sent and expected, under
what conditions, in what format, and with what meaning is called a protocol.

For example, suppose we want to have the ability to send a message from one station in a network to another, have
the receiving station confirms that the message arrived and then display the message on the screen. Sending the
message is not sufficient; even if we are sure the message will be received by the destination machine. What is
needed is an agreement, a protocol, that says.
I will send you a message with a header containing my identification and yours and the letters C and D. I understand
that when you receive the message, you will know that C means confirm and D means display. I expect that when
you receive the message, you will send me a message with a header containing your identification and mine and
message body containing the letters ok. You will also display my message on your screen.

To complete the protocol specification, we include the reciprocal agreement: Furthermore, I agree to respond to a
message from you, containing my identification and yours and the letters C and D, in the same way that you
respond to my message.

Simple protocol operating between two stations on a network and allowing each to display a message on the other's
screen lets call this protocol ConfirmAndDisplay. Now suppose two other stations in the network, or in a different
network, use a protocol called AcknowledgeAndShow.

Figure 2.15: Shows this Simple Protocol in Operation
AcknowledgeAndShow works exactly the same as ConfirmAndDisplay, except that in the message transmission,
the letter S is used instead of C and D. (The S indicates that the message is to be shown on the screen. An
acknowledgement of receipt of the message is always expected and not explicitly requested in the message.) If
stations named Mars and Venus use the protocol ConfirmAndDisplay only and stations Neptune and Uranus, in the
same network, use the protocol AcknowledgeAndShow only, then Mars and Neptune cannot display messages on
each other's screens.

Why would such a situation arise and what can be done about it? The situation may have arisen because the idea of
the mutual screen write by separate computers occurred to more than one person and each implemented it on the
computers he or she wanted it on. Neither was aware of or even interested in the other's existence. Only later did
someone recognize a reason to have the same facility on all the systems. There are four potential solutions to the
problem:

Drop one protocol entirely and use the other on all the systems. This is the obvious solution and seems reasonable
at first glance. Unfortunately, it turns out that each protocol has spread beyond its initial use and is deeply imbedded
in a number of applications. If the protocol is discontinued, all the applications that use it will have to be identified
and modified to use the replacement. Neither side is willing to be the one to go to that trouble. Further, each side is
convinced that its own approach is better and does not wish to give up the perceived advantages.

Drop both protocols and develop a new one that all stations use. The new protocol would be carefully designed after
thorough study and would address all the situations in which such a protocol might be useful. Since the new protocol
would be installed in a very large collection of systems, not just the original Mars, Venus, Neptune, and Uranus,
each of these systems would gain access to many other partners in using the protocol.

Each station runs both protocols, using the correct one for each communication. This is very practical for a simple
case of two competing, similar protocols. However, if there are many other versions of this protocol, the burden on
each system to keep all of them is substantial.

Keep the original protocol in each set of stations and also adapt the new global protocol to extend the range of
stations each can work with. Mars and Venus will still use ConfirmAndDisplay when they work together because
each knows the other will work correctly with that protocol and each has a number of application programs that
include ConfirmAndDisplay. To display a message on Neptune's screen, Mars will use the new global protocol.

This very small example illustrates the current condition in network protocols. There are many reasons why a
number of different protocols exist for accomplishing similar goals. Some of the differences result from requirements
that demand specific features that are not consistent with other requirements. Others grew out of historic events and
have become deeply ingrained in the products and practices of the industry. The various stations on a network come
from different manufacturers, each with a long history of its own way of working. Allowing processes to interoperate,
machines to communicate with each other and to cooperate to share a common transmission medium requires
another kind of cooperation—the development of standards. The standards define protocols independent of
individual manufacturers and applications.

Joining incompatible independent systems by agreeing to some common principles is not a new activity. In the 19th
century, a railroad built for one region had a different width of track from a railroad in another region. As long as the
regions remained separate, and the equipment was manufactured specifically for use in one region, the differences
were immaterial. However, when the use of the railroads expanded, and the ability to travel and ship goods over long
distances was recognized, the problem illustrated in figure 2.16 had to be solved. The solution was an agreement to
build railroad equipment according to a common or standard specification. Notice that agreeing on a standard gauge
track did not allow trains to travel between narrow and wide gauge tracks. The original tracks and trains could still be
used for transportation within the region; but when travel between regions occurred, only standard gauge track and
trains built to run on it would do.

Figure 2.16: A Protocol Data Unit

The incompatible personal computer systems initially built by IBM and Apple provide another example. A lot of
development effort is expended in preparing programs to run in these different environments.

The situation is similar with the developing network standards. Older proprietary protocols can still be used between
the systems that implement them. However, to interoperate with other systems, the global standards must be
observed. As a result, more than one set of protocols may exist in many systems. As more and more systems
adopt the global standards there will be less reason to use the older, more restricted, protocols. However, because
of the investment in the current systems and the number of applications that depend on them, these older protocols
will continue to be important for some time.
COMMUNICATIONS PROTOCOLS
A communications protocol defines the rules for sending blocks of data (each known as a Protocol Data Unit (PDU)) from
one node in a network to another node. Protocols are normally defined in a layered manner and provide all or part of the
services specified by a layer of the OSI reference model. A protocol specification defines the operation of the protocol
and may also suggest how the protocol should be implemented. It consists of three parts:

1. Definition of Protocol Control Information (PCI) format which forms the PDU header.
2. Definition of procedures for transmitting and receiving PDUs.

3. Definition of services provided by the Protocol Data Layers.
A PROTOCOL DATA UNIT
The PDUs exchanged have two parts: a header (also known as the protocol control information (PCI)) and a payload
(also known as a service data unit (SDU)). The protocol does not define or constrain the data carried in the payload part.
It does specify the format of the PCI, defining the fields which are present and the way in which the patterns of bits are
to be interpreted.

A protocol also defines the procedures which determine how the PDU will be processed at the transmit and receive
nodes. The procedures specify the valid values for the PCI fields, and the action be taken upon reception of each PCI
value (usually based on stored control information). Examples of procedures which are implemented in protocols include:
error recovery (e.g., the check point procedure, the gobackn procedure)

flow control

segmentation
service access point selection

connection management

The documents which define a protocol procedures are usually large and are seldom concrete (i.e., they may not be
directly translated to executable programs). They typically define the actions to be taken when a particular condition is
detected, but not how the condition is to be detected. It has been said that "Part of what makes a protocol mature is
good implementation guidelines and folklore". The success of the TCP/IP protocol suite is largely due to the "industrial
strength" code available in freely distributed reference implementations.

Figure 2.17: A Protocol Implemented by Several Processes (represented by circles) Communicating Using Queues
of PDUs

A protocol implemented by several processes (represented by circles) communicating using queues of PDUs, a shared
information area (shown as a rectangle) and function calls between the processes (represented by arrows).

Protocols are usually implemented by writing a number of programs (processes) which communicate with one another
through queues and by function calls. One or more timers are also usually required to ensure correct operation of the
protocol. To start and stop timers, a protocol normally uses an interface to the computer's operating system. This
interface is also used to request new (empty) buffers for received PDUs (or PDUs created by the layer) and to release
buffers which are no longer needed by the protocol.

Protocols are generally described using a layered architecture known as the OSI reference model. Which abstracts the
details of the protocol and allows a simple description of the service provided by the protocol to the protocol layer above
and the service required by protocol layer from the layer below?

Examples of protocols include:

Data link layer—HDLC, MAC, ARP
Network layer—IP, ICMP

Transport layer—UDP, TCP

You may be wondering how do the computers send and receive data across communication links. The answer is data
communication software. It is this software that enables us to communicate with other systems. The data
communication software instructs computer systems and devices as to how exactly data is to be transferred from one
place to another. The procedure of data transformation in the form of software is commonly called protocol.

The data transmission software or protocols perform the following functions for the efficient and error free transmission of
data.
Data sequencing: A long message to be transmitted is broken into smaller packets of fixed size for error
free data transmission.

Data routing: It is the process of finding the most efficient route between source and destination before
sending the data.

Flow control: All machines are not equally efficient in terms of speed. Hence the flow control regulates the
process of sending data between fast sender and slow receiver.

Error control: Error detecting and recovering is the one of the main function of communication software. It
ensures that data are transmitted without any error.
PROTOCOL HIERARCHIES
To reduce the design complexity of networks, they are organized as a series of layers or levels, each one built upon its
predecessor.

The purpose of each layer is to offer certain services to the higher layers. Layer n with one machine carries on a
conversation with layer n with another machine. The rules and conventions used in this conversation are collectively
known as the layer n protocol.

Network Protocol Levels
You should be aware of the fact, that when talking about networking you will hear the word "protocol" all the time. This is
because protocols are sets of standards that define all operations within a network. They define how various operations
are to be performed. They may even define how devices outside the network can interact with the network. Protocols
define everything from basic networking data structures, to higher level application programs. They define various
services and utility programs. Protocols operate at many layers of the network models described below. There are
protocols considered to be transport protocols such as TCP and UDP. Other protocols work at the network layer of the
OSI network model shown below, and some protocols work at several of the network layers.

RFCs
Protocols are outlined in Request for Comments (RFCs). At the end of this document is a list of protocols and
associated RFC numbers protocols. Although RFCs define protocols not all RFCs define protocols but may define other
requirements for the internet such as RFC 1543 which provides information about the preparation of RFCs. The following
RFCs are very central to the TCP/IP protocol.

RFC 1122—Defines host requirements of the TCP/IP suite of protocols covering the link, network (IP), and transport
(TCP, UDP) layers.
RFC 1123—The companion RFC to 1122 covering requirements for internet hosts at the application layer.

RFC 1812—Defines requirements for internet gateways which are IPv4 routers.
REVIEW QUESTIONS
1. Write complete note on ISO OSI reference model.
2. Write short description on all seven layers.
3. What are the differences of TCP/IP and OSI model?
4. What are the different standard organizations?
5. Draw TCP/IP protocol architecture.
Part 2: Physical Layer
CHAPTER LIST
Chapter 3: Data Transmission Signals
Chapter 4: Multiplexing and Switching

Chapter 5: Transmission Media

Chapter 6: Telecommunication
Chapter 7: Cable Television

Chapter 8: High Speed Digital Access—SONET
Chapter 3: Data Transmission Signals

INTRODUCTION TO TRANSMISSION
Today's world of telecommunications is quickly changing from telephone based services into a global network of all
digital services. Telephone service has been one of the primary methods of communications in the twentieth century.
Although the telephone network was designed to transmit voice signals, pressure from other technologies forced it to be
adapted to handle data transmission, facsimile and video.
When computers began sending data electronically the telephone network provided the logical framework for the
connections between systems. The telephone network was ideal for computer communications because telephone
service provided connections into most businesses and homes. The only problem with the telephone network was
converting the nonvoice signals so that they could be carried on voice circuits.

Since computers communicated in a digital format and the telephone network was analog, the transmitted signals had to
be converted by devices called modems. Technology has now shown it is easier to create a network for digital
transmission and convert the voice signals to move them. In actual usage voice, data and video can all be digitized and
moved through the same network.

The main challenge seen today is the majority of the existing public telephone network is set up to carry voice signals.
We are also seeing a growth in communications requirements due to new services like videoconferencing and local area
networks. These new services require transmission speeds ranging from 100 kilobits per second up to a megabit or
more.

Modern transmission technology came to the rescue by showing that it is easier and more cost effective to build a
network that transmits digital signals. All we have to do is convert everything to digital prior to putting it into the network
for transmission. In actual usage, the computers already use digital information, and only the voice and video need to be
digitized. For the past twenty years telephone companies around the world have been installing new transmission
equipment in their internal networks to carry all signals in digital format. Now, they are extending the digital signals all of
the way into businesses and homes by converting the telephone networks from "telephone service" to "digital service."
These new networks will provide you with end to end connections of binary or digital bit streams at various speeds. You
will then select and use the bit stream at the proper speed to encode your voice, video or data information on the circuit.
In essence the telephone companies will become "digital companies" providing a service called Integrated Services
Digital Network (ISDN). The purpose of this training course is to provide an introduction to digital transmission techniques
and systems. The course is divided into five parts that take you from the basic concepts of the digital network through to
the customer interface and use of the network.
SIGNAL
A signal is any physical phenomenon that can be modelled as a function from time or position to some scalar or vector
valued domain that can be real or complex valued.

Signal Characterization
Communications is the process of transmitting meaningful signals from one location to another. Signals have several
classifications and properties.

Signal Classifications

Deterministic signals
Random signals

Energy signals

Power signals
Signal Properties

Time averages

Auto correlation

Bandwidth
DATA TRANSMISSION SIGNALS
The information carrying signals are divided into two broad classes;

Analog

Digital

Analog Signals
Analog signals are continuous electrical signals that vary in time as shown in figure 3.1. Most of the time, the variations
follow that of the nonelectric (original) signal. Therefore, the two are analogous hence the name analog.

Figure 3.1: Analog Signal

Not all analog signals vary as smoothly as the waveform shown in figure 3.2. Analog signals represent some physical
quantity and they are a ‘model’ of the real quantity.

Figure 3.2: Signal Amplitude

Example:

Telephone voice signal is analog. The intensity of the voice causes electric current variations. At the receiving end, the
signal is reproduced in the same proportion. Hence the electric current is a ‘model’ but not one's voice since it is an
electrical representation or analog of one's voice.

Analog signals are what we encounter every day of our life. Speech is an analog signal, and varies in amplitude (volume)
and frequency (pitch). The main characteristics of analog signals are:

Amplitude
This is the strength of the signal. It can be expressed a number of different ways (as volts, decibels). The higher the
amplitude, the stronger (louder) the signal.

Frequency
This is the rate of change the signal undergoes every second, expressed in Hertz (Hz), or cycles per second. A 30Hz
signal changes thirty times a second. In speech, we refer to it as the number of vibrations per second, as the air that we
expel out of our mouth pushes the surrounding air at a specific rate.

Figure 3.3: Signal Frequency

One cycle is defined as a point on the wave to the next identical point. The numbers of cycles there are per second
determine the frequency of the signal.

Analog Transmission
Using analog transmission techniques, such as those found on the existing telephone networks has presented a problem
as newer services are brought into the network. This section will explore some of those problems and contrast them to
the advantages of providing digital services.

Analog Signal Transmission
Although the microprocessor and digital network technologies have fundamentally reinvented the ways in which today's
data acquisition systems handle data, much laboratory and manufacturing information is still communicated the "old"
way, via analog electrical signals. And a fundamental understanding of how analog signal transmission works must first
begin with a discussion of electrical basics.

To understand the ways in which an analog signal is transmitted over a circuit, it is first important to understand the
relationships that make analog signal transmission possible. It is the fundamental relationship between voltage, current,
and electrical resistance (Figure 3.4) that allow either a continuously varying current or voltage to represent a continuous
process variable.

Figure 3.4: A Basic Electric Circuit

While charge flow is electric current, voltage is the work done in moving a unit of charge (1 coulomb) from one point to
another. The unit of voltage is often called the potential difference, or the volt (V). The International System of Units (SI)
unit for electrical flow is the ampere (A), defined as one coulomb per second (c/s).

A signal source of voltage, V, will cause a current, I, to flow through a resistor of resistance, R. Ohm's law, which was
formulated by the German physicist George Simon Ohm (1787–1854), defines the relation:

While most singlechannel analog signal transmissions use direct current (DC) variations in current or voltage to
represent a data value, frequency variations of an alternating current (AC) also can be used to communicate information.
In the early 19th century, Jean Baptiste Joseph Fourier, a French mathematician and physicist, discovered that AC
signals could be defined in terms of sine waves. A sine wave is described by three quantities: amplitude, period, and
frequency. The amplitude is the peak value of the wave in either the positive or negative direction, the period is the time
it takes to complete one cycle of the wave, and the frequency is the number of complete cycles per unit of time (the
reciprocal of the period).

To begin, we must understand that for most of its first one hundred years the worldwide telephone network was
concerned with only carrying voices. This meant that the network was built to connect telephones across the world and
carry a voice between two points. Although the network architecture worked well for voice transmission, the newer
services like computer data, facsimile and video transmission have been constrained by the analog voice network
design.

Looking at figure 3.5, we can see how the existing telephone network is used to provide a variety of office
communications. This drawing shows one key problem with the telephone network. Each of the various lines for
telephone or data communications requires a separate line or circuit. The telephone systems have tielines and trunks to
the outside world, computers use high speed analog or digital circuits and video systems use high speed or broadband
circuits. Each type of circuit has its own problems related to installation, support and maintenance. In many cases the
telephone companies had different service organizations to handle each service creating confusion for the customers.

Figure 3.5: Analog Telephone Services

Telephone companies also found problems with voice quality as it was transmitted across long distances. Analog
transmission and amplification had limits, since noise was usually amplified along with the voice signal. This meant there
was a limit to how many times an analog signal could be amplified and still be recognized.

Analog transmission systems were also difficult to test and service. Usually, the telephone company technicians had to
take an analog transmission system out of service before doing any testing or maintenance.

In addition to the analog transmission problems, telephone companies were faced with increased pressure from
customers for more services and greater bandwidths. Growth in telephone usage in the 1950's and 1960's also added to
the telephone companies requirements to provide more voice circuits globally.

Analog I/O Functionality
Today, digital computers and other microprocessorbased devices have replaced analog recording and display
technologies in all but the simplest data acquisition applications. And while computers have had an undeniably positive
impact on the practice of data acquisition, they speak only a binary language of ones and zeroes. Manufacturing
processes and natural phenomena, however, are still by their very nature analog. That is, natural processes tend to vary
smoothly over time, not discontinuously changing states from black to white, from on to off.

To be meaningfully recorded or manipulated by a computer then, analog measurements such as pressure, temperature,
flow rate, and position must be translated into digital representations. Inherently digital events, too, such as the tripping
of a motor or a pulse generated by a positive displacement flow meter, must be made interpretable as a Transistorto
Transistor Logic (TTL) level changes in voltage. Hence, the origination and ongoing development of input/output (I/O)
systems (Figure 3.6) for converting analog and digital information about realworld processes and events into the
language of computers.

Figure 3.6: Functional Diagram for Data Acquisition and Control
Transmission of video using analog techniques is very popular, especially for shorter distances, where costs can be
minimized and complex multiplexing and timing equipment is unnecessary. The transmission of analog voice signals
may also be attractive in small shorthaul systems.

In addition, fiber optic sensor systems may incorporate analog transmission. Requirements that analog transmission
places on applications include high signaltonoise ratio and high source linearity. While analog transmission can be
attractive for shorthaul or mediumhaul systems, it is unattractive for longhaul systems where digital techniques provide
better performance.

Digital Signals
Digital signals are noncontinuous, they change in individual steps. They consist of pulses or digits with discrete levels
or values. The value of each pulse is constant, but there is an abrupt change from one digit to the next. Digital signals
have two amplitude levels called nodes. The value of which are specified as one of two possibilities such as 1 or 0, high
or low, true or false and so on. In reality, the values are anywhere within specific ranges and we define values within a
given range.

Figure 3.7: Digital Signal

Digital signals are the language of modern day computers. Digital signals normally comprise only two states. These are
expressed as ON or OFF, 1 or 0 respectively. The following diagram shows a digital signal.

Figure 3.8: Digital Signal Showing Two States, 0 and 1

Digital Transmission
In the early 1960's the telephone companies began using a new form of transmission called "digital." This system
allowed them to accept standard analog voice signals and convert them to digital signals for transmission across their
long distance and toll networks. This service was called TCarrier in the North America and ECarrier in Europe (or
sometimes just "digital transmission systems").

Digital transmission offers an advantage with regard to the acceptable signaltonoise ratio (SNR) at the optical receiver.
Digital communication systems can tolerate large amounts of signal loss and dispersion without impairing the ability of
the receiver to distinguish a binary 1 from a binary 0. Digital signalling also reduces the effects that optical source
nonlinearities and temperature have on system performance. Source nonlinearities and temperature variations can
severally affect analog transmission. Digital transmission provides superior performance in most complex systems
(Such as LANs) and longhaul communication systems. In shorthaul systems, the cost and complexity of analogto
digital and digitaltoanalog conversion equipment, in some cases, outweigh the benefits of digital transmission.

Digital Signal Transmission
Industrial networks that transmit data using digital signals often are an integral part of a data acquisition or process
control solution. A basic understanding of the network technologies that are available for various applications is required
to make the best implementation decisions—decisions that can have a profound effect on the ability to adapt to ever
changing technologies.

For example, the type of network(s) or network products selected for a data acquisition application can greatly affect
cost/benefit evaluations for future projects. Until recently, fiber optics was an expensive option that seemed like overkill
for most applications. But with the information load now likely to flow between nodes on an enterprise network, fiber
looks attractive.

Network technologies come in a bewildering array of options, and certain segments of this technology are changing at an
incredible rate. The average user might spend weeks researching the various ways to build or improve a network, just to
find that once a decision is made and products are purchased, the next wave of bigger, better, faster technology is now
available.
A welldesigned, integrated solution to data transmission will yield a competitive advantage to any industrial enterprise.
Users in all aspects of a business should be able to obtain plant and business data from any physical node, local or
remote. "gluing" pieces of new and/or existing networks together is becoming more feasible with the use of bridging,
routing, and media conversion technologies that link local area, wide area, and industrial networks together. And with
Internet and wireless technologies, data transmission over large geographic areas is increasingly feasible.

Digital I/O Functionality
In contrast to analog transducers that sense continuous variables such as pressure and temperature, many transducers
provide an output that is one of two states: high or low, open or closed. A pressure might be too high or a temperature
too low, triggering closure of a switch. Outputs, too, are not strictly analogsolenoid valves typically are opened or
closed, many pumps and heaters are simply turned on or off. Pulse signals are another form of digital I/O, with one
rotation of a turbine flow meter or tachometer corresponding to a single, countable event. Digital I/O also can be used for
parallel communications among plugin expansion cards, and to generate clock and other timing signals.

Already in the binary language of computers, these types of digital, or discrete, inputs and outputs (I/O) are much easier
for microprocessorbased data acquisition systems to deal with than analog signals. Similar to analogtodigital
converters used for analog I/O, digital I/O is designed to deal directly with TransistortoTransistor Logic (TTL) level
voltage changes. TTL typically sets the low voltage level between 0 and 0.8 V and the high voltage level between 2.0
and 5.0 V. Voltage levels between 0.8 and 2.0 V are not allowed. A voltage change, then, from the high range to the low
range (or vice versa) represents a digital change of state from high to low, on to off, etc.

And because acquiring an analog signal is more complex than acquiring a digital one, analog I/O channels also are more
expensive. Hence, if digital I/O is adequate, do not bother with analog.

This system was a great advantage for telephone companies because it reduced their transmission and maintenance
costs. The disadvantage was that it did not extend directly to the customer's office. Up until the 1980's the telephone
company still accepted and switched the majority of telephone circuits using analog techniques. The only benefit the
customer received from this arrangement was better quality sound on long distance telephone calls.
There was also an anomaly since great efforts were being made to build modems that translated computer digital signals
to analog signals for transmission across telephone circuits. So while companies were paying money for modems which
translated computer digital signals into analog transmission, the telephone company was transporting that analog signal
on a digital network. Taking the digital signal and converting it into analog and then converting that analog signal into a
digital transmission format were a very inefficient way to transmit information. In fact the modems had upper speed limits
of 19,200 bits per second, while the signal was actually being carried on a 64,000 bit per second digital carrier.

In the 1980's the telephone companies began extending the digital service to the customer allowing them to do their own
analog to digital conversion. By moving the digital conversion into the customer's office, the telephone company can
provide one type of link and the customer can differentiate the various services. This means that the customer can have
one link into the telephone company and the telephone company has one "easy to maintain" network.

As shown in figure 3.9, the use of digital services allows the one link to provide all of the voice and data connections. As
digital networks evolve, everything can be connected to a common communications link. First, the computer systems
are connected to the PBX or some other type of digital communications controller. This PBX or controller can then be
connected to one or more high speed digital circuits which go into the telephone company. Whether the incoming
transmission is voice, data, video or images it will be translated into common digital format and moved through a
common digital network worldwide.
Figure 3.9: Digital Services

It's easy to understand that if we convert everything to digital it makes communications easier because we only need
one type of circuit for all of our communications. You should understand though that through the use of all digital
communications both the telephone company and the end user are winners.

The payback for the telephone companies is a massive reduction in maintenance costs. Digital communications
networks are easier to test and maintain because there is only one type of signal. Also digital transmission systems can
detect failures and switch to backup systems automatically. This payback of lower maintenance costs can be a greater
incentive to the telephone companies than the customer requirements for digital services.

For end users or customers, the use of fully digital communications can reduce circuit costs by fifty percent or more.
Also the use of digital circuits improves quality and reliability of voice and data communications.

The world's telephone companies are responding to the economic, competitive and customer pressures to install
networks that are fully digital. Eventually this will mean that the ability to provide telephone connections, data
transmission, video links or other services will be done by the customer at their site through a common digital controller.
The organization that we know today as the telephone company or PTT will become only a "digital company" providing
one of the new services like ISDN (Integrated Services Digital Network) or ATM (Asynchronous Transfer Mode).

The Future of Digital Communications
Within today's office the use of communications has advanced far beyond the simple telephone on a desk. The
revolution begins with personal computers on desks tied together on a local area network as shown in figure 3.10. Next,
the modern PBX or telephone system is a sophisticated, computer controlled system that switches voice and data in a
digital format. Next, new uses for digital technologies like videoconferencing allow companies to hold meetings between
offices across the globe. And finally there are still central computer systems providing support for core applications such
as accounts payable, accounts receivable and payroll. Today, most of these systems can be connected over common
high speed digital channels to communicate with other systems worldwide on private or public networks.

Figure 3.10: Modern Office Communications

In reality though the use of digital communications is not that prolific and as a result we will still see two types of
switching systems and networks supported for many years as shown in figure 3.11. The older network will continue to
provide common telephone services to areas that still use the older equipment. The new all digital networks will support
the switching and transmission of digital signals along with modern digital telephones and other services. Both networks
will allow telephone calls and certain other existing services to cross between them and maintain compatibility.
Figure 3.11: The Future Networks

Although there will be bridging to support telephone calls and other common service between the two networks, some of
the newer services will only be available to all digital users.
Again looking figure 3.11, the new all digital networks will be simply providing digital bandwidth. Therefore in order to
compete the telephone companies will have to provide access to other services in order to maintain their market shares.
Rather than using separate lines to connect to these information services, you could simply connect through the digital
company and receive the charges as a part of your regular telephone billing. The types of services customers will see
through the future all digital networks include:

Access to existing and future data networks. This would include the existing packet switched networks (X.25, etc.) and
the future public data networks (frame relay, etc.). Global electronic mail services providing access to services like the
Internet and gateways to the mail services of other providers.

Databases and other Online Information Services
Access to worldwide directories providing information on how to navigate through the future digital world to find
businesses, people and information services. Remember that any of these services will be as easy as dialing a
telephone call. Most of the digital circuits in use today are available either as special services from the telephone
company or provided through private networks of microwave, fiber optic or satellite links. The concept of a single
common digital public network has been emerging over the past ten years under the "Integrated Services Digital
Network" or the acronym ISDN. ISDN is not a theory but, is actually available and in use in several industrialized
countries including Japan, Singapore, France, the United States and Great Britain to name a few.

Advantages Digital Transmission
Digital information is better transmitted in its digital form because converting the signal to analog and sending it through
an analog network can be costly. Digital data is easily compressed; therefore it can be transmitted using a small
bandwidth. Because of the nature of devices used to boost the signal strength during transmission, error performance is
much improved when compared with analog. It is also better to transmit information in digital form because computer
components used in the transmission process are very reliable.

Disadvantages Digital Transmission
Information is widely available as analog. Conversion of analog signal to digital bit stream requires special equipment.
This adds cost and additional equipment may also contribute additional points of failure. Another disadvantage is that
during the conversion process the quality of the signal is negatively affected.

Private Digital Networking Today
It is also possible that a company may want to build a private digital network and bypass the telephone company. Digital
services are making this easier than ever because you only need to meet one common interface regardless of the type
of information you transmit. In today's market, the end user has a variety of private digital transmission systems to
choose from as shown in figure 3.12.

Figure 3.12: Private Digital Link Options

Options for private digital networks today include:

Satellite earth stations that can provide both national and international links directly between customer
sites (bypassing the local telephone company connections).

Microwave radio systems that can provide links of up to fifty miles. In some cases, customers have even
used microwave systems to provide backup or emergency links into their local telephone company.

Fiber optic cables that can provide digital services linking local or remote buildings. Private networks can
also be built around dedicated services purchased from telephone companies.
ANALOG SIGNAL TYPES
Most data acquisition signals can be described as analog, digital or pulse. While analog signals typically vary smoothly
and continuously over time, digital signals are present at discrete points in time in figure 3.13. In most control
applications, analog signals range continuously over a specified current or voltage range, such as 4–20 mA dc or 0 to 5
V dc. While digital signals are essentially on/off (the pump is on or off, the bottle is there or isn't), analog signals
represent continuously variable entities such as temperatures, pressures, or flow rates. Because computerbased
controllers and systems understand only discrete on/off information, conversion of analog signals to digital
representations is necessary.

Figure 3.13: Digital and Analog Signal Representations

Transduction is the process of changing energy from one form into another. Hence, a transducer is a device that
converts physical energy into an electrical voltage or current signal for transmission. There are many different forms of
analog electrical transducers. Common transducers include load cells for measuring strain via resistance, and
thermocouples and resistance temperature detectors (RTDs) for measuring temperature via voltage and resistance
measurement, respectively. Transmission channels may be wires or coaxial cables.

For noiseresistant transmission over significant distances, the raw transducer signal is often converted to a 4–20 mA
signal by a twowire, looppowered transmitter. The bottom value of a process variable's range, for example, a
temperature, is typically designated as 4 mA, making it easy to distinguish transmitter failure (0 mA) from a valid signal.
If the current source is of good quality, current loops tend to be less sensitive to noise pickup by electromagnetic
interference than voltagebased signals.

Noise and Grounding
In transmitting analog signals across a process plant or factory floor, one of the most critical requirements is the
protection of data integrity. However, when a data acquisition system is transmitting low level analog signals over wires,
some signal degradation is unavoidable and will occur due to noise and electrical interference. Noise and signal
degradation are two basic problems in analog signal transmission.

Noise is defined as any unwanted electrical or magnetic phenomena that corrupt a message signal. Noise can be
categorized into two broad categories based on the sourceinternal noise and external noise. While internal noise is
generated by components associated with the signal itself, external noise results when natural or manmade electrical or
magnetic phenomena influence the signal as it is being transmitted. Noise limits the ability to correctly identify the sent
message and therefore limits information transfer. Some of the sources of internal and external noise include:

1. Electromagnetic interference (EMI).

2. Radiofrequency interference (RFI).
3. Leakage paths at the input terminals.

4. Turbulent signals from other instruments.

5. Electrical charge pickup from power sources.

6. Switching of highcurrent loads in nearby wiring.

7. Selfheating due to resistance changes.

8. Arcs.

9. Lightning bolts.

10. Electrical motors.

11. Highfrequency transients and pulses passing into the equipment.

12. Improper wiring and installation.

13. Signal conversion error, and

14. Uncontrollable process disturbances.
Digital Inputs
Many types of digital input signals from switch closures relay contacts, or TTLcompatible interfaces can be read directly
by digital I/O cards (Figure 3.14). Other types of inputs may require some signal conditioning, most likely to reduce
higherlevel voltage changes to TTL levels. A variety of signal conditioning modules are available to provide isolation and
other digital conditioning functions.

Figure 3.14: Signal Processing Requirement

For Digital and Analog Signals
The most common type of digital input is the contact closure (Figure 3.15). Essentially a sensor or switch of some type
closes or opens a set of contacts in accordance with some process change. An applied electrical signal then determines
whether the circuit is open or closed. Current flows if the circuit is closed, registering a "1" in a transistor at the computer
interface. Conversely, an open circuit retains a high voltage (and no current), registering a "0" at the transistor.

Figure 3.15: Digital Input Applied to a Contact Closure

Another type of digital input useful in data acquisition applications is the hardware trigger. This allows an external event
—a high reactor temperature, perhaps, or a low tank levelto control data collection. If during routine operation data is
only being acquired for archival storage on a oncepersecond basis, a hardware trigger can be used to boost the data
acquisition rate during an upset until normal conditions are restored.

Digital Outputs
At its simplest, a digital output provides a means of turning something on or off. Applications range from driving a relay
to turning on an indicator lamp to transmitting data to another computer. For latching outputs, a "1" typically causes the
associated switch or relay to latch, while a "0" causes the switch to unlatch. Devices can be turned on or off, depending
on whether the external contacts are normally open or normally closed.

Standard TTL level signals can be used to drive 5V relay coils; a protective diode is used to protect the digital output
circuitry (Figure 3.16). Because data acquisition boards can typically supply only 24 mA of driving current, they are
intended primarily to drive other logic circuits, not final control elements. Scaling may be needed so that logical voltage
levels are sufficient to cause switching in larger relays. Outputs intended to drive larger solenoids, contactors, motors, or
alarms also may require a boost.

Figure 3.16: Digital Output Applied to a Relay

Pulse I/O
A somewhat separate class of digital I/O is pulse inputs and outputs, which typically is associated with frequency,
counting, or totalization applications. Pulse inputs might be used to count the rotations of a turbine flow meter; pulse
outputs might be used to drive a stepping motor.

Pulse inputs are handled in much the same way as digital logic inputs, but the output of the sensing circuit is normally
connected to a counter rather than a specific bit position in the input register. Successive pulses increment or decrement
the counter. Add an elapsed time measure and a frequency or pulse rate can readily be determined. Similar to an analog
todigital converter, a counter is characterized by its number of bits—an Nbit counter can accumulate up to 2N discrete
events. Thus, a 16bit counter can count to 216 = 65,536.
ANALOG VS. DIGITAL TRANSMISSION
There are three predominant methods of encoding a transmission signal. Amplitude modulation (AM) and Frequency
Modulation (FM) are both analog modulation schemes. The third method is digital modulation. The Table 3.1 outlines the
basic characteristics of the three modulation schemes.

Table 3.1: Comparison of AM, FM and Digital Encoding Techniques
Open table as spreadsheet
Parameter AM FM Digital
Signaltonoise ratio Lowtomoderate Moderatehigh High
Performance vs. Attenuation Sensitive Tolerant Invarian
Transmitter cost Moderatehigh Moderate High
Receiver cost Moderate Moderatehigh High
Receiver gain adjustment Often required Not required Not required
Installation Adjustments requires No adjustments No adjustments
Required Required
Multichannel capabilities Require high linearity Fewer channels Good
Optics
Performance over time Moderate Excellent Excellent
Environmental factors Moderate Excellent Excellent

AM, FM and digital modulation are described in detail in other sections of this web site. One key difference between
analog and digital transmission involves the bandwidth, or transmission capacity required for both schemes. Analog
signals require much less bandwidth, only about 4.5 MHz with a 143.2 Mb/s data rate. For the average NTSC video
signal. By comparison, some digital video transmission standards require as much as 74.25 MHz with a data rate of
1485 Mb/s. Advances in singlemode optical fiber make these higher rates more accessible for longer distances. Copper
coax fails to perform at these data rates.
Another difference between analog and digital transmission deals with the hardware's ability to recover the transmitted
signal. Analog modulation, which is continuously variable by nature, can often require adjustment at the receiver end in
order to reconstruct the transmitted signal. Digital transmission, however, because it uses only 1's and 0's to encode the
signal, offers a simpler means of reconstructing the signal. Both types of modulation can incorporate error detecting and
error correcting information to the transmitted signal. However, the latest trend in signal transmission is Forward Error
Correcting (FEC). This scheme, which uses binary numbers, is suited to digital transmission. Extra bits of information
are incorporated into the digital signal, allowing any transmission errors to be corrected at the receive end.

A third important difference relates to the cost of analog transmission links compared to digital transmission links.
Because the circuitry required for digital transmission is more complex, the cost is often much higher. In short distance
applications, analog modulation will almost always be the most costeffective system to specify. However, today's
demand for high speed Internet, video on demand, videoconferencing, and "pushed" data directly to our home computers
requires moderate to longdistance transmission systems to specify digital equipment. And as is the case with any form
of technology, greater demand will lead to mass production, inevitably driving the cost of digital systems down. However,
it will always be true that the decision to specify one type of modulation over the other involves the same system design
considerations.

Signal Conversion
Conversion is the changing of a signal from one class type to the other. Analog signals are readily available. With this in
mind, we can safely say the natural world is analog in nature. The modern world of computers is digital and because we
live in this modern world, everything around us appears to be going digital, hence the need for conversion. Conversion
from digitaltoanalog and analogtodigital is achieved with the assistance of a device called a modem.

Without the use of a modem, it would be impossible to accomplish computer communication across analog circuits. A
modem enables us to send digital information across an analog circuit.

Resolution and Aliasing
Most sensors for measuring temperature, pressure, and other continuous variables provide a continuously varying
electrical output to represent the magnitude of the variable in question. To make this signal interpretable by a
microprocessor, it must be converted from a smooth continuous value to a discrete, digital number as in figure 3.17.

Figure 3.17: The AnalogtoDigital Interface

This AnalogtoDigital (A/D) conversion process poses two primary challenges: one of quantization and one of sampling
in time as in figure 3.18. Quantization refers to the uncertainty introduced upon conversion of an analog voltage to a
digital number. Measurement transducers or transmitters typically provide continuously varying signals between 0–10 V
dc, ±5 V dc, 0–100 mV dc, or 4–20 mA dc. Thermocouples and resistance temperature devices (RTDs) are other
common low voltage inputs.

Figure 3.18: A/D Conversion Compromises

When this analog value is represented as a digital number, however, this essentially continuous resolution is limited to
discrete steps. This resolution of an A/D conversion often is stated in terms of bitsthe more bits the finer the resolution.
The number of bits determines the number of divisions into which a fullscale input range can be divided to approximate
an analog input voltage. For example, 8bit resolution of a 010 V input signal means that the range is divided into 28 =
256 steps. This yields a step, or interval, size of 10 V/256 = 0.039 V. Thus, a 10V input is equal to the digital number
255 and a 0V input corresponds to 0. Each 0.039V change in the input is indicated by adding or subtracting 1 from the
previous number. (For example, 9.961 V is digitally represented by 254.)

Digital data acquisition systems not only quantize data in terms of magnitude; time, too, is parceled into discrete
intervals as in figure 3.18. In general, there is no information about the behavior of the process between data points
gathered. Special precautions, then, must be taken to ensure no meaningful data is lost and interpolation between
recorded points remains a valid assumption.

The Nyquist theorem defines the necessary relationship between the highest frequency contained in a signal and the
minimum required sampling speed. Nyquist stated that the sample rate must be at least twice the highest frequency
component contained within the input signal. So, to sample a 1Hz sine wave, the sample rate should be at least 2 Hz.
But a rate of 816 Hz would be much better for resolving the true shape of the wave.

The primary implications of ignoring the Nyquist criterion include not only missing high frequency information but of
introducing aliasing; if the sample rate is not fast enough, the presence of totally nonexistent frequencies may be
indicated as in figure 3.19. It is aliasing that makes a helicopter's rotors or a car's wheels appear to turn slowly
backwards when seen in a movie. Lowpass or antialiasing filters can be used to limit the measured waveform's
frequency spectrum so that no detectable component equals or exceeds half of the sampling rate.

Figure 3.19: Aliasing Due to Slow Sample Rate

Designing or specifying a device for A/D conversion consists of a series of tradeoffs. As will be amply demonstrated in
the next section, more resolution (more bits) means more accurate conversion but more expensive hardware. Similarly,
slower sample rates mean cheaper A/D conversion, but the Nyquist criterion must still be satisfied.
NETWORKS
A network is a circuit made up of elements which work together to support the transfer of information.

Analog Networks
These networks were used for decades until around 1980. Traffic was principally speech with low speed data. The
composition of an end to end long distance circuit included signaling, multiplex, cable or radio line and transmission
equipment. Accumulation of noise with circuit length and special conditioning needed for traffic are some of the
disadvantages of analog circuits.

Digital Networks
Communication is going digital and a lot of these networks are been used to replace what started off as analog. Digital
networks offer superior quality service for voice traffic. High speed data are now possible because of these networks.
High capacity of channels was introduced with digital systems using fiber optics.

Digital networks comprise of computer network elements, therefore, they can be managed easily and effectively. They
can also be easily upgraded.
DATA TRANSMISSION CIRCUITS
Here the three main types of transmission circuits, simplex, half duplex and full duplex are described.

Simplex

Data in a simplex channel is always one way. Simplex channels are not often used because it is not
possible to send back error or control signals to the transmit end. An example of a simplex channel in a
computer system is the interface between the keyboard and the computer, in that key codes need only be
sent one way from the keyboard to the computer system.

Figure 3.20: Simplex (oneway) Channel

HalfDuplex

A halfduplex channel can send and receive, but not at the same time. It's like a onelane bridge where two
way traffic must give way in order to cross. Only one end transmits at a time, the other end receives.

Figure 3.21: HalfDuplex Channel

FullDuplex

Data can travel in both directions simultaneously. There is no need to switch from transmit to receive mode
like in halfduplex. It's like a two lane bridge on a twolane highway.

Figure 3.22: FullDuplex Channel
PARALLEL AND SERIAL DATA
Data may be transmitted between two points in two different ways. Let's consider sending 8 bits of digital data (1 byte).

Parallel

Each bit uses a separate wire. The organization looks likes,

Figure 3.23: Parallel Interface

To transfer data on a parallel link, a separate line is used as a clock signal. This serves to inform the
receiver when data is available. In addition, another line may be used by the receiver to inform the sender
that the data has been used, and it's ready for the next data.

Figure 3.24: Parallel Interface Timing

Serial

Each bit is sent over a single wire, one after the another. The organization looks like,
Figure 3.25: Serial Interface

No signal lines are used to convey clock (timing information). There are two ways in which timing information is encoded
with the signal so that the sender and receiver are synchronized (working on the same data at the same time). If no
clock information was sent, the receiver would misinterpret the arriving data (due to bits being lost, going too slow).

Parallel transmission is obviously faster, in that all bits are sent at the same time, whereas serial transmission is slower,
because only one bit can be sent at a time. Parallel transmission is very costly for anything except short links.
A/D CONVERSION
Continuous electrical signals are converted to the digital language of computers using analogtodigital (A/D) converters.
An A/D converter may be housed on a PC board with associated circuitry or in a variety of remote or networked
configurations. In addition to the converter itself, sampleandhold circuits, an amplifier, a multiplexer, timing and
synchronization circuits, and signal conditioning elements also may be on board as shown in figure 3.26. The logic
circuits necessary to control the transfer of data to computer memory or to an internal register also are needed.

Figure 3.26: Analog Input Flow Diagram

When determining what type of A/D converter should be used in a given application, performance should be closely
matched to the requirements of the analog input transducer(s) in question. Accuracy, signal frequency content,
maximum signal level, and dynamic range all should be considered.

Central to the performance of an A/D converter is its resolution, often expressed in bits. An A/D converter essentially
divides the analog input range into 2N bins, where N is the number of bits. In other words, resolution is a measure of the
number of levels used to represent the analog input range and determines the converter's sensitivity to a change in
analog input. Amplification of the signal, or input gain, can be used to increase the apparent sensitivity if the signal's
expected maximum range is less than the input range of the A/D converter. Because higher resolution A/D converters
cost more, it is especially important to not buy more resolution than you needif you have 1% accurate (1 in 100)
temperature transducers, a 16bit (1 in 65,536) A/D converter is probably more resolution than you need.

Absolute accuracy of the A/D conversion is a function of the reference voltage stability (the known voltage to which the
unknown voltage is compared) as well as the comparator performance. Overall, it is of limited use to know the accuracy
of the A/D converter itself. Accuracy of the system, together with associated multiplexer, amplifier, and other circuitry is
typically more meaningful.

The other primary A/D converter performance parameter that must be considered is speedthroughput for a multichannel
device. Overall, system speed depends on the conversion time, acquisition time, transfer time, and the number of
channels being served by the system:

Acquisition is the time needed by the frontend analog circuitry to acquire a signal. Also called aperture time, it is the
time for which the converter must see the analog voltage in order to complete a conversion. Conversion is the time
needed to produce a digital value corresponding to the analog value. Transfer is the time needed to send the digital value
to the host computer's memory. Throughput, then, equals the number of channels being served divided by the time
required to do all three functions.

A/D Converter Options
While all analogtodigital converters are classified by their resolution or number of bits, how the A/D circuitry achieves
this resolution varies from device to device. There are four primary types of A/D converters used for industrial and
laboratory applicationssuccessive approximation, flash/parallel, integrating, and ramp/counting. Some are optimized for
speed, others for economy, and others for a compromise among competing priorities (Table 3.2). Industrial and lab data
acquisition tasks typically require 12 to 16 bits12 are the most common. As a rule, increasing resolution results in higher
costs and slower conversion speed.

Table 3.2: Alternative A/D Converter Designs
Open table as spreadsheet
Design Speed Resolution Noise Immunity Cost
Successive approximation Medium 10–16 bits Poor Low
Integrating Slow 12–18 bits Good Low
Ramp/counting Slow 14–24 bits Good Medium
Flash/parallel Fast 4–8 bits None High
Successive Approximation
The most common A/D converter design used for general industrial and laboratory applications is successive
approximation as in figure 3.27. This design offers an effective compromise among resolution, speed, and cost. In this
type of design, an internal digitaltoanalog (D/A) converter and a single comparatoressentially a circuit that determines
which of two voltages is higherare used to narrow in on the unknown voltage by turning bits in the D/A converter on until
the voltages match to within the least significant bit. Raw sampling speed for successive approximation converters is in
the 50 kHz to 1 MHz range.

Figure 3.27: A/D Conversion by Successive Approximation

To achieve higher sampling speeds, a redundancy technique allows a fast initial approximate conversion, followed by a
correction step that adjust the least significant bit after allowing sufficient settling time. The conversion is therefore
completed faster at the expense of additional hardware. Redundancy is useful when both high speed and high resolution
are desirable.

Flash/Parallel
When higher speed operation is required, parallel, or flashtype A/D conversion is called for. This design uses multiple
comparators in parallel to process samples at more than 100 MHz with 8 to 12bit resolution. Conversion is
accomplished by a string of comparators with appropriate references operating in parallel as in figure 3.28.

Figure 3.28: A/D Conversion by Flash/Parallel Technique

The downside of this design is the large number of relatively expensive comparators that are requiredfor example; a 12
bit converter requires 4,095 comparators.

Integrating
This type of A/D converter integrates an unknown input voltage for a specific period of time, and then integrates it back
down to zero. This time is compared to the amount of time taken to perform a similar integration on a known reference
voltage. The relative times required and the known reference voltage then yields the unknown input voltage. Integrating
converters with 12 to 18bit resolution are available, at raw sampling rates of 10–500 kHz.

Because this type of design effectively averages the input voltage over time, it also smoothes out signal noise. And, if
an integration period is chosen that is a multiple of the AC line frequency, excellent common mode noise rejection is
achieved. More accurate and more linear than successive approximation converters, integrating converters are a good
choice for lowlevel voltage signals.

Ramp/Counter
Similar to successive approximation designs, counting or ramptype A/D converters use one comparator circuit and a
D/A converter as in figure 3.29. This design progressively increments a digital counter and with each new count
generates the corresponding analog voltage and compares it to the unknown input voltage. When agreement is indicated,
the counter contains the digital equivalent of the unknown signal.

Figure 3.29: A/D Conversion by Counting/Ramp Technique

A variation on the counter method is the ramp method, which substitutes an operational amplifier or other analog ramping
circuit for the D/A converter. This technique is somewhat faster.

D/A Conversion
Analog outputs commonly are used to operate valves and motors in industrial environments and to generate inputs for
electronic devices under test. Digitaltoanalog (D/A) conversion is in many ways the converse of A/D conversion, but
tends to be generally more straightforward. Similar to analog input configurations, a common D/A converter often is
shared among multiplexed output signals. Standard analog output ranges are essentially the same as analog inputs: ±5
V DC, ±10 V DC, 0–10 V DC, and 4–20 mA DC.

Essentially, the logic circuitry for an analog voltage output uses a digital word or series of bits, to drop in (or drop out,
depending on whether the bit is 1 or 0) a series of resistors from a circuit driven by a reference voltage. This ladder of
resistors can be made of either weighted value resistors or an R2R network using only two resistor valuesone if placed
in series as in figure 3.30. While operation of the weightedvalue network is more intuitively obvious, the R2R scheme is
more practical. Because only one resistor value need be used, it is easier to match the temperature coefficients of an R
2R ladder than a weighted network, resulting in more accurate outputs. Plus, for high resolution outputs, very high
resistor values are needed in the weightedresistor approach.

Figure 3.30: Weighted Value & Single Value Resistor Networks for D/A Conversion
Key specifications of an analog output include: Settling time: Period required for a D/A converter to respond to a full
scale set point change.

Linearity: This refers to the device's ability to accurately divide the reference voltage into evenly sized increments.

Range: The reference voltage sets the limit on the output voltage achievable.
Because most unconditioned analog outputs are limited to 5 mA of current, amplifiers and signal conditioners often are
needed to drive a final control element. A lowpass filter may also be used to smooth out the discrete steps in output.
BANDWIDTH
Bandwidth has a general meaning of how much information can be carried in a given time period (usually a second) over
a wired or wireless communications link. For example, a link with a broad bandwidth—that is, a broadband link—is one
that may be able to carry enough information to sustain the succession of images in a video presentation.

More technically, bandwidth is the width of the range of frequencies that an electronic signal occupies on a given
transmission medium. Any digital or analog signal has a bandwidth.
In digital systems, bandwidth is expressed as bits (of data) per second (bps). Thus, a modem that works at 57,600 bps
has twice the bandwidth of a modem that works at 28,800 bps. In analog systems, bandwidth is expressed in terms of
the difference between the highestfrequency signal component and the lowestfrequency signal component. The
frequency of an analog signal is measured in the number of cycles of change per second, or hertz. A typical voice signal
has a bandwidth of approximately three kilohertz (3 kHz); an analog television (TV) broadcast video signal has a
bandwidth of six megahertz (6 MHz)— some 2,000 times as wide as the voice signal.

It should be remembered that a real communications path usually consists of a succession of links, each with its own
bandwidth. If one of these is much slower than the rest, it is said to be a bandwidth bottleneck.
FREQUENCY
For an oscillating or varying current, frequency is the number of complete cycles per second in alternating current
direction. The standard unit of frequency is the hertz, abbreviated Hz. If a current completes one cycle per second, then
the frequency is 1 Hz; 60 cycles per second equals 60 Hz (the standard alternatingcurrent utility frequency in some
countries).

Larger units of frequency include the kilohertz (kHz) representing thousands (1,000's) of cycles per second, the
megahertz (MHz) representing millions (1,000,000's) of cycles per second, and the gigahertz (GHz) representing billions
(1,000,000,000's) of cycles per second. Occasionally the terahertz (THz) is used; 1 THz = 1,000,000,000,000 cycles per
second. Note that these prefixes represent specific powers of 10, in contrast to the prefixes for multiples of bytes, which
represent specific powers of 2.
Computer clock speed is generally specified in megahertz and, more recently, in gigahertz.

Frequency is important in wireless communications, where the frequency of a signal is mathematically related to the
wavelength. If f is the frequency of an electromagnetic field in free space as measured in megahertz, and w is the
wavelength as measured in meters, then

And conversely
SIGNAL
1. In electronics, a signal is an electric current or electromagnetic field used to convey data from one place to
another. The simplest form of signal is a direct current (DC) that is switched on and off; this is the principle
by which the early telegraph worked. More complex signals consist of an alternatingcurrent (AC) or
electromagnetic carrier that contains one or more data streams.

Data is superimposed on a carrier current or wave by means of a process called modulation. Signal
modulation can be done in either of two main ways: analog and digital. In recent years, digital modulation
has been getting more common, while analog modulation methods have been used less and less. There
are still plenty of analog signals around, however, and they will probably never become totally extinct.

Except for DC signals such as telegraph and base band, all signal carriers have a definable frequency or
frequencies. Signals also have a property called wavelength, which is inversely proportional to the
frequency.

2. In some information technology contexts, a signal are simply "that which is sent or received," thus
including both the carrier and the data together.
3. In telephony, a signal is special data that is used to set up or control communication.
CARRIER
1. In information technology, a carrier (or carrier signal) is a transmitted electromagnetic pulse or wave at a
steady base frequency of alternation on which information can be imposed by increasing signal strength,
varying the base frequency, varying the wave phase, or other means. This variation is called modulation.
With the advent of laser transmission over optical fiber media, a carrier can also be a lasergenerated light
beam on which information is imposed.

Types of analog modulation of a carrier include Amplitude Modulation (AM), Frequency Modulation (FM),
and phase modulation. Types of digital modulation include varieties of Pulse Code Modulation (PCM),
including Pulse Amplitude Modulation (PAM), Pulse Duration Modulation (PDM), and Pulse Position
Modulation (PPM).

Carrier detect (see modem lights) is a control signal between a modem and a computer that indicates that
the modem detects a "live" carrier that can be used for sending and receiving information.

2. In the telecommunications industry, a carrier is a telephone or other company that sells or rents
telecommunication transmission services. A Local Exchange Carrier (LEC) is a local phone company and
an InterExchange Carrier (IEC or IXC) carries longdistance calls.

Digital T Carriers
As technology improved the speed at which electronic circuits could dependably operate, the need to increase the rate at
which data is transferred also increased. To respond to the need for interconnecting computers and database facilities,
the telephone company turned to supplying lines for carrying data in digital form. As well as supplying the lines, the
telephone company also established the limitations for their use. The basic designation for these lines is T carriers and
they begin with basic T1 carrier specified for the bell system voice network.

T1 carriers are digital, leased twisted –pair lines, operating as 4 – wire trunk lines, designed to handle 24 PCM voice
grade channels operating at a maximum data rate of 64 kbps each. Thus, the t1 line requires a bandwidth large enough
to handle 1.544 mbps (24.64 kbps). T1 lines are designed to interconnect stations that are up to 80 kilometers (km)
apart, using regenerative repeaters every 1.6 km. The last repeater placed before the nearest switch station is to be no
more than 0.8 km from that station. These repeaters reshape the digital data that has been distorted due to attenuation
or pulse spreading caused by effects on the signals of the signals of the propagation delays inherent in the reactive
nature of the cabling.
Each channel is sent, using time division multiplexing of eight bits per channel at a time. Since 24 channels are sent at
once, a total of 8.24 bits, or 192 data bits, are sent per transmission period. An additional frame boundary bit (like a
start/stop bit) increases the total to 193 bits per frame (transmission). Dividing the total data rate by the total number of
bits per frame yields a frame rate of 8,000 frames per second.

TCarriers Lines
Tcarrier lines are highspeed dedicated digital lines that can be leased from telephone companies. This creates an
always open, always available line between you and whomever you choose to connect to when you establish the
service. Tcarrier lines can support both voice and data transmissions and are often used to create pointtopoint private
networks. Because they are a dedicated link, they can be a costly WAN option. Four types of Tcarrier lines are
available:

T1 T1 lines offer transmission speeds of 1.544Mbps, and they can create pointtopoint dedicated digital
communication paths. T1 lines have commonly been used for connecting LANs.

T2 T2 leased lines offer transmission speeds of 6.312Mbps. They accomplish this by using 96 64Kbps B
channels.

T3 T3 lines offer transmission speeds of up to 44.736Mbps, using 672 64Kbps B channels.

T4 T4 lines offer impressive transmission speeds of up to 274.176Mbps by using 4,032 64Kbps B
channels.

Of these Tcarrier lines, the ones commonly associated with networks.

It is important to point out that Tcarrier is the designation to the technology used in the United States and Canada. In
Europe, they are referred to as Ecarriers and in Japan, Jcarriers. Table 3.3 shows the T/E/J carriers.

Table 3.3: Comparing T/E/J
Carriers
Open table as spreadsheet
Name Transmission Speed
T1 1.544Mbps
T1C 3.152Mbps
T2 6.312Mbps
T3 44.736Mbps
T4 274.176Mbps
J0 64Kbps
J1 1.544Mbps
J1C 3.152Mbps
J2 6.312Mbps
J3 32.064 Mbps
J3C 97.728Mbps
J4 397.200Mbps
E0 64Kbps
E1 2.048Mbps
E2 8.448Mbps
E3 34.368Mbps
E4 139.264Mbps
E5 565.148Mbps

MultiCarrier Modulation
MultiCarrier Modulation (MCM) is a method of transmitting data by splitting it into several components, and sending
each of these components over separate carrier signals. The individual carriers have narrow bandwidth, but the
composite signal can have broad bandwidth.

The advantages of MCM include relative immunity to fading caused by transmission over more than one path at a time
(multipath fading), less susceptibility than singlecarrier systems to interference caused by impulse noise, and enhanced
immunity to intersymbol interference. Limitations include difficulty in synchronizing the carriers under marginal
conditions, and a relatively strict requirement that amplification be linear.

MCM was first used in analog military communications in the 1950s. Recently, MCM has attracted attention as a means
of enhancing the bandwidth of digital communications over media with physical limitations. The scheme is used in some
audio broadcast services. The technology lends itself to digital television, and is used as a method of obtaining high data
speeds in Asymmetric Digital Subscriber Line (ADSL) systems. MCM is also used in Wireless Local Area Networks
(WLANs).
SAMPLING THEOREM
A technique for sending analog information on digital carriers involve translating voice or other analog signals into a train
of pulses, which are, in turn, coded into digital format.

The digital data is then transmitted and the process reversed at the receiving end. In effect, this process combines
sampling with Analog to Digital Conversion (ADC) and Digital to Analog Conversion (DAC). An advantage gained by this
method is in the reduction of noise effects usually inherent in analog transmissions.
The process of digitizing an analog signal starts by dividing the original the original signal into uniformly spaced pulses or
samples. The amplitudes of the sample pulses rise and fall with amplitude of the original signal. Specifically, this part of
the process is a form of Pulse Amplitude Modulation (PAM) since the original signal is being separated into individual
pulse (or samples), each of which has different amplitude based on the amplitude of the original signal. At the receive
end, these samples are used to reconstruct the original signal. The more frequently the samples are taken, the more
accurate is the reconstructed waveform.
QUANTIZATION
Quantization is the process of approximating sample levels into their closest fixed value. The values are preselected
and since they are fixed, they are easy to encode. The quantized waveform experiences either quantum changes in
amplitude or no change in amplitude. Given a signal, f(s), with peak voltage points of VH and v1, the size, s, of a
quantum step is determined by the relationship:

Where n is the number of steps or levels between VH and v1.
The quantized waveform appears similar to a sampleandhold approximation of the same f(s) as shown in figure 3.31.
The difference is that the normal sampleandhold result has voltage levels equivalent to the voltage level of f(s) at the
point the sample is taken.

Figure 3.31: Quantized Signal

The difference between the sampleandhold and the quantized results is called the quantization error or quantization
noise. Essentially, the process of sampling and quantizing a signal is a form of Pulse Amplitude Modulation (PAM),
where the samples produce pulses of varying amplitudes. The coding of a quantized signal is as shown in figure 3.32.
When these amplitudes are restricted to discrete quantized values, they can be assigned specific binary codes using a
method called pulse coded modulation.

Figure 3.32: Coding a Quantized Signal
MODEM
A modem (modulator/demodulator) is connected between a computer and a telephone network. It connects computers by
a telephone line. This device converts a digital signal from one computer into an audio signal (tones) suitable for
transmission over a telephone line. The tones can then be passed as normal analog speech. At the receiving end, the
tones are converted back to the digital before being applied to another computer. A modem is one device that makes it
possible for us to surf the Internet, to ‘chat’ with our friend and send them text based messages called electronic mail or
Email for short.

The fact that computers can processes information in digital form has had a deep effect on the modern world of
communication and information processing.
CODECS
Codecs (coder/decoder) make the reverse process to that of a modem. It enables us to send analog information across a
digital circuit. In short we can say it helps us to achieve analogtodigital and digitaltoanalog conversion. A CODECS is
a device designed to convert analog signals, such as voice communications into PCMcompressed samples to be sent
onto digital carriers and to reverse the process, replicating the original analog signal, at the receiver. The term CODECS
is an acronym for coder/decoder signifying the pulse coding/decoding functions of the device. Originally, CODECS
functions were managed by several separate devices, each performing the tasks necessary for PCM communication.
These functions are sampling, quantizing, analogtodigital and digitaltoanalog conversions, filtering, and companding.
Today, all of these functions are available on a single CODECS IC chip like intel's 2913.

The 2913 codec operates in one of two modes—fixed data rate or variable data rate. In the fixed data rate modes, the
chip is capable of operating with one of three fixedclock frequencies: T1's 1.544 MHz, ITU's 2.048 MHz, and also 1.536
MHz. In the variable data rate, the clock frequency can be set anywhere from 64 kHz to 4.096 MHz. The 2913 allows
selection of either mlaw or alaw for the companding function.
MODULATION AND DEMODULATION
Signals consist of two components—the information signal and the carrier signal. The transmission of any signal over
some communication medium usually involves modulation of a carrier. Prior to their transmission the information signal
and the carrier signal are combined and the process of combining these two signals is called modulation.

Example of Modulation
I am sitting at the back of a big spacious hall. I see my friend at the front. I have information which I want to pass to my
friend. Assuming for any reason I cannot get close to my friend. I write the information I want to pass to my friend on a
piece of paper and wrap the paper around a pen (carrier) and throw (transmit) the two (information and carrier) to my
friend. The information will reach her.

The high frequency wave which carries the information through a medium is called the carrier. The information is
superimposed onto the ‘carrier wave’ by modulation. Therefore the carrier provides a means of transferring information at
high frequency. Before modulation the carrier does not contain any ‘intelligence’ and it only serves to carry information.
Once the information is extracted from the carrier at the receiving end, the carrier becomes redundant, the carrier is a
single frequency generally represented as:

Modulating the carrier means changing something in the equation above. Only three things can be changed as shown
above: amplitude, frequency or phase reference. This leads to one class of analog modulation.
Changing A leads to amplitude modulation

Changing F leads to frequency modulation

Changing öleads to phase modulation.

Before being modulated by an information signal, the carrier signal has a single Frequency and steady Amplitude. After
modulation the carrier will have varying amplitude or frequency. There are various types of modulation and the following
three will be covered in this class:

Amplitude Modulation (AM)

Frequency Modulation (FM)

Phase Modulation (PM)

Figure 3.33 shows a modulator. The modulator output is shown drawn both in time spectrum and frequency spectrum. If
we look carefully at the frequency spectrum we can see that there are two new frequencies which were not present at the
beginning. These are the side frequencies we talked about when dealing with frequency division multiplexing. Fc + fm is
the upper sideband and Fc – fm is the lower sideband, i.e., the relationship between modulation and frequency division
multiplexing.

Figure 3.33: Modulator
MODULATION
Modulation is the addition of information (or the signal) to an electronic or optical signal carrier. Modulation can be applied
to direct current (mainly by turning it on and off), to alternating current, and to optical signals. One can think of blanket
waving as a form of modulation used in smoke signal transmission (the carrier being a steady stream of smoke).

Morse code, invented for telegraphy and still used in amateur radio, uses a binary (twostate) digital code similar to the
code used by modern computers. For most of radio and telecommunication today, the carrier is Alternating Current (AC)
in a given range of frequencies. Common modulation methods include:
1. Amplitude Modulation (AM), in which the voltage applied to the carrier is varied over time.

2. Frequency Modulation (FM), in which the frequency of the carrier waveform is varied in small but
meaningful amounts.

3. Phase Modulation (PM), in which the natural flow of the alternating current waveform is delayed
temporarily.

These are sometimes known as continuous wave modulation methods to distinguish them from Pulse Code Modulation
(PCM), which is used to encode both digital and analog information in a binary way. Radio and television broadcast
stations typically use AM or FM. Most twoway radios use FM, although some employ a mode known as Single
Sideband (SSB).

More complex forms of modulation are Phase Shift Keying (PSK) and Quadrature Amplitude Modulation (QAM). Optical
signals are modulated by applying an electromagnetic current to vary the intensity of a laser beam.

1. digital data, digital signal:

NRZ

Multilevel binary

Biphase

Modulation rate
Scrambling technique

2. digital data, analog signal

3. analog data, digital signal

4. analog data, analog signal

Amplitude Shift Key Modulation
In this method the amplitude of the carrier assumes one of the two amplitudes dependent on the logic states of the input
bit stream. A typical output waveform of an ASK modulator is shown in figure 3.34. The frequency components are the
USB and LSB with a residual carrier frequency. The low amplitude carrier is allowed to be transmitted to ensure that at
the receiver the logic 1 and logic 0 conditions can be recognized uniquely.
Figure 3.34: Amplitude Shift Key Modulation

Frequency Shift Key Modulation
In this method the frequency of the carrier is changed to two different frequencies depending on the logic state of the
input bit stream. The typical output waveform of an FSK is shown in figure 3.35. Notice that logic high causes the centre
frequency to increase to a maximum and a logic low causes the centre frequency to decrease to a minimum.

Figure 3.35: Frequency Shift Key Modulation

Phase Shift Key Modulation
With this method the phase of the carrier changes between different phases determined by the logic states of the input
bit stream.

There are several different types of Phase Shift Key (PSK) modulators.

1. TwoPhase (2 PSK)
2. FourPhase (4 PSK)

3. EightPhase (8 PSK)

4. SixteenPhase (16 PSK)

5. SixteenQuadrature Amplitude (16 QAM).

The 16 QAM is a composite modulator consisting of amplitude modulation and phase modulation. The 2 PSK, 4 PSK, 8
PSK and 16 PSK modulators are generally referred to as Binary Phase Shift Key (BPSK) modulators and the QAM
modulators are referred to as Quadrature Phase Shift Key (QPSK) modulators.

TwoPhase Shift Key Modulation
In this modulator the carrier assumes one of two phases. Logic 1 produces no phase change and logic 0 produces a 180°
phase change. The output waveform for this modulator is shown in figure 3.36.

Figure 3.36: Phase Shift Key Modulation

FourPhase Shift Key Modulation
With 4 PSK, 2 bits are processed to produce a single phase change. In this case each symbol consists of 2 bits, which
are referred to as a dibit. The actual phases that are produced by a 4 PSK modulator are shown in figure 3.37.

Dibit Phase Change
00 +225°/135°
01 +135°/225°
10 315°/45°
11 45°/315°
Open table as spreadsheet

Figure 3.37: PSK Table

Because the output bit rate is less than the input bit rate, this results in a smaller bandwidth. A typical 4 PSK circuit and
the constellation are shown in figure 3.38.
Figure 3.38: FourPhase Shift Key Modulation: (a) Circuit; (b) Constellation
PULSE CODE MODULATION
PCM (Pulse Code Modulation) is a digital scheme for transmitting analog data. The signals in PCM are binary; that is,
there are only two possible states, represented by logic 1 (high) and logic 0 (low). This is true no matter how complex the
analog waveform happens to be. Using PCM, it is possible to digitize all forms of analog data, including fullmotion video,
voices, music, telemetry, and Virtual Reality (VR).

To obtain PCM from an analog waveform at the source (transmitter end) of a communications circuit, the analog signal
amplitude is sampled (measured) at regular time intervals. The sampling rate, or number of samples per second, is
several times the maximum frequency of the analog waveform in cycles per second or hertz. The instantaneous
amplitude of the analog signal at each sampling is rounded off to the nearest of several specific, predetermined levels.
This process is called quantization. The number of levels is always a power of 2 – for example, 8, 16, 32, or 64. These
numbers can be represented by three, four, five, or six binary digits (bits) respectively. The output of a pulse code
modulator is thus a series of binary numbers, each represented by some power of 2 bits.
At the destination (receiver end) of the communications circuit, a pulse code demodulator converts the binary numbers
back into pulses having the same quantum levels as those in the modulator. These pulses are further processed to
restore the original analog waveform.

Differential Pulse Code Modulation
Differential Pulse Code Modulation (DPCM): Pulsecode modulation in which an analog signal is sampled and the
difference between the actual value of each sample and its predicted value, derived from the previous sample or
samples, is quantized and converted, by encoding, to a digital signal. There are several variations of differential pulse
code modulation.
The quantization in the section quantizing a signal requires not a priori knowledge about the transmitted signal. In
practice, you can often make educated guesses about the present signal based on past signal transmissions. Using
such educated guesses to help quantize a signal is known as predictive quantization. The most common predictive
quantization method is Differential Pulse Code Modulation (DPCM).

Delta Modulation
Delta Modulation (DM): Analogtodigital signal conversion in which (a) the analog signal is approximated with a series of
segments, (b) each segment of the approximated signal is compared to the original analog wave to determine the
increase or decrease in relative amplitude, (c) the decision process for establishing the state of successive bits is
determined by this comparison, and (d) only the change of information is sent, i.e., only an increase or decrease of the
signal amplitude from the previous sample is sent whereas a nochange condition causes the modulated signal to remain
at the same 0 or 1 state of the previous sample.

Information on Delta Modulation
Delta modulation can be considered to be an incremental PulseCode Modulation (PCM) system where, rather than
transmitting the absolute amplitude at each sampling, only the changes in signal amplitude from sampling instant to
sampling instant are transmitted. This provides a simpler system and requires less information to be transmitted. The
range of signal amplitudes is divided into a similar number of quantization levels as in pulsecode modulation systems.
However, at each sampling, the presence or absence of only one transmitted pulse contains the information. The delta
modulation code is therefore a oneelement code in which a ‘1’ is transmitted if the sampled signal at any sampling
instant is greater than the immediately previous sampling instant. A ‘0’ is transmitted if the sampled signal is lower. The
step size is called delta (the Greek letter) and hence the name delta modulation.

analog data, analog signal

Amplitude Modulation (AM)
In Amplitude Modulation (AM), the information signal is mixed with the carrier signal in such a way as to cause the
amplitude of the carrier to vary at the frequency of the information signal.
Figure 3.39: Amplitude Modulation

Amplitude Modulation (AM) uses the same scheme as AM radio. This simple technique often requires very lowcost
hardware. Two basic types of AM techniques include base band and RF carrier. In a base band system, the input signal
directly modulates the strength of the transmitter output, in this case light. In the RF carrier AM technique, a carrier, with
a frequency much higher than the encoded information, varies according to the amplitude of the information
being encoded. In a fiber optic system, the magnitude of the voltage input signal directly translates into a corresponding
light intensity.

Figure 3.40: AM System Operation

In order to use AM transmission over optical fiber, there must be compensation for optical link loss budget. This problem
may be solved two ways: by taking advantage of some special property of the input waveform (for video, the sync pulse
is an invariant that can be used to distinguish optical loss from signal level variation) or using a technique that allows the
signal level to be interpreted independently from optical loss. To accomplish this, one could send a pilot tone at a high
frequency that is above or below the frequency of the information being encoded.

Three techniques for AM video transmission include base band AM, RF carrier AM, and vestigialsideband AM. Studying
the corresponding frequency spectra of these modulation schemes, illustrated in figure 3.41, allows for a clearer
understanding of their differences. Simple base band AM occupies the region from DC to about 5 MHz and requires the
least bandwidth (assuming we are talking about uncompressed digital encoding techniques). The RF carrier modulation
spectrum is similar; it has been shifted to a nonzero frequency (F). This approach requires additional bandwidth and
offers no advantage over baseband operation in a single channel per fiber system. However, it does allow multiple
channels to be combined onto a single fiber. With vestigialsideband AM, the spectrum again shifts to a nonzero
frequency (F), and filtering removes the lower sideband. It allows for more efficient use of the spectrum as compared to
straight RF carrier AM, requiring half the bandwidth per channel.
Figure 3.41: Frequency Spectra of Three AM Transmission Techniques

AM modulation has two main drawbacks. First, the system requires highly linear components to prevent signal distortion
as it travels through the communication link. Second, because varying the light intensity encodes the signal, the receiver
cannot necessarily differentiate between intended signal level variations and the optical loss that occurs naturally in the
fiber itself. For instance, providing a 100% maximum signal into the optical transmitter with 10 dB of optical loss between
the transmitter and receiver, the receiver would indicate that the signal level is 10%. The receiver cannot readily
distinguish between changes in signal level and changes in optical loss due to the fiber. Automatic gain control can
compensate for these losses in an AM link. One approach uses a sophisticated circuit that analyzes the input waveform
and sets the sync pulse level to the required 40 IRE units. Figure 3.42 illustrates this.

Figure 3.42: Sync Pulse Level AGC

The need for highly linear components can erase much of AM's advantage over other techniques because of the
expense associated with obtaining highly linear LEDs. In spite of the difficulties mentioned above, AM systems
represent the simplest and least expensive approach to encoding information for fiber optic transmission.

Frequency Modulation (FM)
With frequency modulation, the modulating signal and the carrier are combined in such a way that causes the carrier
frequency (fc) to vary above and below its normal (idling) frequency. The amplitude of the carrier remains constant as
shown in figure 3.43.

Figure 3.43: Frequency Modulation
As the voltage of the modulating signal increases in the positive direction from A to B, the frequency of the carrier is
increased in proportion to the modulating voltage. As the voltage reduces from B to C, the frequency of the carrier
decreases until it is back to the original value at C. The modulating voltage increases in the negative direction from C to;
hence the carrier frequency decreases to a minimum value at D. It will start increasing again from D to E until it reaches
its original value again. Thus in one cycle, the frequency has gone up and down (i.e., above and below the idling
frequency) as mentioned earlier.

Throughout this process the amplitude of the carrier is not affected. The carrier frequency change above and below that
of the unmodulated condition is proportional to sign and amplitude of the modulating signal.

Phase or Angle Modulation (PM or Indirect FM)
Phase modulation is a type of frequency modulation. Here, the amount of the carrier frequency shift is proportional to
both the amplitude and frequency of the modulating signal.

The phase of the carrier is changed by the change in amplitude of the modulating signal as shown in figure 3.44.

Figure 3.44: Phase Modulation

The modulated carrier wave is lagging the carrier wave when the modulating frequency is positive. This can be clearly
seen if we concentrate on the peak amplitude along line AB (previous slide). When the modulating frequency is negative,
the modulated carrier wave is leading the carrier wave. This can be seen clearly by looking at the peak amplitude along
line CD (previous slide).
RS232C INTERFACE
Several interface standards have been formulated to interconnect any one of the many modems also known as data sets
of installation and environmental shifts during its lifetime. An additional strength member and an outer jacket surround
the inner jacket to bring the overall cable size to between 0.1 and 0.15 millimeters in diameter.

A fiberoptic cable is very small in diameter, which accounts for its lightweight. Thousands of these fiber cables can be
bundled together and packed into a tough sheathing. This provides for the ruggedness required when laying miles and
miles of fiber cables. This is done to protect against breaks in the cable under normal usage. Attempting to repair a
break in a fiber optic bundle by "splicing" cables together is much too expensive. Each cable would have to be
accurately aligned in the splice so that the light from or other Data Communications Equipment (DTC) systems made
today.
One of the more popular and widely used standards is EIA's RS232C. This standard designates the physical connector
type used, the electrical characteristics, and functions of the signals applied to the lines of the interconnect cabling
between the terminal and modem. The RS232character standard has been incorporated by numerous standards agencies
into their suites of protocols and specifications.
RS232C FUNCTIONS
Serial data is sent out on the transmit data (TX data) line, BA. The RS232C standard specifications require that the
output voltage level of a driving amplifier for mark (logic 1) data lie between −5 and −15 volts and for space (logic 0)
data lies between +5 and +15 volts. Terminating circuits receiver serial data on the receive serial data on the receive
data (rx data) line, bb, which recognizes a mark (logic 1) level as a voltage of at least −3 volt and a space (logic 0) level
of at least +3 volts providing for a 2 volt noise immunity between sending and receiving data. Thus, the data lines
operate using negative logic since logic 1 is more negative than logic 0.

For all control lines, the voltage levels have the same values as the data lines, but the logic type is positive logic. An
"on" condition uses positive voltages while an "off" condition is indicated by negative voltage values. The main control
signals used to establish a data link is:
1. RTS (requesttosend)— sent by the UART to the modem, this signal begins the process by which a data
link is established. The modem, after a short delay, responds by asserting its CTS (cleartosend). The
purpose of these local handshake signals is to establish a communications link between the modem and
the UART. The delay between RTS and CTS allows the modem circuits to come on and settle before data
can be sent to it by the UART.

CD (carrier detect), dad (data carrier detect) or RLSD (receiver line signal detect)— when a modem detects a RTS input
from the UART, one of its responses is to start sending a mark logic 1 tone onto the telephone line. On the receiving
side, when a modem detects the presence of the mark tone on the telephone line, it responds by asserting the CD signal
as a remote handshake signal. The UART detects an active CD and interprets it to mean that a remote station is
connected online and the data link is completed.

DTR (data terminal ready) from DTE and DSR (data set ready) from the modem these lines are used to inform the two
main units (DTE and modem) in the data link that they are electrically ready to establish a data link. In some systems,
these signals are connected directly to positive voltage sources. In this case, they only indicate that the RS232character
cables firmly connected between the two units. Other systems generate an active DTR when the DTE is up and running
(actually I a ready state). Modems in this type of system assert DSR shortly after the modem is powered and operating,
again, essentially saying that it is "ready".

Common signal return—This line establishes a common signal reference point between the modem and DTE.

RI (ring indicator)— This line is asserted by modern capable of detecting an incoming telephone ring signal. These
modems are equipped to automatically answer the call. A DTE connected to an autoanswering modem responds to the
RI signal by asserting its beginning the process required to establish the data link.

The remaining lines are used for secondary channel signaling and for timing control for modems requiring data stream
synchronization. A signal quality detector line cg is an available for DCE's that have the electrical means for error
detection incorporated into them.

Secondary channel data rates are much lower than the data channel. Their primary use is for control signal information.
Frequently, they are configured to send acknowledgement or control data in the opposite direction from the data channel.
When they are configured this way, the secondary channel is referred to as a back channel.
RS449 INTERFACE STANDARD
The RS449 standard contains most of the signal lines and electrical specifications outlines in the RS232c. It specifies
two connectors, one with 37 pins for the main data channel and one containing 9 pins for secondary channel signaling.
The increased number of pins is used to accommodate the added functions. A number of pins shares the same function
to assist in reducing line capacitance. This allows for data transfer rates up to 2 mbps over cables that can reach a
maximum of 200 feet in length. The RS449 pin assignment and connector are as shown in respectively.

New functions include control indications for local and remote loopback test function. Local loopback (ll) indicates the
system is in the local loop—back test mode. Data placed on the transmit data line is looped back through the modem
onto the receiver data line to the DTE. This mode is used to test the modem functions. Since data is not placed onto the
medium, the test is restricted to the local equipment at a station. In contrast, the remote loopback test mode, checks
the telephone system connections and lines as well as the modem at both stations. The data sent by a station is looped
back through the remote modem onto the telephone lines back to the sending station. The remote DTE is not included in
this test mode. A third control line, test control line, test mode TM signals the DTE that the modem is in one or the other
test mode. A similar type of control signal, the data mode (dm) indicates that the modem is in a normal data transfer
mode. This last control line replaces the DSR function of the RS232character standard.

Figure 3.45: RS449 Pin Connector
RS422 AND RS423 INTERFACE STANDARDS
The RS232C and RS449 specifications detail requirements for interfacing DTE and DCE equipment. Two individual
circuits used for transferring serial data. They are the RS422 for balanced circuits and RS423 for unbalanced circuits.
Balanced networks generate two outputs that are the complements of one another. The voltage (or power) of each output
(v1 and v2) can be measured in relation to a common reference point. However, the full output is generally measured
differentially between both outputs (v3).

The output of an unbalanced circuit is taken from a single output with reference to a common return such as ground.
Balanced networks are used to assist in the elimination of noise experienced on the lines between the generator and
receiver units. Unlike the desired outputs, noise on one output is approximately identical to the noise from the other
output. The potential difference between the noise levels is negligible. This reduction in noise power is called common
mode rejection and results in less noise presented to the receiver from the generator.

The RS422 and RS423 specifications set requirements for balanced and unbalanced digital interface circuits. Another
area specified by RS422 is the length of the cable between the generator and receiver. As with the RS232C standard,
RS422 specifies the capacitance per foot as the limiting factor to a cable's length rather than attempting to specify actual
lengths.

Open table as spreadsheet
RS449 Pin Assignment
37Pin Connector (Main Channel)
PIN NO. RS232 CIRCUIT NAME DIRECTION
1 Shield Common
2 CI IS Signal Rate Indicator From DCE
3 Not Used
4, 22 TD SD Send Data From DTE
5, 23 DB ST Send Timing From DCE
6, 24 RD RD Receive Data From DCE
7, 25 RTS RS RequesttoSend From DTE
8, 26 DD RT Receive Timing From DCE
9, 27 CTS CS CleartoSend From DCE
10 LL Local LoopBack From DTE
11, 29 DSR DM DataMode From DCE
12, 30 DTR TR Terminal Ready From DTE
13, 31 RLSD RR Receiver Ready From DCE
14 RI Remote LoopBack From DTE
15 RI IC Incoming Call From DCE
16 CH SF Select Frequency From DTE
SR Signaling Rate Selector
17, 35 DA TT Terminal Timing From DTE
19 SG SG Signal Ground Common
20 RC Receive Common From DTE
23 CG SQ Signal Quality From DCE
28 IS Terminal in Servgice From DTE
32 SS Select Standby From DTE
34 NS New Signal From DTE
36 SB Standby Indicator From DCE
37 SC Send Common From DCE
9Pin Connector (S—Secondary Channel)
1 Shield Common
2 SCF SRR Receive Ready From DCE
3 SBA SSD Send Data From DTE
4 SBB SRD Receive Data From DCE
5 SG SG Signal Ground Common
RS449 Pin Assignment
37Pin Connector (Main Channel)
PIN NO. RS232 CIRCUIT NAME DIRECTION
6 RC Receive Common From DCE
7 SCA SRS RequesttoSend From DTE
8 SCB SCS CleartoSend From DCE
9 SC Send Common From DCE

Figure 3.46: RS449 Pinassignment
REVIEW QUESTIONS
1. What are the different transmission signals?
2. What are the advantages and disadvantages of analog and digital signals?
3. What are the difference between analog and digital signals?
4. Write successive approximation method of A/D conversion.
5. Write short notes on bandwidth and frequency.
6. What are digital T carriers?
7. Why do we use sampling theorem?
8. What are modulation and demodulation, explain?
9. What are the different types of modulations?
10. Why is fullduplex Frequency Shift Keying (FSK) an example of Frequency Division Multiplexing (FDM)?
11. How are FDM groups created?
12. How are multiple channels sent on a single line using TDM?
13. What is meant by Pulse Amplitude Modulation (PAM)?
14. What is the Nyquist sampling rate when sampling a signal whose highest frequency is 1630 Hz and lowest
frequency is 250 Hz?
15. What is quantizing error?
16. What happens to quantized samples in a PCM system?
17. Give four characteristics of a T1 carrier.
18. What is a CODEC and how does it differ from a modem?
19. What is the basic idea behind delta modulation?
20. Explain the two modes for transmitting binary data across a link.
21. What are the advantages and disadvantages of parallel transmission?
22. Compare the two methods of serial transmission. Discuss the advantages and disadvantages of each.
23. What is digitaltoanalog modulation?
24. What is analogtoanalog modulation?
25. Why is frequency modulation superior to amplitude modulation?
26. What is the advantage of QAM over ASK or PSK?
27. What are the methods that convert a digital signal to an analog signal?
28. What is the difference between bit rate and baud rate? Give an example where both are the same. Give an example
where they are different.
29. What is modulation?

30. What is the purpose of a carrier signal in modulation?
30. How is baud rate related to transmission bandwidth in ASK?
31. How is baud rate related to transmission bandwidth in FSK?
32. How is baud rate related to transmission bandwidth in PSK?
33. How is baud rate related to transmission bandwidth in QAM?
34. How is QAM related to ASK and PSK?
35. What is the function of a modulator? What is the function of a demodulator?
36. What does the term modem stand for?
37. Explain the asymmetry of 56 k modems.
38. Why are modems needed for telephone communications?
39. Compare the FM bandwidth with the AM bandwidth in terms of the modulating signal.
40. What are the three major multiplexing techniques?
41. How does FDM combine multiple signals into one?
42. What is the purpose of a guard band?
43. How does TDM combine multiple signals into one?
44. How are T lines related to DS service?
45. How can T lines be used for analog transmission?
46. What is inverse multiplexing?
47. Which errordetection method is used most frequently with asynchronous data streams?
48. How many errors can parity reliably detect in a single character?
49. What is one drawback when using parity for error detection?
50. How many hamming bits are used for the message in question 7?
51. Why is fullduplex Frequency Shift Keying (FSK) an example of Frequency Division Multiplexing (FDM)?
52. How are FDM groups created?
53. How are multiple channels sent on a single line using TDM?
54. What is meant by Pulse Amplitude Modulation (PAM)?
55. What are the functions of RS449 interface standard?
56. What are the voltage levels for RS232C data on the transmit line?
57. What is the benefit of using RS422 balanced interface circuits?
58. What are the main functions of a modem in a lowspeed data communications system?
59. What is the common data rate for V34 modems?
60. What is the common data rate for V90 modems?
61. What is one modulation technique used with V90 modem?
62. Which tone is initially used to cause CarrierDetect (CD) line to be asserted by a modem? What logic level does that
tone represent?
63. Describe the three characteristics of a sine wave.
64. Contrast an analog signal with a digital signal.
65. Name three types of transmission impairment.
66. What do you understand by transmission impairments? Discuss the major transmission impairments.

(IV BCA (4.1)– DCCN)
67. Differentiate between the following pairs:

i. Periodic signal—Aperiodic signal
ii. Digital signal—Analog signal

iii. Time domain—Frequency domain

iv. Half duplex—Full duplex.

(IV BCA (4.1) –DCCN)
68. Explain about RS232C interface.

(IV BCA (4.1) –DCCN)
69. Explain amplitude modulation and frequency modulation with suitable diagram.

(IV BCA (4.1) –DCCN)
Chapter 4: Multiplexing and Switching

MULTIPLEXING BASICS
Multiplexing is a process in which multiple data channels are combined into a single data or physical channel at the
source. Multiplexing can be implemented at any of the OSI layers. Conversely, demultiplexing is the process of
separating multiplexed data channels at the destination. One example of multiplexing is when data from multiple
applications is multiplexed into a single lowerlayer data packet.
Another example of multiplexing is when data from multiple devices is combined into a single physical channel (using a
device called a multiplexer).

A multiplexer is a physical layer device that combines multiple data streams into one or more output channels at the
source. Multiplexers demultiplex the channels into multiple data streams at the remote end and thus maximize the use
of the bandwidth of the physical medium by enabling it to be shared by multiple traffic sources.

Some methods used for multiplexing data are TimeDivision Multiplexing (TDM), Asynchronous TimeDivision
Multiplexing (ATDM), FrequencyDivision Multiplexing (FDM), and statistical multiplexing.
In TDM, information from each data channel is allocated bandwidth based on preassigned time slots, regardless of
whether there is data to transmit. In ATDM, information from data channels is allocated bandwidth as needed by using
dynamically assigned time slots. In FDM, information from each data channel is allocated bandwidth based on the signal
frequency of the traffic. In statistical multiplexing, bandwidth is dynamically allocated to any data channels that have
information to transmit.
MULTIPLEXING
Multiplexing is sending multiple signals or streams of information on a carrier at the same time in the form of a single,
complex signal and then recovering the separate signals at the receiving end as shown in figure 4.1 (a). In analog
transmission, signals are commonly multiplexed using FrequencyDivision Multiplexing (FDM), in which the carrier
bandwidth is divided into subchannels of different frequency widths, each carrying a signal at the same time in parallel.
In digital transmission, signals are commonly multiplexed using TimeDivision Multiplexing (TDM), in which the multiple
signals are carried over the same channel in alternating time slots. In some optical fiber networks, multiple signals are
carried together as separate wavelengths of light in a multiplexed signal using Dense Wavelength Division Multiplexing
(DWDM). The categorization of multiplexing is shown in figure 4.1 (b).

Figure 4.1: Multiplexing
INVERSE MULTIPLEXER
An inverse multiplexer (often abbreviated to "inverse mux" or "imux") allows a data stream to be broken into
multiple lower data rate communications links. An inverse multiplexer differs from a demultiplexer in that each of the
low rate links coming from it is related to the other ones and they all work together to carry the same data. By
contrast, the output streams from a demultiplexer may each be completely independent from each other and the
demultiplexer does not have to understand them in any way. Note that this is the opposite of a multiplexer which
creates one high speed link from multiple low speed ones.
MULTIPLEXING METHODS
One way to increase the capacity of data transmission is to multiplex numerous channels into a single transmission
channel, using a multiplexer (MUX). This has the advantage of requiring of multiplexing is that there is a requirement for
either additional time or bandwidth to achieve the multiplexing. There are four basic techniques used for multiplexing:

1. TimeDivision Multiplexing (TDM)
2. Frequency Division Multiplexing (FDM)

3. Wavelength Division Multiplexing (WDM)

4. Statistical Time Division Multiplexing
TimeDivision Multiplexing
TimeDivision Multiplexing (TDM) is a method of putting multiple data streams in a single signal by separating the signal
into many segments, each having a very short duration. Each individual data stream is reassembled at the receiving end
based on the timing. Using the pointtopoint conventional means of transferring data on a base band system, sending
three messages of varying lengths is accomplished a message is transmitted first, followed, in time, by second
message, and then third message. This is a simple and straight forward process.

The circuit that combines signals at the source (transmitting) end of a communications link is known as a multiplexer. It
accepts the input from each individual end user, breaks each signal into segments, and assigns the segments to the
composite signal in a rotating, repeating sequence. The composite signal thus contains data from multiple senders. At
the other end of the longdistance cable, the individual signals are separated out by means of a circuit called a de
multiplexer, and routed to the proper end users. A twoway communications circuit requires a multiplexer/demultiplexer at
each end of the longdistance, highbandwidth cable.

If many signals must be sent along a single longdistance line, careful engineering is required to ensure that the system
will perform properly. An asset of TDM is its flexibility. The scheme allows for variation in the number of signals being
sent along the line, and constantly adjusts the time intervals to make optimum use of the available bandwidth. The
Internet is a classic example of a communications network in which the volume of traffic can change drastically from
hour to hour. In some systems, a different scheme, known as FrequencyDivision Multiplexing (FDM), is preferred. The
TDM system is illustrated as in figure 4.2.

Figure 4.2: TDM System

TDM Methods
Two basic types of TDM are in use today, synchronous TDM (STDM) and asynchronous TDM (ATDM), also known as
statistical TDM (STATDM) or Stat MUX. STDM systems assign time slots of equal length to all packets regardless if
whether or not anything is to be sent by each station with an assigned time slot. STDM systems are comparatively easy
to implement once the applications software allocates the time slots. Stat MUX networks assign time slots only when
they are to be used and deletes them when they are idle. The total time used for a stat MUX frame varies with the
amount of traffic currently being handled.

Stat MUX systems are most suitable for these highdensity, high–traffic applications. A constant flow of messages are
assigned time slot and interleaved as each channel on the send side becomes active and requires communications with
another channel. If a channel does not have any traffic, its time slots are deleted and reassigned to an active channel.
This way, the interconnecting media achieves a higher state of efficiency usage than with STDM systems.

TDM Applications
Secondgeneration cellular telephone companies use TDM. For example, the digital version of the cellular telephony
divides the available bandwidth into 30 kHz bands and uses FDM to combine these bands. For each band, it applies
TDM so that six users share the band.

FrequencyDivision Multiplexing
FrequencyDivision Multiplexing (FDM) is a scheme in which numerous signals are combined for transmission on a
single communications line or channel. Each signal is assigned a different frequency (subchannel) within the main
channel.

A typical analog Internet connection via a twisted pair telephone line requires approximately three kilohertz (3 kHz) of
bandwidth for accurate and reliable data transfer. Twistedpair lines are common in households and small businesses.
But major telephone cables, operating between large businesses, government agencies, and municipalities, are capable
of much larger bandwidths.

Suppose a longdistance cable is available with a bandwidth allotment of three megahertz (3 MHz). This is 3,000 kHz, so
in theory, it is possible to place 1,000 signals, each 3 kHz wide, into the longdistance channel. The circuit that does this
is known as a multiplexer. It accepts the input from each individual end user, and generates a signal on a different
frequency for each of the inputs. This results in a highbandwidth, complex signal containing data from all the end users.
At the other end of the longdistance cable, the individual signals are separated out by means of a circuit called a de
multiplexer, and routed to the proper end users. A twoway communications circuit requires a multiplexer/demultiplexer
at each end of the longdistance, highbandwidth cable.

When FDM is used in a communications network, each input signal is sent and received at maximum speed at all times.
This is its chief asset. However, if many signals must be sent along a single longdistance line, the necessary bandwidth
is large, and careful engineering is required to ensure that the system will perform properly. In some systems, a different
scheme, known as timedivision multiplexing, is used instead.

In its simplest form, FDM is represented by a Frequency Shift Key (FSK) modem. Here two frequencies were used to
represent a one and a zero. In a fullduplex system, four sets of frequencies occupied the line with a maximum of two
signals on the line at any one time. In effect, the bandwidth of the telephone line was shared by each of the signal
frequencies. This example falls short, since it deals with single channel of communication and does not involve any form
of true multiplexing. It does, however, as shown in figure 4.3 frequency allocating the transmit frequency bands within the
bandwidth.
Figure 4.3: FDM System

FDM applies itself to the increased usage of a network's capacity by utilizing as much available frequency bandwidth as
is allocated to the system. Applications of FDM include fiber optic links, microwave, and satellite communications that
have bandwidths in the megahertz and gigahertz ranges.

Applications of FDM
A very known and common application of FDM is AM and FM radio broadcasting. Radio uses the air as the transmission
medium. A special band, from 530 to 1700 kHz, is assigned to AM radio. All radio stations need to share this band.
Another common use of FDM is in television broadcasting. Each TV channel has its own bandwidth of 6 MHz.

The concept corresponding to frequency division multiplexing in the optical domain is known as wavelength division
multiplexing. In optical communications, FDM is referred to as WavelengthDivision Multiplexing (WDM).
Wavelength Division Multiplexing
Until the late 1980s, optical fiber communications was mainly confined to transmitting a single optical channel. Because
fiber attenuation was involved, this channel required periodic regeneration, which included detection, electronic
processing, and optical retransmission. Such regeneration causes a highspeed optoelectronic bottleneck and can handle
only a single wavelength. After the new generation amplifiers were developed, it enabled us to accomplish highspeed
repeaterless singlechannel transmission. We can think of single ~Gbps channel as a single highspeed lane in a
highway in which the cars are packets of optical data and the highway is the optical fiber. However, the ~25 THz optical
fiber can accommodate much more bandwidth than the traffic from a single lane. To increase the system capacity we
can transmit several different independent wavelengths simultaneously down a fiber to fully utilize this enormous fiber
bandwidth. Therefore, the intent was to develop a multiplelane highway, with each lane representing data travelling on a
different wavelength. Thus, a WDM system enables the fiber to carry more throughput. By using wavelengthselective
devices, independent signal routing also can be accomplished.

Basic Operation
As explained before, WDM enables the utilization of a significant portion of the available fiber bandwidth by allowing
many independent signals to be transmitted simultaneously on one fiber, with each signal located at a different
wavelength. Routing and detection of these signals can be accomplished independently, with the wavelength determining
the communication path by acting as the signature address of the origin, destination or routing. Components are
therefore required that are wavelength selective, allowing for the transmission, recovery, or routing of specific
wavelengths.

In a simple WDM system figure, each laser must emit light at a different wavelength, with all the lasers. Light
multiplexed together onto a single optical fiber. After being transmitted through a highbandwidth optical fiber, the
combined optical signals must be demultiplexed at the receiving end by distributing the total optical power to each output
port and then requiring that each receiver selectively recover only one wavelength by using a tunable optical filter. Each
laser is modulated at a given speed, and the total aggregate capacity being transmitted along the highbandwidth fiber is
the sum total of the bit rates of the individual lasers. An example of the system capacity enhancement is the situation in
which ten 2.5 Gbps signals can be transmitted on one fiber, producing a system capacity of 25 Gbps. This wavelength
parallelism circumvents the problem of typical optoelectronic devices, which do not have bandwidths exceeding a few
gigahertz unless they are exotic and expensive. The speed requirements for the individual optoelectronic components
are, therefore, relaxed, even though a significant amount of total fiber bandwidth is still being utilized.

Statistical Time Division Multiplexing
Statistical time division multiplexing uses intelligent devices capable of identifying when a terminal is idle. They allocate
time only to lines when required. This means that more lines can be connected to a transmission medium as this device
statistically compensates for normal idle time in data communication line.

Figure 4.4: Diagram of a Simple WDM System

A TDM where slots are allocated on demand. Unlike synchronous TDM, empty slots are never sent. All sources
connected to a multiplexer are not always transmitting. Stat TDM exploits this property by allowing the sum of the
maximum data rates of each source to exceed the outgoing data rate. However, since almost always all the sources are
never transmitting at their maximum rate, the instantaneous sum of their data rates is within the capacity of the
multiplexer line. If there are n source and k slots per fame in the outgoing frame then k < n. During each scan of the
input, only up to k buffers are read. Statistical TDM—Issues in synchronous TDM every slot is allocated to one source,
so the demux knows where to output each slot. With statistical TDM however, each slot could contain arbitrary source
data, hence addressing information is required for each slot. This is an overhead that was not present in synchronous
TDM. If the sources maintain average data rate, stat TDM will work fine, however during peak periods, there might not be
enough slots so the input will have to buffer data till slots become available. This causes a buffering requirement at
sender, however more importantly the subsequent backlog due to buffering causes an delay in sending of data. So the
backlog/delay of frames is directly related to the output capacity of the line.
MEDIA SHARING TECHNIQUES
Media sharing occurs when two communication channels use the same media. This section presents how some
communication channels can use the same media link without architecture considerations. There are three main
techniques, they are as:

Time Division Multiple Access (TDMA)
Time Division Multiple Access (TDMA) is a technology for shared medium (usually radio) networks. It allows several
users to share the same frequency by dividing it into different time slots. The users transmit in rapid succession, one
after the other, each using there own timeslot. This allows multiple users to share the same transmission medium (e.g.,
radio frequency). So, this simple method consists on multiplexing data in time. Each user transmits a period of time
equal to 1/(number of possible channels) in full bandwidth W. This sharing mode can be synchronous or asynchronous.
Figure 4.5 shows a synchronous TDMA system. Each channel uses a time slot each T periods. Selecting a time slot
identifies one channel. Classical wired phone uses this technique.

Figure 4.5: Synchronous and Asynchronous TDMA Diagram

In synchronous TDMA, if an established channel stops transferring data without freeing the assigned time slot, the
unused bandwidth is lost and hence, other channels can not take advantage of this. This technique has evolved to
asynchronous TDMA to avoid this problem.

Figure 4.5 shows an asynchronous TDMA system. Each channel uses a time slot when the user needs to transfer data
and when a time slot is unused. The header of each time slot data stream identifies the channel identification. ATM
networks use this technique.
These two techniques are used to connect users. Providing broadcast channels in TDMA can not be done easily.
Frequency division multiple access technique avoids this problem. Next section presents this sharing mode.

Frequency Division Multiple Access (FDMA)
Frequency division multiple access (FDMA) is the oldest and most important of the three main ways for multiple radio
transmitters to share the radio spectrum. This sharing method consists on giving to each channel a piece of available
bandwidth.

Each user transmits over a constant bandwidth equal to W/(number of possible channels). Filtering with a bandwidth
equal to ‘W’ = W/(number of possible channels) the whole W bandwidth spectrum selects one channel. TV and radio
broadcasters use this media sharing technique.

Another method has been developed based on the frequency dimension. This method called code division multiple
access uses an encodingdecoding system used, initially, for military communications. Today consumer market
applications also use this technique. In the next section this method is discussed in detail.
Code Division Multiple Access (CDMA)
Code Division Multiple Access (CDMA) is a form of spread spectrum which is sent by multiple transmitters to the same
receiver on the same frequency channel at the same time without harmful interference.

Each user transmits using the full bandwidth. Demodulating the whole W band (as given in FDMA) using a given
identification code selects one channel out of the others. Next mobile phones standard (IS95 or WCDMA) uses this
media sharing technique. These techniques can be merged together. For example, the Global System Mobile (GSM =
Natel D) phone standard uses an FDMATDMA technique. After this description, we will see in next section how a
network node routes data from an input port to a given output port.

Node Sharing Technique
Node sharing occurs when two communication channels use the same network node. The question is how some
communication channels can use the same node in a cell switching network, i.e., an ATM network. Before answering
this question, we have to define the specification of the switching function. This concept is presented in next section.
MULTIPLEXING AND SIGNAL CONDITIONING
A/D converters seldom function on their own but must be considered in a systems context with associated circuitry for
signal conditioning, multiplexing, amplification, and other functions. Every application will dictate a unique mix of addons
that may be implemented in a variety of physical configurationson a PC I/O board, inside a remote transmitter, or at a
local termination panel.

Multiplexing. In many industrial and laboratory applications, multiple analog signals must be converted to digital form.
And if speed is not the limiting factor, a single A/D converter often is shared among multiple input channels via a
switching mechanism called a multiplexer. This is commonly done because of the relatively high cost of converters.
Multiplexers also allow amplification and other signal conditioning circuitry to be timeshared among multiple channels.
Software or auxiliary hardware controls the switch selection.
Sampleandhold. It is important to acknowledge that a multiplexer does reduce the frequency with which data points
are acquired, and that the Nyquist samplerate criterion still must be observed. During a typical data acquisition process,
individual channels are read in turn sequentially. This is called standard, or distributed, sampling. A reading of all
channels is called a scan. Because each channel is acquired and converted at a slightly different time, however, a skew
in sample time is created between data points.
MULTIPLEXED CHANNELS
If time synchronization among inputs is important, some data acquisition cards offer burst mode operation or
simultaneous sampleandhold circuitry. Burst mode, or pseudosimultaneous sampling, acquires each channel at the
maximum rate of the board, then waits a userspecified amount of time before sampling again.

True simultaneous sampleandhold systems can sample all channels within a few nanoseconds of each other,
eliminating phase and time discontinuities for all but the fastest processes. Essentially, a switched capacitor on each
channel tracks the corresponding input signal. Before starting the A/D conversion process, all switches are opened
simultaneously, leaving the last instantaneous values on the capacitors.
Because A/D converters work best on signals in the 110 V range, low voltage signals may need to be amplified before
conversioneither individually or after multiplexing on a shared circuit. Conversely, high voltage signals may need to be
attenuated.

Amplifiers also can boost an A/D converter's resolution of lowlevel signals. For example, a 12bit A/D converter with a
gain of 4 can digitize a signal with the same resolution as a 14bit converter with a gain of 1. It's important to note,
however, that fixedgain amplifiers, which essentially multiply all signals proportionately, increase sensitivity to low
voltage signals but do not extend the converter's dynamic range.

Programmable Gain Amplifiers (PGAs), on the other hand, can be configured to automatically increase the gain as the
signal level drops, effectively increasing the system's dynamic range. A PGA with three gain levels set three orders of
magnitude apart can make a 12bit converter behave more like an 18bit converter. This function does, however, slow
down the sample rate.
From a systems perspective, amplifier performance should be on par with that of the A/D converter itselfgain accuracy
should be specified as a low percentage of the total gain. Amplifier noise and offset error also should be low. Other
conditioning functions: Other A/D signal conditioning functions required will vary widely from application to application.

In practical telecommunication systems, a nominal channel allocation of 0–4 kHz is assumed to allow for finite cutoff
filters. Hence the 4kHz channel spacing in F.D.M. assemblies. The frequency shifting process (also called frequency
translation) of the information signal produces two new frequency bands. One is called the Upper Sideband (USB) and
the other Lower Sideband (LSB). Both are referred to as side frequencies.

In practical systems, complex multiplexing formats are encountered. For example, the Table 4.1 shows an FDM
structure for telephony which is recommended by the Consultative Committee of the International Telecommunications
Union (CCITT). CCITT is an international advisory body. The structure allows telephone calls to be combined in blocks of
appropriate sizes for transmission through a national or international telephone network.

Table 4.1: CCITT FDM Hierarchy
Open table as spreadsheet
Channel 1 channel (4 kHz)
Group 12 channels (48 kHz)
Super group (5 groups) 60 channels (240 kHz)
Master gruop (5 groups) 300 channels (1.2 MHz)
Super master group (3 groups) 900 channels (3.6 MHz)

The frequency assignments set forth by CCITT, which is based in Geneva, Switzerland, makes sure that systems using
these in various countries are compatible.

We can now take a look at the basic group (known as the standard telephone group12 channel) and see how the
information signals are combined into a composite signal for transmission over a single communication channel. It is
also worth mentioning at this point that the CCITT recommends that all multichannel systems should be based on two
alternative 12 channel group (i.e., one group should utilize the USB and the other should use the LSB).

We will take a look at an example of a 12 channel multiplex system utilizing the lower sideband. LSB BPF is a band
pass filter which will allow only the lower sidebands to pass through to the transmission line.

Synchronization
Synchronization is the coordination of occurrences to operate in unison with respect to time. This process can be a
premeditated arrangement set forth on a parallel time scape, or it can be an observable coincidence in eventuality. In
terms of digital logic and data transfer, a synchronous object requires a clock signal.

Systems operating with all their parts in synchrony are said to be synchronous or in sync. Some systems may be only
approximately synchronized, or plesiochronous. For some applications relative offsets between events need to be
determined, for others only the order of the event is important.
Interleaving
The transmission of pulses from two or more digital sources in timedivision sequence over a single path. Bit padding is
the addition of one or more extra bits to a transmission or storage unit to make it conform to a standard size. Some
sources identify bit padding as a type of bit stuffing.

Timeslots
Time Division Multiple access (TDMA) allows several users to share the same frequency by dividing it into
different timeslots. The users transmit in rapid succession, one after the other, each using their own timeslot.
MODULATION AND MULTIPLEXING
Modulation is a technique whereby an analogue signal can be changed (modulated) so as to carry information or a
message.

Communication can be achieved simply by sending a signal directly over a wire link. Such a simple method has several
disadvantages. One of them is that: only one conversation can be carried out at a time. This means that between the
number of pairs of individuals who wish to communicate and the number of lines provided, there has to be a one to one
correspondence. A subscriber has his own link to the exchange and a connection to any other line on the exchange. This
can be done for shorter distances and a small population.
It is impractical and unnecessary to do so (i.e., provide as many links as there are subscribers) for long distances and a
large population. This is where the concept of multiplexing comes into play.
MODULATION STANDARDS
There are many standards for modulation. Some terms are defined below:

Bell 103 & 212A: These are old standards. Bell 103 transmits at 300 bps at 1 bit per baud. Bell 212A was
the next step up, capable of two bits per baud. It was capable of 1200 bps of 600 baud. Each used a
different type of modulation.
V.21: Uses mainly outside the U.S., it was a sort of international standard. It is not compatible with Bell
protocols, and is only capable of 300 bps.

V.22: Shares the features of V.21, but was capable of 1200 bps. Later, the V.22bis protocol was used. It
was used both in the U.S. and outside, and ran at 4 bits per baud for a total of 2400 bps.

V.23: Used mainly in Europe, this allowed the modem to send and receive data at the same time, although
it could send data only at 75 bps. This standard was developed to lower the cost of modems. A 1200 bps
modem was very expensive at the time.

V.29: A halfduplex standard, meaning oneway. It works at 9600 bps. The standard is not laid out well for
modems, therefore isn't much used. The protocol is most often used for fax machines.

V.32: This standard began to get users into the ballpark we are all in now. It was a fullduplex standard and
operated at 9600 bps, with a 2400 baud rate. It incorporates errorcorrecting and negotiation. The error
correcting allowed V.32 to work well over phone line noise.

V.32bis: This is one step up from V.32. It transmitted 6 bits per baud, allowing throughputs of 14,400 bps.
It also allowed fallback onto regular V.32 if the phone line was impaired. Many still use this modem
standard primarily.

V.32fast: Better known as the 28,800 bps modem.
V.34: The latest real modem standard. It provides a reliable 28,800 bps connection. With upgrades to the
ROM BIOS on the modem, the standard is often used for 33,600 bps transmissions.

V.90: is the latest and fastest ITUratified "V." specification, offering a top dialup rate of 56 kbps.
REVIEW QUESTIONS
1. Discuss the basic of multiplexing.
2. What are the different methods of multiplexing?
3. What do you understand by multiplexing and modulation?
4. Discuss different switching methods.
5. Which switching technique is also known as storeandforeward?
6. Contrast message and packet switching.
7. What is multiplexing? Explain the various categories of multiplexing used in data communication.

(IV–BCA(4.1)– DCCN)
8. Explain about RS232C interface.

(IV–BCA (4.1)– DCCN)
9. Explain the Pulse Code Modulation (PCM) technique of encoding.

(IV–BCA(4.1)– DCCN)
10. Explain amplitude modulation and frequency modulation with suitable diagram.

(IV–BCA(4.1)– DCCN)
Chapter 5: Transmission Media

OVERVIEW
Cable is the medium through which information usually moves from one network device to another. There are several
types of cable which are commonly used with LANS. In some cases, a network will utilize only one type of cable, other
networks will use a variety of cable types. The type of cable chosen for a network is related to the network's topology,
protocol, and size. Understanding the characteristics of different types of cable and how they relate to other aspects of a
network is necessary for the development of a successful network.
The physical layer provides for physical connectivity between networked devices. Transmission and receipt of data from
the physical media (copper wire, fiber, radio frequencies, barbed wire, string etc.) is managed at this layer.

The physical layer receives data from the data link layer, and transmits it to the wire. The physical layer controls the
electrical and mechanical functions related to the transmission and receipt of a communications signal. It also manages
the encoding and decoding of data contained within the modulated signal.

Note that for two devices to communicate, they must be connected to the same type of physical medium (wiring). 802.3
Ethernet to 802.3 Ethernet, FDDI to FDDI, serial to serial etc. Two end stations using different protocols can only
communicate through a multiprotocol bridge or a router.
The physical layer is responsible for two jobs:

1. Communication with the data link layer above it.

2. Fragmentation of data into frames

3. Reassembly of frames into data link protocol data units.

4. Transmission and receipt of data.

It should be noted that in most modern network interface adaptors, the physical and data link functions are performed by
the adaptor.

The physical layer is the hardware level of the protocol model that is concerned with electronic signals. Physical layer
protocols send and receive data in the form of packets. A packet contains a source address, the transmission itself, and
a destination address.

TCP/IP supports a number of physical layer protocols, including Ethernet. Ethernet is an example of a packetswitching
network; its communications channels are occupied only for the duration of the transmission of a packet. On the other
hand, the telephone network is an example of a circuitswitching network.

The physical layer (layer 1) of the OSI reference model serializes the frame (i.e., converts it to a series of bits) and
sends it across a communications circuit (i.e., cable) to the destination (or an intermediate) system. There are a number
of types of circuit/cable (sometimes also called "media") that may be used, these include:

1. Open wire circuits

2. Twisted pair cables

3. Coaxial cables

4. Fiber optics cable

5. Signaling of bits

The physical layer defines the representation of each bit as a voltage, current, phase, or frequency. Four basic schemes
are used:

1. RZ, Return to Zero (Pulse signaling)

2. NRZ (non return to zero transmission)

3. Manchester encoding (Edge/phase signaling)

In NRZ transmission, each data bit is represented by a level. A high level may represent a logic 1, where as a low level
may represent a logic 0. The term is derived from the earlier transmission technique of sending pulses to represent bits
(called Return to Zero, RZ) in which a logic 1 is represented by a pulse and a logic 0 by the absence of a pulse. (AMI and
HDB3 are techniques derived from RZ). Manchester encoding uses a still different scheme where a logic 1 is represented
by a transition in a particular direction (usually a rising edge) in the centre of each bit. A transition in the opposite
direction (downward in this case) is used to represent a logic 0.
TIMING OF BITS
At the receiver, the remote system reassembles the series of bits to form a frame and forwards the frame for processing
by the link layer. A clock (timing signal) is needed to identify the boundaries between the bits (in practice it is preferable
to identify the centre of the bit—since this usually indicates the point of maximum signal power). There are two systems
used to providing timing:

1. Asynchronous communication (independent transmit & receive clocks)
2. Simple interface (limited data rate < 19.2 kbps)
TRANSMISSION MEDIA
There are two basic categories of transmission media:

Guided media

Open wire

Twisted pair

Coaxial cable
Optical fiber

Unguided Media

Wireless

Radiowaves
Microwave

Infrared

Satellite

1 Guided Media
Guided transmission media uses a cabling system that guides the data signals along a specific path. The data
signals are bound by the cabling system. Guided media is also known as bound media. "Cabling" is meant in a
generic sense, and is not meant to be interpreted as copper wire cabling only. There four basic types of guided
media:

i. Open wire

ii. Twisted pair

iii. Coaxial cable

iv. Optical fiber
i. Open Wire. Open Wire is traditionally used to describe the electrical wire strung along power poles.
There is a single wire strung between poles. No shielding or protection from noise interference is used.
We are going to extend the traditional definition of open wire to include any data signal path without
shielding or protection from noise interference. This can include multiconductor cables or single wires.
This media is susceptible to a large degree of noise and interference and consequently not acceptable
for data transmission except for short distances under 20 ft.

Figure 5.1: Open Wire

ii. Twisted Pair. The wires in twisted pair cabling are twisted together in pairs. Each pair would consist of
a wire used for the +ve data signal and a wire used for the −ve data signal. Any noise that appears on
1 wire of the pair would occur on the other wire. Because the wires are opposite polarities, they are 180
degrees out of phase (180 degrees—phasor definition of opposite polarity). When the noise appears on
both wires, it cancels or nulls itself out at the receiving end. Twisted pair cables are most effectively
used in systems that use a balanced line method of transmission: polar line coding (Manchester
encoding) as opposed to unipolar line coding (TTL logic).
Figure 5.2: Unshielded Twisted Pair

The degree of reduction in noise interference is determined specifically by the number of turns per foot.
Increasing the number of turns per foot reduces the noise interference. To further improve noise
rejection, a foil or wire braid shield is woven around the twisted pairs. This "shield" can be woven
around individual pairs or around a multipair conductor (several pairs).

Figure 5.3: Shielded Twisted Pair

Cables with a shield are called Shielded Twisted Pair and commonly abbreviated STP. Cables without
a shield are called Unshielded Twisted Pair or UTP. Twisting the wires together results in a
characteristic impedance for the cable. A typical impedance for UTP is 100 ohm for Ethernet 10BaseT
cable.

UTP or Unshielded Twisted Pair cable is used on Ethernet 10BaseT and can also be used with token
ring. It uses the RJ line of connectors (RJ45, RJ11, etc.) STP or Shielded Twisted Pair is used with
the traditional token ring cabling or ICSIBM cabling system. It requires a custom connector. IBM STP
(Shielded Twisted Pair) has a characteristic impedance of 150 ohms.

iii. Coaxial Cable. Coaxial cable was invented in 1929 and first used commercially in 1941. AT and T
established its first crosscontinental coaxial transmission system in 1940. Depending on the carrier
technology used and other factors, twisted pair copper wire and optical fiber are alternatives to coaxial
cable.

Coaxial cable is the kind of copper cable used by cable TV companies between the community
antenna and user homes and businesses. Coaxial cable is sometimes used by telephone companies
from their central office to the telephone poles near users. It is also widely installed for use in business
and corporation Ethernet and other types of local area network. Coaxial cable is called "coaxial"
because it includes one physical channel that carries the signal surrounded (after a layer of insulation)
by another concentric physical channel, both running along the same axis. The outer channel serves
as a ground. Many of these cables or pairs of coaxial tubes can be placed in a single outer sheathing
and, with repeaters, can carry information for a great distance.

Coaxial cable consists of 2 conductors. The inner conductor is held inside an insulator with the other
conductor woven around it providing a shield. An insulating protective coating called a jacket covers
the outer conductor.

Figure 5.4: Coax Cable

The outer shield protects the inner conductor from outside electrical signals. The distance between the outer
conductor (shield) and inner conductor plus the type of material used for insulating the inner conductor determine the
cable properties or impedance. Typical impedances for coaxial cables are 75 ohms for cable TV, 50 ohms for
Ethernet thinnet and thicknet. The excellent control of the impedance characteristics of the cable allow higher data
rates to be transferred than twisted pair cable.

Baseband Coaxial Cable
The medium consists of a copper core surrounded by insulating material and a braided outer conductor.

The term baseband indicate it is for digital transmission.

Physical connection consists of a metal pin touching the copper core. There are two possible ways.

— Use of a T connector. This is often used with thin Ethernet cable.

— Use of a vampire tap in which a metal pin is inserted into the core. A clipertype of
device is used to hold the connection. This is often used with thick Ethernet cable.
Data rates of a few 10 s of Mbps to a couple of Gbps are possible.

Broadband Coaxial Cable
A similar type of cable, but the bandwidth is wider than 4 kHz. (In computer networking world, any network using
analog transmission is considered as broadband).

Typical bandwidth of 300 MHz.

Operate at distances up to 100 km.

Use analog signal.

Often the traffic is oneway.

Requires amplifiers to boost signal strength.

Cable TV uses this technology, so the infrastructure is already there.

Two types of systems have emerged.

— Dual cable systems use two cables, one for each direction.

— Midsplit systems use one cable, but the frequency is divided such that one portion of the
frequency is used in one direction, the second is used for the other direction.

(iv) Optical Fiber. Optical fiber consists of thin glass fibers that can carry information at frequencies in
the visible light spectrum and beyond. The typical optical fiber consists of a very narrow strand of
glass called the core. Around the core is a concentric layer of glass called the cladding. A typical core
diameter is 62.5 microns (1 micron = 10−6 meters). Typically cladding has a diameter of 125 microns.
Coating the cladding is a protective coating consisting of plastic, it is called the jacket.

An important characteristic of fiber optics is refraction. Refraction is the characteristic of a material to either pass or
reflect light. When light passes through a medium, it "bends" as it passes from one medium to the other. An
example of this is when we look into a pond of water.

Figure 5.5: Optical Fiber

Figure 5.6: Characteristic of Fiber Optics

If the angle of incidence is small, the light rays are reflected and do not pass into the water. If the angle of incident
is great, light passes through the media but is bent or refracted.

Figure 5.7: Angle of Incidence

Optical fibers work on the principle that the core refracts the light and the cladding reflects the light. The core
refracts the light and guides the light along its path. The cladding reflects any light back into the core and stops light
from escaping through it.

In fiber optic technology, lights are used to transmit signals instead of electronics. At a fixed rate, the presence of
the light means 1, the absence 0, for example.

Three components are required.

1. Light source: Typically a Light Emitting Diode (LED), or laser diode.

2. Fiber medium: Current technology carries light pulses for tremendous distance (e.g., 100s of
kilometers) with virtually no signal loss (i.e., no amplifiers or repeaters are needed).

3. Detector: Which detects the light and converts it to electric signals.

Advantages:

1. High data rate, Gbps.
2. Low error rate, less than

3. Difficult to tap onto the network, so better security. In wire connection, it is difficult to know if
somebody has a tap on it. In fiber, tapping is much more difficult.

4. Much thinner in dimension and lighter in weight.
5. Not susceptible to electrical interference or corrosion.

Disadvantages:

1. Difficult to tap (to add new computer to network, or to connect to a different network)

2. One way signal only. Two fibers needed to get full duplex communication.

Applications of Fiber Optics
Longhaul trunks—backbone network, telephone network

Metropolitan trunks—without repeaters (average 8 miles in length)

Rural exchanges trunks—link towns and villages

Local loops—direct from central exchange to a subscriber

Local area network—100 Mbps ring networks.

Our campus (UTPA) has a fiber optic backbone network.

A popular fiber optics network is SONET (Synchronous Optical NETwork) which was developed in the U.S. by
Bellcore around 1985. Later CCITT recommended its standard in 1989, called SDH (Synchronous Digital Hierarchy).

Optical Transmission Modes
There are three primary types of transmission modes using optical fiber. They are:

1. Step index

2. Graded index
3. Single mode

1. Step Index Mode. Step index has a large core, so the light rays tend to bounce around inside the core,
reflecting off the cladding. This causes some rays to take a longer or shorter path through the core.
Some take the direct path with hardly any reflections while others bounce back and forth taking a
longer path. The result is that the light rays arrive at the receiver at different times. The signal
becomes longer than the original signal. LED light sources are used. Typical core : 62.5 microns.

Figure 5.8: Step Index Mode

2. Graded Index Mode. Graded index has a gradual change in the core's refractive index. This causes the
light rays to be gradually bent back into the core path. This is represented by a curved reflective path
in the attached drawing. The result is a better receive signal than with step index. LED light sources
are used. Typical core: 62.5 microns.

Both step index and graded index allow more than one light source to be used (different colors
simultaneously), so multiple channels of data can be run at the same time!
3. Single Mode. Single mode has separate distinct refractive indexes for the cladding and core. The light
ray passes through the core with relatively few reflections off the cladding. Single mode is used for a
single source of light (one color) operation. It requires a laser and the core is very small : 9 microns.

Figure 5.9: Cladding

Comparison of Optical Fibers
We don't use frequency to talk about speed any more, we use wavelengths instead. The wavelength of light sources
is measured in nanometers or 1 billionth of a meter.
Indoor cable specifications:

LED (Light Emitting Diode) light source

3.5 dB/Km attenuation (loses 3.5 dB of signal per kilometer)
850 nM—wavelength of light source

Typically 62.5/125 (core dia/cladding dia)

Multimode—can run many light sources.

Outdoor cable specifications:
Laser light source

1 dB/Km attenuation (loses 1 dB of signal per kilometer)

1170 nM—wavelength of light source
Monomode (single mode)

Advantages of Optical Fiber
Noise immunity: RFI and EMI immune (RFI—Radio Frequency Interference, EMI—Electromagnetic Interference)
Security: Cannot tap into cable.

Large capacity due to BW (bandwidth)

No corrosion

Longer distances than copper wire

Smaller and lighter than copper wire

Faster transmission rate
Figure 5.10: Comparison of Optical Fibers

Disadvantages of Optical Fiber
Physical vibration will show up as signal noise

Limited physical arc of cable. Bend it too much and it will break

Difficult to splice

The cost of optical fiber is a tradeoff between capacity and cost. At higher transmission capacity, it is cheaper than
copper. At lower transmission capacity, it is more expensive.

Media versus Bandwidth
The following table 5.1 compares the usable bandwidth of the different guided transmission media.

Table 5.1: Bandwidths of
Transmission Media
Open table as spreadsheet
Cable Type Bandwidth
Open cable 0 – 5 MHz
Twisted pair 0 – 100 MHz
Coaxial cable 0 – 600 MHz
Optical fiber 0 – 1 GHz
HISTORY OF FIBER OPTIC TECHNOLOGY
People have used light to transmit information for hundreds of years. However, it was not until the 1960s, with the
invention of the laser, that widespread interest in optical (light) systems for data communications began. The invention of
the laser prompted researchers to study the potential of fiber optics for data communications, sensing, and other
applications. Laser systems could send a much larger amount of data than telephone, microwave, and other electrical
systems. The first experiment with the laser involved letting the laser beam transmit freely through the air. Researchers
also conducted experiments letting the laser beam transmit through different types of waveguides. Glass fibers, gas
filled pipes, and tubes with focusing lenses are examples of optical waveguides. Glass fibers soon became the preferred
medium for fiber optic research.

Initially, the very large losses in the optical fibers prevented coaxial cables from being replaced. Loss is the decrease in
the amount of light reaching the end of the fiber. Early fibers had losses around 1,000 dB/km making them impractical for
communications use. In 1969, several scientists concluded that impurities in the fiber material caused the signal loss in
optical fibers. The basic fiber material did not prevent the light signal from reaching the end of the fiber. These
researchers believed it was possible to reduce the losses in optical fibers by removing the impurities. By removing the
impurities, construction of lowloss optical fibers was possible.
There are two basic types of optical fibers, multimode fibers and single mode fibers. In 1970, corning glass works made
a multimode fiber with losses under 20 dB/km.

This same company, in 1972, made a high silicacore multimode optical fiber with 4 dB/km minimum attenuation (loss).
Currently, multimode fibers can have losses as low as 0.5 dB/km at wavelengths around 1300 nm. Single mode fibers
are available with losses lower than 0.25 dB/km at wavelengths around 1500 nm.

Advantages and Disadvantages of Fiber Optics
Fiber optic systems have many attractive features that are superior to electrical systems. These include improved
system performance, immunity to electrical noise, signal security, and improved safety and electrical isolation.

Other advantages include reduced size and weight, environmental protection, and overall system economy. Here, details
are given of the main advantages of fiber optic systems.

Advantages of Fiber Optics
System performance

Greatly increased bandwidth and capacity

Lower signal attenuation (loss)

Immunity to electrical noise

Immune to noise (electromagnetic interference [EMI] and radiofrequency interference [RFI]).

No crosstalk

Lower bit error rates

Signal security

Difficult to tap
Nonconductive (does not radiate signals) electrical isolation

No common ground required

Freedom from short circuit and sparks

Size and weight

Reduced size and weight cables

Environmental protection

Resistant to radiation and corrosion

Resistant to temperature variations

Improved ruggedness and flexibility

Less restrictive in harsh environments
Overall system economy

Low perchannel cost

Lower installation cost
PARTS OF A FIBER OPTIC LINK
Fiber optic transmission uses the same basic elements as copperbased transmission systems: A transmitter, a
receiver, and a medium by which the signal is passed from one to the other, in this case, optical fiber. Figure
5.11 illustrates these elements.

Figure 5.11: Elements of a Fiber Optic Link

The transmitter uses an electrical interface to encode the use information through AM, FM or digital modulation. A laser
diode or an LED do the encoding to allow an optical output of 850 nm, 1310 nm, or 1550 nm (typically).

The optical fiber connects the transmitter and the receiver. This fiber may be either singlemode or multimode. The fiber
consists of three main regions, as illustrated in (figure 5.12). The core, the center of the fiber, actually carries the light.
The cladding surround the core in a glass with a different refractive index than the core, allowing the light to be confined
in the fiber core. A coating or buffer, typically plastic, provides strength and protection to the optical fiber.

Figure 5.12: Crosssection of an Optical Fiber

The receiver uses either a PIN photodiode or an ADP to receive the optical signal and convert it back into an electrical
signal. A data demodulator converts the data back into its original electrical form. These elements comprise the simplest
link, but other elements may also appear in a fiber optic transmission system.

For example, the addition of WDM components allows two separate signals to be joined into a composite signal for
transmission, and then can be separated into their original signals at the receive end. Other wavelengthdivision
multiplexing techniques allow up to eight signals (CWDM) or more (DWDM) to be combined onto a single fiber.

Long distance fiber optic transmission leads to further system complexity. Many longhaul transmission systems require
signal regenerators, signal repeaters, or optical amplifier such as EDFAs in order to maintain signal quality. System
drop/repeat/add requirements, such as those in multichannel broadcast networks, further add to the fiber optic system,
incorporating adddrop multiplexers, couplers/splitters, signal fanouts, dispersion management equipment, remote
monitoring interfaces, and errorcorrection components.
FIBER OPTIC NETWORK TOPOLOGIES FOR ITS AND OTHER SYSTEMS
All networks involve the same basic principle: information can be sent to, shared with, passed on, or bypassed within a
number of computer stations (nodes) and a master computer (server). Network applications include LANs, MANs, WANs,
SANs intrabuilding and interbuilding communications, broadcast distribution, Intelligent Transportation Systems (ITS),
telecommunications, Supervisory Control and Data Acquisition (SCADA) networks, etc. In addition to its advantages
(i.e., bandwidth, durability, ease of installation, immunity to EMI/RFI and harsh environmental conditions, longterm
economies, etc.), optical fiber better accommodates today's increasingly complex network architectures than copper
alternatives. Figure 5.13 illustrates the interconnection between these types of networks.

Figure 5.13: Interconnections between Different Network Types

Networks can be configured in a number of topologies. These include a bus, with or without a backbone, a star network,
a ring network, which can be redundant and/or selfhealing, or some combination of these. Each topology has its
strengths and weaknesses, and some network types work better for one application while another application would use
a different network type. Local, metropolitan, or wide area networks generally use a combination, or "mesh" topology.
BUS NETWORK
A bus network topology, also called a daisychain topology has each computer directly connected on a main
communication line. One end has a controller, and the other end has a terminator. Any computer that wants to talk to the
main computer must wait its turn for access to the transmission line. In a straight network topology, only one computer
can communicate at a time. When a computer uses the network, the information is sent to the controller, which then
sends the information down the line of computers until it reaches the terminating computer. Each computer in the line
receives the same information. Figure 5.14 illustrates a bus network topology.

Figure 5.14: Bus Network Topology

A bus network with a backbone operates in the same fashion, but each computer has an individual connection to the
network. A bus network with a backbone offers greater reliability than a simple bus topology. In a simple bus, if one
computer in the network goes down, the network is broken. A backbone adds reliability in that the loss of one computer
does not disrupt the entire network. Figure 5.15 illustrates this topology with a backbone.

Figure 5.15: Bus Network with Backbone
STAR NETWORKS
Star networks incorporate multiport star couplers in to achieve the topology. Once again, a main controlling computer or
computer server interconnects with all the other computers in the network. As with the bus topology with a backbone, the
failure of one computer node does not cause a failure in the network. Figure 5.16 illustrates a star network topology.

Figure 5.16: Star Network Topology

Both the bus and the star network topologies use a central computer that controls the system inputs and outputs. Also
called a server, this computer has external connections, to the Internet for example, as well as connections to the
computer nodes in the network.
RING NETWORKS
Ring networks operate like bus networks with the exception of a terminating computer. In this configuration, the
computers in the ring link to a main communication cable. The network receives information via a "token" containing
information requested by one or more computers on the network. The token passes around the ring until the requesting
computer(s) have received the data. The token uses a packet of information that serves as an address for the computer
that requested the information. The computer then "empties" the token, which continues to travel the ring until another
computer requests information to be put into the token.

An advanced version of the ring network uses two communication cables sending information in both directions. Known
as a counterrotating ring, this creates a fault tolerant network that will redirect transmission in the other direction, should
a node on the network detect a disruption. This network uses fiber optic transceiver, one controlling unit in set in
"master" mode along with several nodes that have been set as "remote" units. The first remote data transceiver receives
the transmission from the master unit and retransmits it to the next remote unit as well as transmitting it back to the
master unit. An interruption in the signal line on the first ring is bypassed via the second ring, allowing the network to
maintain integrity.

Figure 5.17: Token Ring Network Topology

Figure 5.18: Selfhealing Ring Topology

2 Unguided Media
Unguided transmission media is data signals that flow through the air. They are not guided or bound to a channel to
follow. They are classified by the type of wave propagation as given below:

Radiowaves (30 MHz–1 GHz)

Like radio and TV broadcast

Can penetrate buildings
Travel long distances

Omnidirectional

Microwave (2–40 GHz)

Higher frequency than radiowaves

Unidirectional–can be aimed

Long used for long–distance telephone network

Do not bend with the curvature of the earth, so tall towers and repeaters are needed (lineof
sight)

Do not pass through buildings well

Used for longhaul telephone transmission.
Infrared

Higher frequency than microwave

Directional

Do not pass through solid objects

Used for small signal–room network or indoor wireless LANs.

Radiowaves Propagation

There are three types of RF (Radio frequency) propagation:

Ground Wave

Ionospheric

Line of Sight (LOS)

Ground wave propagation follows the curvature of the earth. Ground waves have carrier frequencies upto 2 MHz. AM
radio is an example of ground wave propagation. Radio frequencies are in the range of 300 kHz to 10 GHz. We are
seeing an emerging technology called wireless LANs. Some use radio frequencies to connect the workstations together,
some use infrared technologies.

Microwave

Microwave transmission is line of sight transmission. The transmitting station must be in visible contact
with the receiving station. This sets a limit on the distance between stations depending on the local
geography. Typically the line of sight due to the earth's curvature is only 50 km to the horizon. Repeater
stations must be placed so that data signal can hop, skip and jump across the country.

Mircowaves operate of high operating frequencies of 3 to 10 GHz. This allows them to carry large quantities
of data due to their large bandwidth.

Infrared Transmission

Infrared communication is now common as a means of wireless communication between devices. It will not
penetrate buildings and therefore is secure.

Used for very short line of sight transmission, remote car locking systems, wireless security alarms,
remote TV channel changes etc.

It utilises a wide frequency band, 2 to 40 GHz but is susceptible to attenuation and interferences.
Attenuation can rise markedly in poor atmospheric conditions, e.g. rain but adversely affects the higher end
of the frequency band which is only used for short distance transmission. Natural noise severally affects
transmission frequencies below 2 GHz.
REVIEW QUESTIONS
1. Name the two major categories of transmission media.
2. How do guided media differ from unguided media?
3. What are the three major classes of guided media?
4. What is the form of the signal in twistedpair cable and coaxial cable? How does this differ from the signal in fiber
optic cable?
5. Give a use for each class of guided media.
6. What is the major advantage of shielded twistedpair over unshielded twistedpair cable?
7. What is the significance of the twisting in twistedpair cable?
8. Why is coaxial cable superior to twistedpair cable?
9. What are the disadvantages of optical fiber as a transmission medium?
10. Write short history of fiber optic technology.
11. What are the different topologies?
12. Explain the various unguided media used in data communication.
Chapter 6: Telecommunication

INTRODUCTION
Telecommunication is the technique of transmitting a message, from one point or place to another with the typical
additional attribute of being bidirectional. In practice it also recognizes that something may be lost in the process; hence
the term ‘telecommunication’ covers all forms of distance communication, including radio, telegraphy, television,
telephony data communication and computer networking.
The elements of a telecommunication system are a transmitter, a medium (line) and possibly a channel imposed upon
the medium, and receiver. The transmitter is a device that transforms or encodes the message into a physical
phenomenon; the signal. The transmission medium, by its physical nature, is likely to modify or degrade the signal on its
path from the transmitter to the receiver. The receiver has a decoding mechanism capable of recovering the message
within certain limits of signal degradation. In some cases, the final "receiver" is the human eye and/or ear (or in some
extreme cases other sense organs) and the recovery of the message is done by the brain.

Telecommunication can be pointtopoint, pointtomultipoint or broadcasting, which is a particular form of pointto
multipoint that goes only from the transmitter to the receivers. The art of the telecommunications engineer is to analyse
the physical properties of the line or transmission media, and the statistical properties of the message in order to design
the most effective encoding and decoding mechanisms.

When systems are designed to communicate through human sense organs (mainly vision and hearing), physiological and
psychological characteristics of human perception will be taken into account. This has important economic implications
and engineers will research what defects may be tolerated in the signal yet not affect the viewing or hearing experience
too badly.
EXAMPLES OF HUMAN (TELE) COMMUNICATIONS
In a simplistic example, take a normal conversation between you and a friend. The message is the sentence, your mind
decides to communicate to your friend. The transmitter is the language areas in your brain, the motor cortex, your vocal
cords, the larynx, and your mouth that produce those sounds called speech. The signal is the sound waves that can be
identified as speech. The channel is the air carrying those sound waves, and all the acoustic properties of the space you
are in: echoes, ambient noise, reverberation. Between you and your friend (the receiver), may be other technologies that
do or do not introduce their own distortions of the original vocal signal (e.g., telephone, HAM radio, IP phone, etc.) The
penultimate receiver is your friend's ear, the auditory nerve, the language areas in your friend's brain that will make the
difference between your voice and the sound of a car passing by, and decode your speech into, hopefully, the same
sentence.

The car passing by is an example of an important property of the channel called noise. Another important aspect of the
channel is called the bandwidth, and you would become very aware of the effects of a limited bandwidth if you were now
talking to your friend on a telephone or a walkietalkie.
TELEPHONE NETWORK
The telephone system has been developed over many years and has gone through many incremental evolutionary steps.
The traditional service of the telephone network has been for voice communication and only recently the network has
been used to support the high amount of data transfer we are currently experiencing. For cost effective voice
communications it has been identified that the humans can communicate at frequencies between 300 Hz to 3500 Hz.
Though we can hear and speak at higher and lower frequencies, voice communications between 300 Hz and 3500 Hz are
clear and efficient for the telephone network to transmit and receive. A voice channel goes from 0 Hz to 4000 Hz and
was developed to avoid any overlapping to any other adjacent voice channels.

Figure 6.1: Telephone Voice Channel

Through the telephone network development it has been discovered that it is more efficient to transmit a voice channel in
a digital form. In a digital form the voice channel can be routed to its destination with very low noise, higher reliability and
more cost effectiveness. To do this the voice channel is converted to digital by an Analog to Digital Converter (A/D
converter or ADC) at the central office. The Central Office (CO) is the location in your area that the 2Wires that come
from your house called tip and ring are terminated. Once the voice channel has been digitized it is transmitted over the
network to the CO of the number you called. At this remote CO the voice channel is reconstructed back into an analog
form so that remote person can understand it. This transformation back to analog is done by a digital to analog converter
also know as a D/A converter or DAC.
One important point to know is that the A/D converter samples the voice channel at twice the frequency of the voice
channel; that is at 8 kHz. The reason the sampling is done at twice the original signals frequency is due to a law called
the Nyquist rate, which states to digitize a waveform and have enough information to reconvert it back to the an analog
waveform one must sample the original waveform at no less than twice the frequency of the original waveform. Also
each sample is identified by 8 unique bits that represent 256 different states. 8 bits sampled 8000 times a second is
equal to 64 kbps.

Figure 6.2: The Traditional Communication Process
PUBLIC SWITCHED TELEPHONE NETWORK
The Public Switched Telephone Network (PSTN) is the concatenation of the world's public circuit switched telephone
networks, in much the same way that the internet is the concatenation of the world's public IPbased packetswitched
networks. Originally a network of fixedline analog telephone systems, the PSTN is now almost entirely digital, and now
includes mobile as well as fixed telephones.

Homeworking and very small business have traditionally been supported by modem connection to the PSTN (Telephone
System), received by another modem at the other end. Requires digital signals to be converted to analogue and then
back to the digital at the receiver.
Today's modems can handle a maximum transmission rate of 56 Kbps using complex compression techniques but
upload transmission rates can be much less. The actual amount of information transferred is much less than the
theoretical maximum.

The PSTN is largely governed by technical standards created by the ITUT, and uses E.163/E.164 addresses (known
more commonly as telephone numbers) for addressing.

The basic digital circuit in the PSTN is a 64kilobitpersecond channel, originally designed by Bell labs, called a "DS0" or
digital signal 0. To carry a typical phone call from a calling party to a called party, the audio sound is digitized at an 8
kHz sample rate using 8bit pulse code modulation.

The DS0's are the basic granularity at which switching takes place in a telephone exchange. DS0's are also known as
timeslots because they are multiplexed together in a timedivision fashion. Multiple DS0's are multiplexed together on
higher capacity circuits, such that 24 DS0's make a DS1 signal, which when carried on copper is the wellknown, T
carrier system, T1 (the European equivalent is an E1, containing 32–64 kbit/s channels). In modern networks, this
multiplexing is moved as close to the end user as possible, usually into cabinets at the roadside in residential areas, or
into large business premises.

The timeslots are conveyed from the initial multiplexer to the exchange over a set of equipment collectively known as
the access network. The access network and interexchange transport of the PSTN use Synchronous Optical
Transmission (SONET and SDH) technology, although some parts still use the older PDH technology.

Within the access network, there are a number of reference points defined. Most of these are of interest mainly to ISDN
but one—the V reference point—is of more general interest. This is the reference point between a primary multiplexer
and an exchange. The protocols at this reference point were standardised in ETSI areas as the V5 interface.

Only the very oldest and most backward parts of the telephone network still use analog technology for anything other
than the last mile loop to the end user, and in recent years digital services have been increasing rolled out to end users
using services such as DSL and ISDN.

Advantages

There are following advantages of PSTN.

Cheap equipment

Uses ordinary telephone connection

Local call charges for internet connection.

Disadvantages

There are following disadvantages of PSTN.

Low bandwidth for multimedia

Ties up the telephone line

Need an ISP to connect to the Internet.
ARCHITECTURE AND CONTEXT
In the 1970s the telecommunications industry conceived that digital services would follow much the same pattern as
voice services, and conceived a vision of endtoend circuit switched services, known as the Broadband Integrated
Services Digital Network (BISDN). The BISDN vision has been overtaken by the disruptive technology of the Internet.

Many observers believe that the long term future of the PSTN is to be just one application of the Internet—however, the
Internet has some way to go before this transition can be made. The quality of service guarantee is one aspect that
needs to be improved on in the voice over IP technology.
There are a number of large private telephone networks which are not linked to the PSTN, usually for military purposes.
There are also private networks run by large companies which are linked to the PSTN only through limited gateways, like
a large PABX system.
IMPORTANCE OF NETWORK IN TELEPHONES
Telephones and networking work hand in hand. The telecommunications industry has been gradually integrating with the
computer industry and the computer industry has been gradually integrating with the telecommunications industry. The
common goal is to join distantly located local area networks into Metropolitan and Wide Area Networks (MANs and
WANs).
VOICE CHANNELS
When we think of telecommunications, the first thing that comes to mind is telephone systems and the phone at
home. Talking to someone on the phone uses voice channels.

We use voice channels for modem communications to connect to BBSs (Bulletin Board Services) or to connect to
the Internet. We also use voice channels to connect LANs using remote access. Due to the bandwidth limits on
voice channels, the data transfer rate is relatively slow.
Typical voice channel communication rates are: 300, 1200, 2400, 9600, 14.4 k, 19.2 k, 28.8 k, 33.6 k and 56 kbps
(bits per second).

Voice Channel Constraints

Due to the limited Bandwidth (BW) of the voice channel (0–4 kHz), we are limited to the amount of data that we can
pass through the voice channel. The Nyquist Theorem addresses this limitation.
DATA CHANNELS
Data channels are dedicated lines for communicating digitized voice and data. At the end of 1996, there was a major
milestone when more data was communicated in North America's telecommunications system than voice.

Data channels are special communications channels provided by "common carriers" such as Telus, Sprint, Bell Canada,
AT and T, etc. for transferring digital data. Data channels are also called "leased lines". They are "directly" connected:
you don't have to dial a connection number. The connections are up and running 24 hours per day. They appear to work
as if there were a wire running directly between the source and the destination. Typical transfer rates for data channels
are: 56 k, 128 k, 1.544 M, 2.08 M, 45 M and 155 Mbps.
Common carriers charge for data connections by:

1. The amount of data transferred (megabytes per month)

2. The transfer rate (bits per second)
3. The amount of use (time per month)
ADSL—ASYMMETRICAL DIGITAL SUBSCRIBER LINE
Asymmetrical Digital Subscriber Line (ADSL) uses the existing analog local loop lines for digital data transfer (to and
from the home). It is asymmetrical because the upstream transfer rate is slower than the downstream data rate. This
means that the data transfer from the premise (home) to the CO is a different rate than the data transfer from the CO to
the home.

The data transfer rate is adaptive. This means that, depending on the condition of the local loop lines, ADSL will
automatically compensate (i.e., find the fastest transfer rate possible). The range for upstream data transfer is from 64
kbps to 768 kbps. The range for downstream data transfers is from 1.5 Mbps to 8 Mbps. The reasoning for the
asymmetrical transfer rate is that most users will be surfing the Internet; upstream requests tend to be small web page
addresses. The downstream data consists of downloads of large graphic intensive web pages. Small upstream requests,
larger downstream response.

The data transfer rate depends on the distance from the central office, the quality of the line, and the wire gauge. If the
distance from the central office is 15,000 to 18,000 ft, then the maximum transfer rate is 1.5 Mbps. If the distance is
9,000 ft or less, the maximum transfer rate is 8 Mbps.

ADSL Standards
At the time of this writing, there are 3 competing standards for ADSL (shown below):

1. Carrierless phase modulation ADSL

2. Splitterless ADSL

3. Discrete multitone ADSL

Carrierless Phase Modulation (CAP) ADSL is a modulation technique similar to quadrature amplitude modulation. It
provides echo cancellation, and overlaps upstream and downstream signals.

Splitterless ADSL (also called ADSL Lite, G. Lite, PnP ADSL, Universal ADSL) has a lower transmitting rate, and is
easier to implement.

DMT—Discrete multitone is an ANSI T1.413 standard that uses a broadband modem (covering the 4 kHz to 2.2 MHz
range). It has 256 channels of 4 kHz: each channel is assigned 15 bits of data to transfer. In addition, each channel is
checked for signal quality, and bits are assigned accordingly. A poor responding channel may have less bits assigned, or
none at all. DMT adjusts for the local loop line conditions, and attempts to make the fastest transfer rate possible.

ADSL OSI Model
ADSL is a physical layer protocol that covers the transmission of data and cabling requirements.

Figure 6.3: ADSL Premise Equipment

ADSL shares the bandwidth of the local loop with the existing phone system. It does not require modification to the
central office switch. Instead, a splitter combines the ADSL information with the POTS switch's analog information. At
the central office end, the ADSL signal is sent to the Digital Subscriber Line Access Module (DSLAM), and then to a
communication server.

At the premise end, another splitter separates the ADSL information from the analog information. An ADSL modem,
called an ATUR device, decodes the ADSL information, and sends it to the Service Module (SM). The service module
translates it to Ethernet. In plain network terms, ADSL comes in and an Ethernet signal goes out for connection to a
network interface card.

ADSL Advantages:

1. No expensive modification is required to CO switch.

2. Simple splitter splits ADSL signal from the existing analog line.

3. High bandwidth is available.
4. The POTS works regardless of ADS.

5. ADSL has competitive pricing versus other technologies.

ADSL Disadvantages:

1. The transfer rate depends on distance from the central office.

2. The presence of bridged taps and load coils on the local loop affect the transfer rate.

3. ADSL must be installed to test if it will work.

4. 25% of existing local loops will not work with ADSL.

5. There is an 18,000 ft distance limit from the central office.

6. There can be a bottleneck at the communication server (at the central office).
DIGITAL SWITCH
In U.S. and military telecommunication, a digital switch is a switch that performs timedivisionmultiplexed switching of
digitized signals. All switches built since the 1970s are digital, so for practical purposes this is a distinction without a
difference.

Telephone exchange for the building
Telephone switch for the switching equipment

Concentrator for the concentrator, whether or not it is colocated with the switch

Automatic telephone exchanges came into existence in the early 1900s. They were designed to replace the need for
human telephone operators. Before the exchanges became automated, operators had to complete the connections
required for a telephone call. Almost everywhere, operators have been replaced by computerized exchanges.

The local exchange automatically senses an off hook (tip) telephone condition, provides dial tone to that phone, receives
the pulses or DTMF tones generated by the phone, and then completes a connection to the called phone within the same
exchange or to another distant exchange.
The exchange then maintains the connection until a party hangs up, and the connection is disconnected. Additional
features, such as billing equipment, may also be incorporated into the exchange. Early exchanges used motors, shaft
drives, rotating switches and relays. Some types of automatic exchanges were stronger, all relay, XY, panel and
crossbar.

A telephone switch is the brains of an exchange. It is a device for routing calls from one telephone to another, generally
as part of the public switched telephone network. They work by connecting two or more digital virtual circuits together,
according to a dialed telephone number. Calls are setup between switches using the ISUP protocol, or one of its
variants.

Digital switches encode the speech going on, in extremely minute time slices—many per second. At each time slice, a
digital representation of the tone is made. The digits are then sent to the receiving end of the line, where the reverse
process occurs, to produce the sound for the receiving phone. In other words, when you use a telephone, you are
generally having your voice "encoded" and then reconstructed for the person on the other end. Your voice is very slightly
delayed in the process (probably by only a small fraction of one second)— it is not "live", it is reconstructed—delayed
only minutely.

Individual local loop telephone lines are connected to a remote concentrator. In many cases, the concentrator is co
located in the same building as the switch. The interface between concentrators and telephone switches has been
standardised by ETSI as the V5 protocol.
Some telephone switches do not have concentrators directly connected to them, but rather are used to connect calls
between other telephone switches. Usually a complex machine (or series of them) in a central exchange building, these
are referred to as "carrierlevel" switches or tandems.

Some telephone exchange buildings in small towns now house only remote switches, and are homed "parent" switch,
usually several kilometres away. The remote switch is dependent on the parent switch for routing and number plan
information. Unlike a digital loop carrier, a remote switch can route calls between local phones itself, without using trunks
to the parent switch.

Telephone switches are usually owned and operated by a telephone service provider or "carrier" and located in their
premises, but sometimes individual businesses or private commercial buildings will house their own switch, called a
PBX, or Private Branch Exchange.

The Switch's Place in the System
Telephone switches are a small part of a large network. The majority of work and expense of the phone system is the
wiring outside the central office, or the "outside plant".
Some companies use "pair gain" devices to provide telephone service to subscribers. These devices are used to provide
service where existing copper facilities have been exhausted or by sitting in a neighborhood, can reduce the length of
copper pairs, enabling digital services such as ISDN or DSL. Pair gain or Digital Loop Carriers (DLCs) are located outside
the central office, usually in a large neighborhood distance from the CO.

DLCs are often referred to as Subscriber Loop Carriers (SLCs). Early SLC systems (SLC1) used an analog carrier for
transport between the remote site and the central office. Later systems (SLC96, SLC5) and other vendors' DLC
products contain line cards that convert the analog signal to a digital signal (usually PCM). This digital signal can then be
transported over copper, fiber, or other transport medium to the central office. Other components include ringing
generators to provide ringing current and battery backups.

DLCs can be configured as universal (UDLCs) or integrated (IDLCS). Universal DLCs have two terminals, a Central
Office Terminal (COT) and a Remote Terminal (RT), that function similarly. Both terminals interface with analog signals,
convert to digital signals, and transport to the other side where the reverse is performed. Sometimes, the transport is
handled by separate equipment. In an Integrated DLC, the COT is eliminated. Instead, the RT is connected digitally to
equipment in the telephone switch. This reduces the total amount of equipment required. Switches are used in both local
central offices and in long distance centers.

Switch Design
Long distance switches may use a slower, more efficient switchallocation algorithm than central offices, because they
have near 100% utilization of their input and output channels. Central offices have more than 90% of their channel
capacity unused.

While traditionally, telephone switches connected physical circuits (e.g., wire pairs), modern telephone switches use a
combination of space—and timedivision switching. In other words, each voice channel is represented by a time slot (say
1 or 2) on a physical wire pair (A or B). In order to connect two voice channels (say A1 and B2) together, the telephone
switch interchanges the information between A1 and B2. It switches both the time slot and physical connection. To do
this, it exchanges data between the time slots and connections 8000 times per second, under control of digital logic that
cycles through electronic lists of the current connections. Using both types of switching makes a modern switch far
smaller than either a space or time switch could be by itself.

The structure of a switch is an odd number of layers of smaller, simpler subswitches, interconnected by a web of wires
that goes from each subswitch, to a set of the next layer of subswitches. In most designs, a physical (space) switching
layer will alternate with a time switching layer. The layers are symmetric, because every call is symmetric (there's a
connection in both directions). A spacedivision subswitch uses digital multiplexers controlled by a cyclic memory. This
takes physical space for the wiring.

A timedivision subswitch reads a complete cycle of time slots into a memory, and then writes it out in a different order,
also under control of a cyclic computer memory. This causes some delay in the signal.

Switch Control Algorithms
The scarce resources in a telephone switch are the connections between layers of subswitches. The control logic has to
allocate these connections. The connections consist of both time slots and wires. The first thing to try is to search for a
subswitch that contains the needed in and out connections. There are two design paths to go if this simple search fails.

Fullyconnected mesh network is one way to have enough switching fabric to assure that the pairwise allocation will
always succeed by building a fullyconnected mesh network. This is the method usually used in central office switches,
which have low utilization of their resources.

Topological Sort
Another way is to have a minimal switching fabric that still can theoretically make all the connections, and reorganize the
switch's connections when a new connection won't fit.

If a subswitch with the needed pair of connections can't be found, a pair of subswitches will still have the necessary in
and out, because there has to be at least the same number of connections between each layer of the switch, or else the
switch will not be able to complete a full set of connections.

The pair of subswitches' connections can be reorganized with a clever algorithm called a topological sort, so that all the
existing connections continue, though they might migrate between the two different subswitches. This is the method
usually used in long distance switches, which have high utilization of their switching fabric.

A topological sort picks two subswitches. One has a needed input connection. The other has a needed output
connection. The connections of both subswitches are placed in a list that also includes the desired new connection.

In the list, the basic trick is to trace connections. Starting from some input or output, the computer traces a connection
to an output, then traces the other connection at that output to an input, and so forth, until it comes to an end. Each time
it traces from input to output, the connection is placed in one subswitch, and removed from the list. When it traces from
output to input, the connection is placed in the other subswitch and removed from the list. To complete correctly, tracing
must begin with single connection inputs and outputs, and only then trace doubleended inputs and outputs, which might
form loops.

Fault Tolerance
Composite switches are inherently faulttolerant. If a subswitch fails, the controlling computer can sense it during a
periodic test. The computer marks all the connections to the subswitch as "in use". This prevents new calls, and does
not interrupt old calls that remain working. As calls are ended, the subswitch then becomes unused. Some time later, a
technician can replace the circuit board. The next test succeeds, the connections to the repaired subswitch are marked
"not in use", and the switch returns to full operation.
To prevent frustration with unsensed failures, all the connections between layers in the switch are allocated using firstin
firstout lists. That way, when a disgusted customer hangs up and redials, they will get a different set of connections and
subswitches. A lastinfirstout allocation of connections might cause a continuing string of very frustrating failures.
TELEPHONE STRUCTURE
The telephone network consists of your phone at home that is connected (by the local loop) to the central office. The
central office is in turn connected to a hierarchical phone network. Worldwide, there are over 300 million
(300,000,000) telephones—98% of them interconnected.

POTS—Plain Old Telephone Set
The POTS, or Plain Old Telephone Set, consists of these 5 sections:

1. Ringer unit

2. Hook switch
3. Dialer unit

4. Hybrid/Speech network

5. Hand set

Figure 6.4: Plain Old Telephone Set

The connection to the CO (Central Office) comprises only 2 wires: Tip and Ring. This connection is called the "local
loop."

The Tip is +ve and colored green. The ring is −ve and colored red. If you look at a phone jack in your house, you
will see that it is wired for 4 wires: Red, green, black and yellow. However, black and yellow are not normally used.

The black and yellow wires can be used for a second telephone line or they can be used for running a network
physical layer protocol called Phonenet (by Farralon). Phonenet uses the black and yellow for network
communications. It is for use with Appletalk, and is a replacement for Localtalk. It runs at the Localtalk speed of 230
Kbps, reasonable for small networks.

1 Ringer Unit
The ringer is a device that alerts you to an incoming call, it interprets the ringing voltage from the central office.
Originally, the ringer was a electromagnetic bell. Today, though, most ringers are electronic devices.

The central office sends the following:

i. 90 to 120 VAC ringing voltage

ii. Frequency of 20 Hz

iii. Cadence for North America is 2 sec ON or 4 sec Off

2 Hook Switch
The hook switch is activated by lifting the handset off of the cradle. The position of the hook switch determines
whether the telephone is waiting for a call, or is actively using the line. The offhook position informs the network of
a request for use. The onhook position releases the use of the network.

3 Dialer Unit
There are two types of dialer units: Rotary and touch tone. Rotary is the old "put your finger in the hole and spin"
type. The rotary dial operates by toggling the hook switch on and off.

Touch tone is the modern method where 2 frequencies per push button are sent. Touch tone is a trade name; the
correct name is DTMF (Dual Tone Multi Frequency).

4 Hybrid/Speech Network
The Hybrid/Speech Network performs these functions:

i. It converts the Tx/Rx 4 wires from the handset to the 2 wires for the local loop.
ii. It interfaces the signals from the dialer unit to the telephone line.

iii. It provides auto line compensation for line length to keep the volume constant.

5 Handset
The Handset contains transducers that convert mechanical energy into electrical energy. The microphone converts
speech into electrical energy while the diaphragm (or speaker) converts electrical signals into audible signals.
FUNCTIONS OF A TELEPHONE SET
There are following functions of a telephone set:

1. Request use of network from the CO (Central Office).

2. Inform you of the network status: Dialtone, ringing, busy, fast busy (Talk Mail).

3. Informs CO of desired number.

4. Informs you when a call is incoming (phone rings).
5. Releases use of network when call is complete (hangup).

6. Transmit speech on network and receives speech from distant caller.

7. Adjust power levels and compensates for line length.
LOCAL LOOPS
The local loop is the connection between the central office and the home or business. Two wires (1 pair) are run into
every home. The pair does not go directly to the central office. Instead, it goes to those big green boxes—that you see
on the street corners—called "Serving Area Interfaces" (SIA). Large multiconductor bundles of wires then go from there
to the central office.
CENTRAL OFFICE
The central office provides the following functions:

It supplies the battery voltage for the telephone system. The onhook voltage is 48 VDC +/− 2V. The off
hook voltage is −6.5 VDC.

It supplies the ringing generator − 90 to 120 VAC, 20 Hz, 2 sec on/4 sec off

It supplies the busy signal (480 + 620 Hz, 0.5 sec on/0.5 sec off), dial tone (350 + 440 Hz) and fast busy
(480 + 620 Hz, 0.2 sec on/ 0.3 sec off).

Figure 6.5: Central Office

It has the digital switching gear that determines if the number is an interoffice call (local) or an intraoffice call (Toll—long
distance).
HIERARCHICAL PHONE NETWORKS
The PSTN (Public Switch Telephone Network) is divided into a hierarchical network. Here are the 5 classes of switching
centers in North America:

Open table as spreadsheet
Class Abbreviation Symbol Examples
Center
1 Regional RC 2 in
center Canada:
West
ReginaEast
Montreal
2 Sectional SC Calgary
center serves
Alberta

3 Primary PC Edmonton
center

4 Toll center TC Drumheller

4b Toll point TP Rainbow
lake

5 Central office CO 284xxxx
(Local loop)

In the following example:

Figure 6.6: Hierarchical Phone Networks

The Hierarchical portion is seen as follows:

Trunk Long distance telephone cable
Tool Trunk Connects CO (Central Office) to TC (Toll Center)
Inter Tool Everything above TC (Toll Center) and TC to TC
Trunk
Inter Office Between CO (Central Office)
Trunk
Intra Office Call between 2 subscribers within the same CO
Trunk (284–7079 to 284–8181).
Open table as spreadsheet
CALL ROUTING
Call routing is determined by network engineering and physical location. When all lines are idle, the call routing selects
the preferred route. If the preferred route is busy, then the call is routed to the second choice. Because the second
choice is routed through one toll center, the charge for the call is greater than the preferred route. The third choice is
used when the second choice is busy. The third choice goes through 2 toll centers, and is the most expensive route.

A central office can have up to 10,000 subscribers (for example, 284–0000 to 284–9999). Most have 4,000 to 5,000
subscribers. The central office bases the loading requirements on roughly 10% of the phones that will be in use at any
one time. However, the use of Internet dialup access has drastically changed this statistic.
TELEPHONE LINE CHARACTERISTICS
Telephone lines are not perfect devices due to their analog nature. The quality of the telephone line determines the rate
that modulated data can be transferred. Good noisefree lines allow faster transfer rates (such as 14.4 kbps) while poor
quality lines require the data transfer rate to be stepped down to 9600 bps or less. Phone lines have several measurable
characteristics that determine the quality of the line.

Attenuation distortion
Propagation delay

Envelope delay distortion

Figure 6.7: Attenuation Distortion

Attenuation distortion is the change in amplitude of the transmitted signal over the voice band: it is the frequency
response curve of the voice band.
ATTENUATION VERSUS FREQUENCY
To measure attenuation distortion, the phone line has a test frequency. This frequency is transmitted from 0 – 4 kHz into
the line at a standard amplitude of 0 db. The loss of signal—or attenuation—is measured at the receiving end, and
compared to a standard reference frequency of 1004 Hz.

Decibel (db) is a relative unit of measure. It is a log unit and a +3 db gain will indicate an amplitude of 2x the reference. It
is a logarithmic ratio between input voltage and output voltage, calculated by the following formula:

The resulting information is graphed on an attenuation vs. frequency chart. Attenuation is a loss of signal amplitude (the
receive signal is a smaller amplitude than the transmitted signal). It is indicated by a positive db. It's also possible to
have a signal appear at the receiving end, with a larger amplitude than when it started (this is indicated by negative db).

Attenuation occurs because the signal has to pass through many pieces of electronic equipment and transmission
media. Some can amplify the signal (make it a larger amplitude) and some may attenuate the signal (make it smaller).

There are maximum and minimum acceptable limits for the attenuation distortion that is on phone lines. The basic
channel conditioning is as follows:

Open table as spreadsheet
Frequency Range Loss (db)
5002500 2 to +8
3003000 3 to +12

The above loss is a range of acceptable values for the frequency range. In the Basic channelling conditioning, it is
acceptable to have a loss in signal—in the frequency range of 5002500 Hz—of "8 db loss to −2 db loss" (referenced to
the amplitude at 1 kHz). Note that this is shown as −8 db and +2 db (see the graph).

A +3 db attenuation is equal to −3 db in signal amplitude and a +8 db attenuation equates to −8 db in signal amplitude.
PROPAGATION DELAY
Signals transmitted down a phone line will take a finite time to reach the end of the line. The delay from the time the
signal was transmitted to the time it was received is called propagation delay. If the propagation delay was the exact
same across the frequency range, then there would be no problem. This would imply that all frequencies from 300 to
3000 Hz have the same amount of delay in reaching their destination over the phone line. They would arrive at the
destination at the same time, but delayed by a small amount of propagation delay.

For example, this delay is heard when talking on long distance telephones. In this instance, we have to wait a little longer
before we speak (to ensure that the other person hasn't already started to talk). Actually, all phone lines have
propagation delay.

Figure 6.8: Propagation Delay
If the propagation delay is long enough, the modem or communications package may timeout and close the connection.
In other words, it may think that the receive end has shut off.
ENVELOPE DELAY DISTORTION
If the propagation delay changes with frequency, we would then have the condition where the lower frequencies—such as
300 Hz—may arrive earlier or later than the higher frequencies—such as 3000 Hz. For voice communication, this would
probably not be noticeable. However, for data communication using modems, this could affect the phase of the carrier or
the modulation technique that's used to encode the data.

When the propagation delay varies across the frequency range, we call this envelope delay distortion. We measure
propagation delay in microseconds (µs), and the reference is from the worst case to the best case.

Figure 6.9: Envelope Delay Distortion
LINE IMPAIRMENTS
Line impairments are faults in the line that occur due to either improper line terminations or equipment out of
specifications. These cannot be conditioned out, but can be measured to determine the amount of the impairment.

Crosstalk
Crosstalk is when one line induces a signal into another line. In voice communications, we often hear this as another
conversation going on in the background. In digital communication, this can cause severe disruption of the data transfer.
Cross talk can be caused by the overlapping of bands in a multiplexed system, or by poor shielding of cables running
close to one another. There are no specific communications standards that are applied to the measurement of crosstalk.

Figure 6.10: Crosstalk

Echo or Signal Return
All media have a preferred termination condition for perfect transfer of signal power. The signal arriving at the end of a
transmission line should be fully absorbed, otherwise it will be reflected back down the line to the sender (and appear as
an echo). Echo suppressors are often fitted to transmission lines to reduce this effect.

Figure 6.11: Echo or Signal Return

Usually during data transmission, these suppressors must be disabled or they will prevent return communication in full
duplex mode. Echo suppressors are disabled on the phone line if they hear carrier for 400 ms or more. If the carrier is
absent for 100 ms, the echo suppressor is reenabled.

Echo cancellers are currently used in modems to replicate the echo path response. These cancellers then combine the
results to eliminate the echo (thus, no signal interruption is necessary).

Frequency Shift
Frequency shift is the difference between the transmitted frequency and the received frequency. This is caused by the
lack of synchronization of the carrier oscillators.

Figure 6.12: Frequency Shift

Nonlinear Distortion
Nonlinear distortion changes the wave shape of the signal. If the signal was transmitted as a sine wave (and arrived as a
square wave), it would be an example of severe nonlinear distortion. Amplitude modulated carriers would suffer
drastically if the original wave shape was distorted.
Nonlinear distortion
JITTER: AMPLITUDE AND PHASE
Here are the two types of jitter:

Amplitude jitter

Phase jitter

Amplitude jitter is the small constantly changing swing in the amplitude of a signal. It is principally caused by power
supply noise (60 Hz) and ringing tone (20 Hz) on the signal.

Figure 6.13: Amplitude Jitter and Phase Jitter

Phase jitter is the small constantly changing swing in the phase of a signal. It may result in the pulses moving into time
slots that are allocated to other data pulses when used with time domain multiplexing.

Telephone company standards call for no more than 10 degrees between 20 and 300 Hz and no more than 15 degrees
between 4 and 20 Hz.

Transients: Impulse noise, gain hits, dropouts and phase hits.

Transients are irregulartimed impairments. They appear randomly, and are very difficult to troubleshoot. There are four
basic types of transients.

1. Impulse noise

2. Gain hits
3. Dropouts

4. Phase hits

1 Impulse Noise
Impulse noise is a sharp and quick spike on the signal that can come from many sources: electromagnetic interference,
lightning, sudden power switching, electromechanical switching, etc.. These appear on the telephone line as clicks and
pops: they're not a problem for voice communication, but can appear as a loss of data (or even as wrong data bits)
during data transfers. Impulse noise has a duration of less than 1 ms and their effect is dissipated within 4 ms.

2 Gain Hits
Gain hits are sudden increases in amplitude that last more than 4 ms. Telephone company standards allow for no more
than 8 gain hits in any 15 minute interval. A gain hit would be heard on a voice conversation as if the volume were turned
up for just an instance. Amplitude modulated carriers are particularly sensitive to gain hits.

Figure 6.14: Impulse Noise

Figure 6.15: Gain Hits

3 Dropouts
Dropouts are sudden losses of signal amplitude that are greater than 12 db, and last longer than 4 m sec. They cause
more errors than any other type of transients. Telephone company standards allow no more than 1 dropout for every 30
minute interval. Dropouts can be heard on a voice conversation (similar to call waiting), where the line goes dead for a
1/2 second. This is a sufficient loss of signal for some digital transfer protocols (such as SLIP), where the connection is
lost and would then have to be reestablished.

Figure 6.16: Dropouts

4 Phase Hits
Phase hits are either a sudden—and large—change in the received signal phase (20 degrees), or a frequency that lasts
longer than 4 m sec. Phase hits generally occur when switching between telcos, common carriers, or transmitters. FSK
and PSK are particularly sensitive to phase hits. The data may be incorrect until the outofphase condition is rectified.
The telephone company standard allows no more than 8 phase hits in any 15 minute period.

Figure 6.17: Phase Hits
MODEMS
Modems (modulators/demodulators) are data communication devices that convert digital signals to analog signals, and
vice versa. Modems allow digital transmissions over analog telephone lines. They allow people to connect their
computers with other computers, corporate office LANs, and the Internet. Modems are used to establish connections of
the telephone company's local loop, which is the analog copper cable that runs between homes and businesses to the
telephone company Central Office (CO). The phone company then switches the call like any other voice call to create a
pointtopoint link to the destination.

Figure 6.18: Voice Channel

A modem (modulator/demodulator) connects a terminal/computer (DTE) to the voice channel (dialup line).

Modems are devices which allow digital data signals to be transmitted across an analogue link. Modem stands for
modulator/demodulator. A modem changes the digital signal to an analogue frequency, and sends this tone across the
analogue link. At the other end, another modem receives the signal and converts it back to digital.

Figure 6.19: Modems in a network

Modem Commands
Modems can receive and act on commands from the computer terminal. Commands begin with the symbol AT and are
terminated with a carriage return. The range of commands is called the AT command set. Depending upon how the
modem is configured (whether to echo results of commands), the modem will return the symbol OK in response to a
command request if it is performed.

The AT Command Set
Hayes corporation developed a smart modem which accepted AT type commands. This is now a widely accepted
standard. The following is a brief list of the AT command set.

ATA Answer call

ATA/Repeat last command

ATC Turn modems carrier signal ON (ATC1) or OFF (ATC0)

ATD Dial a telephone number (ATDT2550789)
ATE Enable (ATE1) or disable (ATE0) the echo of characters to the screen

ATH Hang up the phone (ATH0) or pick up the phone (ATH1)

ATM Turn on modem speaker (ATM1) or turn off speaker (ATM0)

ATO Place modem online

ATP Pulse dial
ATS Set values in modem ‘S’ registers

ATT Touch tone dial

ATZ Reset the modem

Modem Status Lights
On external modems, a number of lights are provided to indicate modem status.

High Speed: the modem is operating at its highest available speed

Auto Answer: the model will automatically answer incoming calls

Carrier Detect: means it has detected a carrier signal from a remote computer

Off Hook: lights when the modem takes control of the phone line

Modem Ready: the modem is turned on and is ready

Receive Data: flickers to indicate incoming data

Send Data: flickers to indicate data is being transmitted

Basic Definition of Modem
The modem (DCE—Data Communication Equipment) is connected between the terminal/computer (DTE—Data Terminal
Equipment) and the phone line (voice channel). A modem converts the DTE (Data Terminal Equipment) digital signal to
an analog signal that the voice channel can use.

A modem is connected to the terminal/computer's RS232 serial port (25 pin male D connector) and the outgoing phone
line with an RJ11 cable connector (the same as on a telephone extension cord). Male connectors have pins, female
connectors have sockets.

Digital Connection
The connection between the modem and terminal/computer is a digital connection. A basic connection consists of a
Transmit Data (TXD) line, a Receive Data (RXD) line and many hardware handshaking control lines.

Figure 6.20: Digital Connection

The control lines determine whose turn it is to talk (modem or terminal), if the terminal/computer is turned on, if the
modem is turned on, if there is a connection to another modem, etc.

Analog Connection
The connection between the modem and the outside world (the phone line) is an analog connection. The voice channel
has a bandwidth of 0–4 kHz but only 300–3400 Hz is usable for data communications.

Figure 6.21: Analog Connection

The modem converts digital information into tones (frequencies) for transmitting through the phone lines. The tones are in
the 300–3400 Hz voice band.

External/Internal Modems
There are 2 basic physical types of modems: Internal and External modems. External modems sit next to the computer
and connect to the serial port using a straightthrough serial cable.

Figure 6.22: External/Internal Modems

An internal modems is a plugin circuit board that sits inside the computer. It incorporates the serial port onboard. They
are less expensive than external modems because they do not require a case, power supply and serial cable. They
appear to the communication programs as if they were an external modem for all practical purposes.

Figure 6.23: External/Internal Modems

Modem Types
There are many types of modems, the most common of which are:
Optical modem: Uses optical fiber cable instead of wire. The modem converts the digital signal to pulses
of light to be transmitted over optical lines (more commonly called a media adapter or transceiver).

Short haul modem: A modem used to transmit data over 20 miles or less. Modems we use at home or to
connect computers together among different offices in the same building are short haul modems.

Acoustic modem: A modem that couples to the telephone handset with what looks like suction cups that
contain a speaker and microphone. Used by traveling salespeople to connect to hotel phones.

Smart modem: A modem with a CPU (microprocessor) on board that uses the hayes AT command set.
This allows autoanswer and dial capability rather than manually dialing and answering.
Digital modem: Converts the RS232 digital signals to digital signals more suitable for transmission (also
called a media adapter or transceiver).
V.32 modem: A milestone modem that uses a 2400 baud modem with 4 bit encoding. This results in a
9600 bps (bits per second) transfer rate.

Baud is the speed at which the analog data is changing on the voice channel and bps is the speed at which the decoded
digital data is being transferred.

Figure 6.24: Baud Rate and bps

Features of Modems
Speed: The speed at which the modem can send data in bps (bits per second). Typical modem speeds
are: 300, 600, 1200, 2400, 4800, 9600, 14.4K, 19.2K, 28.8K bps.

Auto Dial/Redial: Smart modems can dial the phone number and auto redial if a busy signal is received.

Auto answer: Most modems have ring detect capability and can automatically answer the telephone at the
time of incoming call.
Self testing: Newer modems have selftesting features. They can test the digital connection to the
terminal/computer and the analog connection to a remote modem. They can also check the modem's
internal electronics.

Voice over data: Voice over data modems allow a voice conversation to take place while data is being
transmitted. This requires both the source and destination modems to have this feature.

Synchronous or Asynchronous transmission: Newer modems allow a choice of synchronous or
asynchronous transmission of data. Normally, modem transmission is asynchronous (we send individual
characters with just start and stop bits). Synchronous transmission or packet transmission is used in
specific applications.

Standards to all Modems
There are many standards for modem protocols, but the field has evened out lately. A protocol is a method by which two
devices, modems, agree to communicate. You can think of it as a language that is agreed on.

Bell Labs and the CCITT have both set standards for modem protocols. Bell Labs no longer sets standards, although
they are still used in some locations. Most newer modems conform to the CCITT standards.
Some companies branched out and created modems using their own proprietary standard protocol. Such protocols are
published broadly so that other manufacturers can create modems to work with it. Some of these proprietary standards
have become popular. Most modems are designed to work with several different protocols.

Almost all modems are said to be Hayescompatible. This refers to the set of commands used to operate the modem.
These commands will be listed in another section.

There is often a confusion between baud rate and bitspersecond (bps). The baud rate is the rate at which a signal
between two modems changes in one second. It is actually independent of the bps. Some modems send one bit of data
per modulation, so 300 baud would be 300 bps. Other modems are set to send more data per modulation, meaning that
the bps will be higher than the baud rate. Therefore, the baud rate is really rather meaningless in purchasing a modem.
You should be interested in the bps, as this is concerned with the actual speed.

Modem Speeds/Standards
The following table 6.1 lists some of the ITUT modem standards. The modulation techniques described in the table are
FSK (frequencyshift keying), PSK (phaseshift keying), QAM (quadrature amplitude modulation), and TCM (trelliscoded
modulation). See "modulation techniques."

Table 6.1
Open table as spreadsheet
V.22 1,200 bits/sec, fullduplex, PSK encoding
V.22bis 2,400 bits/sec, fullduplex, QAM encoding
V.32 Asynchronous/synchronous, 4,800/9,600 bits/sec, QAM encoding
V.32bis Asynchronous/synchronous, 14,400 bits/sec, TCM encoding
V.35 Defines high data rates over combined circuits
V.42 Defines error checking standards
V.42bis Defines modem compression using the Lempel Ziv method
V.44 The newest data compression standard, announced with V.92 in 2000
V.34 A standard for 28 Kbits/sec using TCM encoding
V.34+ Boosts V.34 to 33.6 Kbits/sec
V.90 56 Kbits/sec download speed, assuming only one digitalto analog conversion
V.92 Improves the upload speed of the V.90 standard from 33.6 Kbits/sec to 47 Kbits/sec
V.90 56Kbit/sec Modems

Two competing modem standards emerged in the late 1990s to take advantage of higher throughput rates that were
made possible by the configuration. Note that the ISP is connected to the CO with a digital connection. This allows for a
cleaner modem signal and faster rates, as explained in a moment. The two competing standards were U.S. robotics X2
and lucent/rockwell semiconductor systems K56flex. Both companies rushed to release 56K products, creating two
standards and a lot of consumer confusion. 56K technology requires that ISPs also install compatible modems, so the
technology advanced slowly at first because ISPs could not decide on which standard to use.

In 1998, the ITU created the V.90 standard, which is designed for connections that are digital at one end and have only
one digitaltoanalog conversion. V.90 harmonized the two competing X2 and K56flex proposals and has become a
worldwide standard.

V.90 specifies up to 56 Kbits/sec in the downstream mode and 31.2 Kbits/sec in the upstream mode. However, it is very
unlikely that anyone can reach those speeds.
The lastest modem standard is V.92, which is V.90 (56Kbits/sec) technology with a few refinements and some new
features. "V.PCM upstream" allows for upstream communication at up to 48 Kbits/sec (V.90 supported on 31.2 Kbits/sec
upstream). The bit rate should be high since most people always dial the same number for Internet access. "modem on
hold" allows a modem to suspend a data call and take an incoming voice call. When the data call resumes, a
transmission continues where it left off. This feature benefits homes with only one line, and is a useful feature if you've
ever tried to call someone that has been online for hours. "quick connect" shortens connect times by up to 50%. The
modem uses stored settings from previous calls (if they can be used) rather than renegotiating during the connect phase.
This feature is also useful in conjunction with "modem on hold" since it speeds up the resumption of the call.

Be aware of socalled "soft modems" (also called "Winmodems" or "controllerless modems"). These modems use your
computer's processors instead of an onboard processor. They are cheap because they have fewer chips, but it's a good
idea to pay a little more for a fullfunction modem. Some soft modems run modulation/demodulation in hardware, but run
other functions such as compression, error control, and V.90 protocols in software. HSP (host signal processing)
modems run everything in software and are really nothing more than plugin telephone jacks. You're paying for software.
Usually, low price is the only indicator that a modem is a softmodem. Look at the box carefully. If it indicates that the
modem only runs under windows, it's a softmodem.

Bell 300 bps FSK—Half duplex
103
Bell 300 bps FSK—Full duplex
113
Bell 1200 baud half duplex
202
Bell 1200 bps DPSK (Dibit Phase Shift Keying)—
212A V.22 compatible 300 bps FSK (Frequency Shift
Keying)— NOT V.22 compatible
MNP1 Microcon networking protocol—Basic error
3 detection and control of errors
MNP4 Error correction + adapts to line conditions
MNP5 Error correction + adapts to line conditions; adds
compression technique to double the data
transfer rate
RS Cable and connector standard
232D
V.22 1200 bps DPSK (Dibit Phase Shift Keying) Bell
212A compatible 600 bps PSK (Phase Shift
Keying)— NOT Bell 212A compatible
V.22bis 2400 bps—International standard fall back in
Europe to V.22 Fall back in America to Bell 212A
V.24 European mechanical specifications for RS232D
V.26. Synchronous 2400 bps modem 1200 bps DPSK
full duplex
V.27 Synchronous 4800 bps DPSK modem
V.28 European electrical specifications for RS232D
V.29 Synchronous 9600 bps QAM
V.32 9600 bps QAM
V.32bis 14.4 Kbps QAM1
V.33 14.4 Kbps Trellis coded modulation for noise
immunity
V.34 28.8 Kbps modem standard
V.34bis 33.6 Kbps modem standard
V.42bis Compression technique to roughly double the
data transfer rate. Uses Automatic Repeat
Request ARQ and CRC (Cyclic Redundancy
Checking).
WE201 Synchronous Western Electric 2400 bps DPSK
WE208 Synchronous 4800 bps DPSK
WE209 Synchronous 9600 bps
Open table as spreadsheet

Summary of Modem Standard
CCITT/ITU Standards

(Comite Consultatif International Telegraphique et Telephoneique/International Telecommunications Union)

Open table as spreadsheet
Standard baud Bits Per Second Comment
(Hz)
V.22 600 1,200 The same as
the Bell
212A
standard but
different
handshaking.
V.22bis 600 2,400 (bis stands
for ‘second’)
V.32 2,400 4,800/9,600
V.32bis 2,400 V.32 +
7,200/12,000/14,400
V32terbo 2,400 V.32bis + 19,000 These two
‘proposed’
standards
contributed
V.FC 3,429 V.32bis + 28,000 and were
consolidated
in the V34
standard
V.34 3,429 V.32bis + Introduced in
28,000/33,600 June 94
V42/MNP4 Error correction standard Includes the
Microcom
MNP4
standard
Standard baud Bits Per Second Comment
(Hz)
V.42bis Compression standard

X2 Pushed by These two ‘proposed’ standards
(56K) 3Com/U.S. contributed and were
Robotics consolidated in the V90 standard
K56 Pushed by
Flex Rockwell (now
(56k) conexant),
Lucent
V90 V34 + 56,000 Introduced February ‘99
56k Asymmetric communication
Max upstream of 33.6k
Max downstream of 54k
Good connections only possible
if within 31/2 miles of local
exchange (NOT as the crow
flies)
V92 V34 + 56,000 Introduced January 2000
Faster Connection times
Increased Compression
throughput
Open table as spreadsheet
TRANSFER RATE VS. PC BUS SPEED
The lowest XT PC can outperform the fastest modem transfer rate. For example, an XT has an 8 bit parallel expansion
bus operating at 4.77 MHz. This equates to a data transfer rate of:

Compare this to the fastest modem transfer rates of 57.6 kbps.

Before the Internet, the usual connection was for one user to dial another user, or for a user to dial into a modem at a
bulletin board, place of business, or a school. In these cases, the connection is modem to modem across the PSTN.
The computer's outgoing digital signal is converted to an analog signal for transmission across the copper local loop to
the CO. At the CO, the analog signal is converted back to digital for transmission across its digital backbones. At the
destination CO, the signal is once again converted from digital to analog and transmitted to the end user's modem, where
it is converted back to digital.

In recent years, the 2400 bits per second modem that could carry email has become obsolete. 14.4 Kbps and 28.8 Kbps
modems were temporary landing places on the way to the much higher bandwidth devices and carriers of tomorrow.
From early 1998, most new personal computers came with 56 Kbps modems. By comparison, using a digital Integrated
Services digital network adapter instead of a conventional modem, the same telephone wire can now carry up to 128
Kbps. With Digital Subscriber Line (DSL) systems, now being deployed in a number of communities, bandwidth on
twistedpair can be in the megabit range.

Today, most users dial into the Internet, where they connect with other users via email, collaborative applications,
instant messaging, and so on. Users also connect with their home networks over the Internet by using secure VPN
technology. This allows users to dial into a local number and connect to the corporate office across the Internet. The
bottom half of figure 6.25 illustrates how most modem connections are made in today's Internetconnected environment.
The important point of this configuration is that the ISP is connected to the telephone company's CO with a digital
connection. This configuration makes highspeed 56Kbit/sec modem technology possible.

Figure 6.25: Modem

There are also different types of technical modems:

Consumer voicegrade Modems: Most offtheshelf modems are designed to allow PC users to communicate over the
voice phone system. The modems employ compatible communication techniques that comply with several standards,
most notably the ITU V series standards (previously called the CCITT standards).

Broadband Modems: These are modems for nontelephone system connections. A company may set up its own
dedicated lines or microwave towers and use broadband modems to achieve very high data rates between those sites.

A modem puts an AC (alternating current) carrier signal on the line (in the 1,000–2,000 Hz range) and adds digital
information to this signal. A modem at the receiving end then extracts the digital information from the carrier signal.
Modulation and demodulation are covered more fully under "modulation techniques." Two other related topics are "error
detection and correction" and "signals."

There are some restrictions in the local loop. Long ago, the telephone company established the frequency range of 300–
3,300 Hz as the range for voice. The telephone switching system filters out higher frequencies to eliminate noise.
Unfortunately, this filtering equipment prevents taking advantage of higher frequencies and wider bandwidth that could
help boost data transmission rates. Therefore, a number of techniques, including encoding and compression, have been
developed to squeeze as much digital data into the 300–3,300 Hz range as possible.

A voicegrade modem may be either an external box or an internally mounted circuit board. The PC card format is also
available for portable computers. The external modem connects to a computer serial port via an RS232 serial cable.

Carriers and service providers often have racks of modems that users dial into. These are connected to remote access
servers that manage the modems and authenticate users. Modems are usually integrated into RAC (remote access
concentrator) devices
Modems are asynchronous, synchronous, or both. An asynchronous transmission can be compared to the sporadic flow
of cars on a highway, while a synchronous transmission can be compared to a steady stream of boxcars on a train track.
In asynchronous mode, data is framed and each frame is sent independently. The receiver must be able to detect when
a frame starts and ends. In asynchronous mode, a clock signal separates the beginning and end of each frame of data.
Synchronous modems are generally more efficient and are typically used for dedicated links between two sites and long
streaming data transfers. Most consumer modems are asynchronous.

When one modem "calls" another, the destination modem answers and a signal exchange takes place that establishes
the parameters for a communication session. The negotiation process determines the maximum signaling rate available
between the two modems, as well as the use of compression.

A fullduplex modem will send signals in both directions at the same time and at the same rate. Newer highspeed (56
Kbit/sec) voicegrade modems such as the V.90 series are asymmetrical, meaning that the download channel has a
higher rate than the upload channel.

FSK Modems
The last functional block in a basic lowtomedium speed data link is the modulator/demodulator or modem, dataset, or
DCE. It is the job of the modem to convert the serial digital data stream from a UART to a series of audio sine waves or
tones. For a Frequency Shift Key (FSK) modem, one tone represents a logic one (mark) and another is used for a logic
zero (space). On the receiver end, these tones are converted back to digital levels by the receiver's modem.

On the transmit side, serial data is supplied by the UART through a RS232 character interface, on the TX data input and
fed to a VoltageControlled Oscillator (VCO). The VCO acts as a frequency modulator by converting DC voltages into
different analog sine waves. These sine waves have a different frequency for each level of DC voltage applied. Hence, a
tone at one frequency is produced for a logic one's DC voltage and a second tone at a different frequency is produced for
the space logic DC voltage level.

The requesttosend (RTS) input from the terminal (or UART) initiates the modem's RTS/CTS delay, which varies with
each modem type and requirements. The general purpose of the delay is to give modem circuits time to turn on and
stabilize before handling data from the UART. Once the delay is completed, the modem generates a CleartoSend (CTS)
signal back to the terminal completing the local data link handshaking. An active CTS enables the voltage controlled
oscillator, which, sensing idle line once from the UART, immediately outputs a mark tone on the telephone lines.

Intelligent Modems
Almost every modem does more, with appropriate software, a user can enter commands directing the modem to take
certain actions such as dialing or answering the phone. Such intelligent modems have a processor to carry out functions.

Hays microcomputer products, inc., developed a modem that has become commonplace. The hayes modems are
designed to respond to a set of instructions called AT commands. Using appropriate software, the user enters the letters
AT followed by the command. Each command consists one or more letters followed by parameters.

Two common parameters are the D and T commands. The letter D represents dial and instructs the modem to dial the
number that follows the command. The letter T indicates the modem should use tone dialing as opposed to pulse dialing.
To illustrate, a user wanting the modem to dial the number 5551483 from a touchtone phone would enter the command
ATDT5551483.

In some cases, when dialing from within an organization we must first enter 9, wait for another dial tone, and then dial the
number. If it simply dials the number, before the second dial tone, it may try to connect to an operator's recorded
message saying, "Your phone call cannot be completed…." We can instruct some modems to pause by including a
comma (,) as part of the command. For example, if we must dial 9, wait, and then dial 5551483, the appropriate
command would be ATDT9,5551483.

The different modem commands can be entered from a PC using appropriate software. Some software also allows us to
write several commands in a file. The files may be called scripts or macros; they allow the user to execute the
commands on a file many times without having to type them each time. The feature is useful in cases where we
frequently dial up the same line and go through the same logon procedures.

Null Modems
Sometimes, we want to allow two devices such as PCs to communicate directly, that is, with no network or DCES
between them, in this case, our first reaction might be to connect their RS232 ports with a cable and let the protocols do
their job. After all, they both send and receive from the RS232 ports. Using a simple cable, however, connects the same
pins on each side. For example, the cable would connect pin 2 of each DTE. The problem is that both pins try to send
over the same line. The first DTE sends data and the second receives it over line 2. Since the second DTE expects to
receive data over line 3, the direct connection will not work. Similarly, since the cable connects pin 3 on each end, both
expect to receive on the same circuit, but neither sends over it.
One solution to this problem is to connect the DTEs but cross some circuits. A null modem may be either a cable
connecting different pins on each connector or a device that simply crosses connections using existing cables. Thus
Null modem is used for connecting two PCs in the same room.
CARRIER
1. In information technology, a carrier (or carrier signal) is a transmitted electromagnetic pulse or wave at a
steady base frequency of alternation on which information can be imposed by increasing signal strength,
varying the base frequency, varying the wave phase, or other means. This variation is called modulation.
With the advent of laser transmission over optical fiber media, a carrier can also be a lasergenerated light
beam on which information is imposed.

Carrier detect (see modem lights) is a control signal between a modem and a computer that indicates that
the modem detects a "live" carrier that can be used for sending and receiving information.

2. In the telecommunications industry, a carrier is a telephone or other company that sells or rents
telecommunication transmission services. A Local Exchange carrier (LEC) is a local phone company and
an InterExchange Carrier (IEC or IXC) carries longdistance calls.

Digital T Carriers
As technology improved the speed at which electronic circuits could dependably operate, the need to increase the rate at
which data is transferred also increased. To respond to the need for interconnecting computers and database facilities,
the telephone company turned to supplying lines for carrying data in digital form. As well as supplying the lines, the
telephone company also established the limitations for their use. The basic designation for these lines are T carriers and
they begin with basic T1 carrier specified for the bell system voice network.

Each channel is sent, using time division multiplexing of eight bits per channel at a time. Since 24 channel are sent at
once, a total of 8.24 bits, or 192 data bits, are sent per transmission period. An additional frame boundary bit (like a
start/stop bit) increases the total to 193 bits per frame (transmission). Dividing the total data rate by the total number of
bits per frame yields a frame rate of 8,000 frames per second.

MCM was first used in analog military communications in the 1950s. Recently, MCM has attracted attention as a means
of enhancing the bandwidth of digital communications over media with physical limitations. The scheme is used in some
audio broadcast services. The technology lends itself to digital television, and is used as a method of obtaining high data
speeds in Asymmetric Digital Subscriber Line (ADSL) systems. MCM is also used in Wireless Local Area Networks
(WLANs).
SWITCHING
We know that networks establish communication links based on either circuit or packet switching. For highspeed optical
transmission, packet switching holds the promise for more efficient data transfer.

Network packet switching can be accomplished in a straightforward manner by requiring a node to optoelectronically
detect and transmit each and every incoming optical data packet. As for the routing, all the switching functions can
occur in the electrical domain prior to optical retransmission of the signal. Unfortunately, this approach suffers from an
optoelectronic speed bottleneck. Alternatively, much research is focused toward maintaining an alloptical data path and
performing the switching functions all optically with only some electronic control of the optical components. However,
there are many difficulties with optical switching, for instance:

1. A redirection of an optical path is not easy since photons do not have as strong interaction with their
environment as electrons do.

2. Switching has to be extremely fast due to the high speed of the incoming signal.

3. Switching nodes cannot easily tap a signal and acquire information about the channel.

Switching Network
Figure 6.26 shows a switching network. Lines are the media links. Ovals are called network nodes. Media links simply
carry data from one point to other. Nodes take the incoming data and route them to an output port.

Figure 6.26: Switching Network

If two different communication paths intersect through this network they have to share some resources. Two paths can
share a media link or a network node.
NETWORKS
Historically, most communication systems have started with pointtopoint links which directly connect together the users
wishing to communicate using a dedicated communication circuits. As the distance between users increases beyond the
length of the cable (e.g., the length of a telegraph wire), the connection between the users was formed by a number of
sections which were connected endtoend in series to form the circuit. The connection between the users (A and D) in figure
6.27 (i.e., A and D) is represented by a series of links (AB, BC, and CD) each link connects two entities known as nodes.
For a pointtopoint circuit, (also known as a permanent circuit) the nodes are patch panels which provide a simple
connection between the two links (i.e., the two transmission circuits).

Figure 6.27: A Connection between Two Systems A and D Formed from 3 Links

As t