DATA COMMUNICATION

LECTURES NOTE

Department of Computer Science & Engineering

Government College of Engineering Keonjhar
Odisha - 758001
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Module - I
INTRODUCTION TO DATA COMMUNICATION AND NETWORKING: Standards
Organizations for Data Communication, Layered Network Architecture, Open Systems
Interconnection, Data Communication Circuits, Serial and parallel Data Transmission, Data
Communication Networks, Alternate Protocol Suites.
SIGNALS, NOISE, MODULATION, AND DEMODULATION: Signal Analysis,
Electrical Noise and Signal-to-Noise Ratio, Analog Modulation Systems, Information
Capacity, Bits, Bit Rate, Baud, and M-ary Encoding, Digital Modulation.

Module - II
METALLIC CABLE TRANSMISSION MEDIA: Metallic Transmission Lines,
Transverse Electromagnetic Waves, Characteristics of Electromagnetic Waves.
OPTICAL FIBER TRANSMISSION MEDIA : Advantages of Optical Fiber Cables,
Disadvantages of Optical Fiber Cables, Electromagnetic spectrum, Optical Fiber
Communications System Block Diagram, Optical Fiber construction, Propagation of Light
Through an Optical fiber Cable, Optical Fiber Modes and Classifications, Optical Fiber
Comparison, Losses in Optical Fiber Cables, Light sources, Light Detectors, Lasers.

Module - III
DIGITAL TRANSMISSION: Pulse Modulation, Pulse code Modulation, Dynamic Range,
Signal Voltage–to-Quantization Noise Voltage Ratio, Linear Versus Nonlinear PCM Codes,
Commanding, PCM Line Speed, Delta Modulation PCM and Differential PCM.
MULTIPLEXING AND T CARRIERS: Time- Division Multiplexing, T1 Digital
Carrier System, Digital Line Encoding, T Carrier systems, Frequency- Division Multiplexing,
Wavelength- Division Multiplexing, Synchronous Optical Network.

Module-IV:
WIRELESS COMMUNICATIONS SYSTEMS: Electromagnetic Polarization,
Electromagnetic Radiation, Optical Properties of Radio Waves, Terrestrial Propagation of
Electromagnetic Waves, Skip Distance, Free-Space Path Loss, Microwave Communications
Systems, Satellite Communications Systems.

Module-V:
DATA COMMUNICATION CODES, ERROR CONTROL, AND DATA FORMAT:
Data Communication Character Codes, Bar Codes, Error Control, Error Detection and
Correction, Character Synchronization.
DATA COMMUNICATION EQUIPMENT: Digital Service Unit and Channel Service
Unit, Voice-Band Data Communication Modems, Bell Systems- Compatible Voice- Band
Modems, Voice-Band Modern Block Diagram, Voice- Band Modem Classifications,
Asynchronous Voice-Band Modems, Synchronous Voice-Band Modems, Modem
Synchronization, 56K Modems, Modem Control: The AT Command Set, Cable Modems .

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BOOKS:
1. Introduction to Data Communication and Networking, Wayne Tomasi, Pearson
Education.
2. Data Communication and Networking, Behrouz A Forouzan, Fourth
Edition.TMH.
3. Data and Computer communications, 8/e, William Stallings, PHI.
4. Computer Communications and Networking Technologies, Gallow, Second
Edition Thomson .
5. Computer Networking and Internet, Fred Halsll, Lingana Gouda Kulkarni, Fifth
Edition, Pearson Education.

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Module – I

PART – A
INTRODUCTION TO DATA COMMUNICATION AND NETWORKING:

Standards Organizations for Data Communications

Layered Network Architecture
Open Systems Interconnection
Data Communications Circuits
Serial and parallel Data Transmission
Data Communications Circuit Arrangements
Data Communications Networks
Alternate Protocol Suites.

PART – B
SIGNALS, NOISE, MODULATION, AND DEMODULATION:

Signal Analysis
Electrical Noise and Signal-to-Noise Ratio
Analog Modulation Systems
Information Capacity
Bits, Bit Rate, Baud
M-ary Encoding, Digital Modulation

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Introduction to Data Communications:

In Data Communication, data generally are defined as information that is stored in digital form.
Data Communication is the process of transferring digital information between two or more points.
When we communicate, we are sharing information. This sharing can be local or remote. Between
individuals, local communication usually occurs face to face, while remote communication takes
place over distance.
Information is defined as the knowledge or intelligence. Data Communication can be
summarized as the transmission, reception, and processing of digital information. For Data
Communication to occur, the communicating devices must be part of a communication system made
up of a combination of hardware (physical equipment) and software (programs).
The effectiveness of a Data Communication system depends on four fundamental characteristics:
delivery, accuracy, timeliness, and jitter.

1. Delivery: The data should be delivered to the correct destination and correct user.
2. Accuracy: The communication system should deliver the data accurately, without introducing
any errors. The data may get corrupted during transmission affecting the accuracy of the
delivered data.
3. Timeliness: Audio and Video data has to be delivered in a timely manner without any delay;
such a data delivery is called real time transmission of data.
4. Jitter: It is the variation in the packet arrival time. Uneven Jitter may affect the timeliness of
data being transmitted.

A Data Communication system has five components:

1. Message: The message is the information (data) to be communicated. Popular forms of

information include text, numbers, pictures, audio, and video.
2. Sender: The sender is the device that sends the data message. It can be a computer, workstation,
telephone handset, video camera, and so on.
3. Receiver: The receiver is the device that receives the message. It can be a computer,
workstation, telephone handset, television, and so on.
4. Transmission medium: The transmission medium is the physical path by which a message
travels from sender to receiver. Some examples of transmission media include twisted-pair wire,
coaxial cable, fiber-optic cable, and radio waves.
5. Protocol: A protocol is a set of rules that governs data communication. A Protocol is a necessity
in data communications without which the communicating entities are like two persons trying to
talk to each other in a different language without know the other language.

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DATA:
Data is collection of raw facts which is processed to deduce information.
There may be different forms in which data may be represented. Some of the forms of data used in
communications are as follows:

1. Text:
Text includes combination of alphabets in small case as well as upper case.
It is stored as a pattern of bits. Prevalent encoding system: ASCII, Unicode

2. Numbers:
Numbers include combination of digits from 0 to 9.
It is stored as a pattern of bits. Prevalent encoding system: ASCII, Unicode
3. Images:
An image is worth a thousand word is a very famous saying. In computers images are digitally
stored.

A Pixel is the smallest element of an image. To put it in simple terms, a picture or image is a
matrix of pixel elements.

The pixels are represented in the form of bits. Depending upon the type of image (black n
white or color) each pixel would require different number of bits to represent the value of a
pixel.

• The size of an image depends upon the number of pixels (also called resolution) and the bit
pattern used to indicate the value of each pixel.

Example: if an image is purely black and white (two color) each pixel can be represented by
a value either 0 or 1, so an image made up of 10 x 10 pixel elements would require only 100
bits in memory to be stored.

On the other hand an image that includes gray may require 2 bits to represent every pixel
value (00 - black, 01 – dark gray, 10– light gray, 11 –white). So the same 10 x 10 pixel image would
now require 200 bits of memory to be stored.
Commonly used Image formats: jpg, png, bmp, etc

4. Audio: Data can also be in the form of sound which can be recorded and broadcasted.

Audio data is continuous, not discrete.

Example: What we hear on the radio is a source of data or information.

5. Video: Video refers to broadcasting of data in form of picture or movie.

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Standards Organizations for Data Communication:

An association of organizations, governments, manufacturers and users form the standards
organizations and are responsible for developing, coordinating and maintaining the standards. The
intent is that all Data Communication equipment manufacturers and users comply with these
standards. The primary standards organizations for Data Communication are:

ISO

ITU-T IEEE ANSI

EIA TIA

IAB

IETF IRTF

Fig. 1 Standards Organisations for Data & Network Communication

1. International Standard Organization (ISO)

ISO is the international organization for standardization on a wide range of subjects. ISO was
started in 1946.The ISO creates the sets of rules and standards for graphics and document exchange
and provides models for equipment and system compatibility, quality and reduced costs. It is
comprised mainly of members from the standards committee of various governments throughout the
world. The ISO is also responsible for endorsing and coordinating the work of the other standards

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organizations. The member body of the ISO from the United States is the American National
Standards Institute (ANSI).

2. International Telecommunications Union-Telecommunication Sector (ITU-T)

ITU-T is one of the four permanent parts of the International Telecommunications Union
based in Geneva, Switzerland. It was formerly called as CCITT (Committee Consultant for
International Telephony and Telegraphy). It develops the recommended sets of rules and standards
for telephone and data communications. ITU-T membership consists of government authorities and
representatives from many countries and it is the present standards organization for the United
Nations. It has developed three sets of specifications mentioned below:
(a) The V series for modem interfacing and data transmission over telephone lines
(b) The X series for data transmission over public digital networks, Email and
directory services.
(c) The I and Q series for Integrated Services Digital Network (ISDN) and its extension
is Broadband ISDN.
The ITU-T is separated into 14 study groups.

3. Institute of Electrical and Electronics Engineers (IEEE)

IEEE is an international professional organization founded in United States and is
compromised of electronics, computer and communications engineers. It is currently the world’s
largest professional society with over 400,000 members. It develops communication and information
processing standards with the underlying goal of advancing theory, creativity, and product quality in
any field related to electrical engineering.

4. American National Standards Institute (ANSI)

ANSI is the official standards agency for the United States and is the U.S voting
representative for the ISO. ANSI is a completely private, non-profit organization comprised of
equipment manufacturers and users of data processing equipment and services. ANSI membership is
comprised of people form professional societies, industry associations, governmental and regulatory
bodies, and consumer goods.

5. Electronics Industry Association (EIA)

EIA is a non-profit U.S. trade association that establishes and recommends industrial
standards. EIA activities include standards development, increasing public awareness, and lobbying
and it is responsible for developing the RS (recommended standard) series of standards for data and
communications.

6. Telecommunications Industry Association (TIA)

TIA is the leading trade association in the communications and information technology
industry. It facilitates business development opportunities through market development, trade
promotion, trade shows, and standards development. It represents manufacturers of communications
and information technology products and also facilitates the convergence of new communications
networks.

7. Internet Architecture Board (IAB)

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IAB earlier known as Internet Activities Board is a committee created by ARPA (Advanced
Research Projects Agency) so as to analyze the activities of ARPANET whose purpose is to
accelerate the advancement of technologies useful for U.S military. IAB is a technical advisory group
of the Internet Society and its responsibilities are:
I. Oversees the architecture protocols and procedures used by the Internet.
II. Manages the processes used to create Internet Standards and also serves as an appeal board
for complaints regarding improper execution of standardization process.
III. Responsible for administration of the various Internet assigned numbers
IV. Acts as a representative for Internet Society interest in liaison relationships with other
organizations.
V. Acts as a source of advice and guidance to the board of trustees and officers of Internet
Society concerning various aspects of internet and its technologies.

8. Internet Engineering Task Force (IETF)

The IETF is a large international community of network designers, operators, vendors and
researchers concerned with the evolution of the Internet architecture and smooth operation of the
Internet.

9. Internet Research Task Force (IRTF)

The IRTF promotes research of importance to the evolution of the future Internet by creating
focused, long-term and small research groups working on topics related to Internet protocols,
applications, architecture and technology.

Layered Network Architecture:

To reduce the design complexity, most of the networks are organized as a series of layers or
levels, each one build upon one below it. The basic idea of a layered architecture is to divide the
design into small pieces. Each layer adds to the services provided by the lower layers in such a
manner that the highest layer is provided a full set of services to manage communications and run the
applications. The benefits of the layered models are modularity and clear interfaces, i.e. open
architecture and comparability between the different providers' components. A basic principle is to
ensure independence of layers by defining services provided by each layer to the next higher layer
without defining how the services are to be performed. This permits changes in a layer without
affecting other layers. The basic elements of a layered model are services, protocols and interfaces. A
service is a set of actions that a layer offers to another (higher) layer. Protocol is a set of rules that a
layer uses to exchange information with a peer entity. These rules concern both the contents and the
order of the messages used. Between the layers service interfaces are defined. The messages from
one layer to another are sent through those interfaces.
In a n-layer architecture, layer n on one machine carries on conversation with the layer n on
other machine. The rules and conventions used in this conversation are collectively known as the
layer-n protocol.
Basically, a protocol is an agreement between the communicating parties on how
communication is to proceed. Five-layer architecture is shown below; the entities comprising the
corresponding layers on different machines are called peers. In other words, it is the peers that
communicate using protocols.

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In reality, no data is transferred from layer n on one machine to layer n of another machine.
Instead, each layer passes data and control information to the layer immediately below it, until the
lowest layer is reached. Below layer-1 is the physical layer through which actual communication
occurs.
With layered architectures, communications between two corresponding layers requires a unit
of data called a protocol data unit (PDU). A PDU can be a header added at the beginning of a
message or a trailer appended to the end of a message.
Data flows downward through the layers in the source system and upwards at the destination
address. As data passes from one layer into another, headers and trailers are added and removed from
the PDU. This process of adding or removing PDU information is called
encapsulation/decapsulation.
Between each pair of adjacent layers there is an interface. The interface defines which
primitives operations and services the lower layer offers to the upper layer adjacent to it. A set of
layers and protocols is known as network architecture. A list of protocols used by a certain system,
one protocol per layer, is called protocol stack.

CONCEPT OF LAYERED TASK:

The main objective of a computer network is to be able to transfer the data from sender to
receiver. This task can be done by breaking it into small sub tasks, each of which are well defined.
Each subtask will have its own process or processes to do and will take specific inputs and give
specific outputs to the subtask before or after it. In more technical terms we can call these sub tasks
as layers.
In general, every task or job can be done by dividing it into sub task or layers. Consider the
example of sending a letter where the sender is in City A and receiver is in city B.

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The process of sending letter is shown below:

Fig. 2: Concept of layer task: Sending a letter

The Fig. 2 shows Sender, Receiver & Carrier Hierarchy of layers;
The sender site, the activities take place in the following descending order:
Higher Layer: The sender writes the letter along with the sender and receivers address and put it in
an envelope and drop it in the mailbox.
Middle Layer: The letter is picked up by the post man and delivered to the post office
Lower Layer: The letters at the post office are sorted and are ready to be transported through a
carrier.
During transition the letter may be carried by truck, plane or ship or a combination of transport
modes before it reaches the destination post office. At the Receiver site, the activities take place in
the following ascending order:
Lower Layer: The carrier delivers the letter to the destination post office
Middle Layer: After sorting, the letter is delivered to the receivers mail box
Higher Layer: The receiver picks up the letter, opens the envelope and reads it.
Hierarchy of layers: The activities in the entire task are organized into three layers. Each activity at
the sender or receiver side occurs in a particular order at the hierarchy.

The important and complex activities are organized into the Higher Layer and the simpler
ones into middle and lower layer.

Open Systems Interconnection (OSI):

International standard organization (ISO) established a committee in 1977 to develop
architecture for computer communication and the OSI model is the result of this effort. In 1984, the
Open Systems Interconnection (OSI) reference model was approved as an international standard for
communications architecture. The term “open” denotes the ability to connect any two systems which
conform to the reference model and associated standards. The OSI model describes how information
or data makes its way from application programs (such as spreadsheets) through a network medium
(such as wire) to another application program located on another network. The OSI reference model
divides the problem of moving information between computers over a network medium into SEVEN
smaller and more manageable problems. The seven layers are:

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The lower 4 layers (transport, network, data link and physical - Layers 4, 3, 2, and 1) are
concerned with the flow of data from end to end through the network. The upper four layers of the
OSI model (application, presentation and session - Layers 7, 6 and 5) are orientated more toward
services to the applications. Data is encapsulated with the necessary protocol information as it moves
down the layers before network transit.

As with any layered architecture, overhead information is added to a PDU in the form of
headers and trailers. Each layer provides a service to the layer above it in the protocol specification.
Each layer communicates with the same layer’s software or hardware on other computers.

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Physical Layer (The physical layer is responsible for transmitting individual bits from one node to the next)
The physical layer is the lowest layer of the OSI hierarchy and coordinates the functions
required to transmit a bit stream over a physical medium. It also defines the procedures and functions
that physical devices and interfaces have to perform for transmission occur. The physical layer
specifies the type of transmission medium and the transmission mode (simplex, half duplex or full
duplex) and the physical, electrical, functional and procedural standards for accessing Data
Communication networks.

Transmission media defined by the physical layer include metallic cable, optical fiber cable
or wireless radio-wave propagation. The physical layer also includes the carrier system used to
propagate the data signals between points in the network. The carrier systems are simply
communication systems that carry data through a system using either metallic or optical fiber cables
or wireless arrangements such as microwave, satellites and cellular radio systems.

Data-link Layer (the data link layer is responsible for transmitting frames from one node to the next)
The data link layer transforms the physical layer, a raw transmission facility, to a reliable
link and is responsible for node-to-node delivery.

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It makes the physical layer appear error free to the upper layer (network layer).The data link
layer packages data from the physical layer into groups called blocks, frames or packets. If frames
are to be distributed to different systems on the network, the data link layer adds a header to the
frame to define the physical address of the sender (source address) and/or receiver (destination
address) of the frame. The data-link layer provides flow-control, access-control, and error-control.

Network Layer (is responsible for the delivery of individual packets from the source host to the destination host)

The network layer provides details that enable data to be routed between devices in an
environment using multiple networks, sub-networks or both. This is responsible for addressing
messages and data so they are sent to the correct destination, and for translating logical addresses and
names (like a machine name FLAME) into physical addresses. This layer is also responsible for
finding a path through the network to the destination computer.

The network layer provides the upper layers of the hierarchy with independence from the
data transmission and switching technologies used to interconnect systems. Networking components
that operate at the network layer include routers and their software.

Transport Layer (is responsible for delivery of a message from one process to another)
The transport layer controls and ensures the end-to-end integrity of the data message
propagated through the network between two devices, providing the reliable, transparent transfer of
data between two endpoints.

Transport layer responsibilities include message routing, segmenting, error recovery and two
types of basic services to an upper-layer protocol: connection oriented and connectionless. The
transport layer is the highest layer in the OSI hierarchy in terms of communications and may provide
data tracking, connection flow control, sequencing of data, error checking, and application
addressing and identification.

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Session Layer (responsible for dialog control and synchronization)

Session layer, sometimes called the dialog controller provides mechanism for controlling the
dialogue between the two end systems. It defines how to start, control and end conversations (called
sessions) between applications.

Session layer protocols provide the logical connection entities at the application layer. These
applications include file transfer protocols and sending email. Session responsibilities include
network log-on and log-off procedures and user authentication. Session layer characteristics include
virtual connections between applications, entities, synchronization of data flow for recovery
purposes, creation of dialogue units and activity units, connection parameter negotiation, and
partitioning services into functional groups.

Presentation Layer (responsible for translation, compression, and encryption)

The presentation layer provides independence to the application processes by addressing any
code or syntax conversion necessary to present the data to the network in a common communications
format. It specifies how end-user applications should format the data.

The presentation layer translated between different data formats and protocols. Presentation
functions include data file formatting, encoding, encryption and decryption of data messages,
dialogue procedures, data Compression algorithms, synchronization, interruption, and termination.

Application Layer (responsible for providing services to the user)

The application layer is the highest layer in the hierarchy and is analogous to the general
manager of the network by providing access to the OSI environment. The applications layer provides
distributed information services and controls the sequence of activities within and application and
also the sequence of events between the computer application and the user of another application.

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The application layer communicates directly with the user’s application program. User
application processes require application layer service elements to access the networking
environment. The service elements are of two types: CASEs (common application service elements)
satisfying particular needs of application processes like association control, concurrence and
recovery. The second type is SASE (specific application service elements) which include TCP/IP
stack, FTP, SNMP, Telnet and SMTP.

Data Communication Circuits:

The underlying purpose of a digital communications circuit is to provide a transmission path
between locations and to transfer digital information from one station (node, where computers or
other digital equipment are located) to another using electronic circuits. Data Communication circuits
utilize electronic communications equipment and facilities to interconnect digital computer
equipment. Communication facilities are physical means of interconnecting stations and are provided
to Data Communication users through public telephone networks (PTN), public data networks
(PDN), and a multitude of private Data Communication systems.
The following figure shows a simple two-station Data Communication circuit. The main
components are:

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Source: This device generates the data to be transmitted; examples are mainframe computer,
personal computer, workstation etc. The source equipment provides a means for humans to enter data
into system.

Transmitter: - A transmitter transforms and encodes the information in such a way as to produce
electromagnetic signals that can be transmitted across some sort of transmission system. For
example, a modem takes a digital bit stream from an attached device such as a personal computer and
transforms that bit stream into an analog signal that can be handled by the telephone network.

Transmission medium: - The transmission medium carries the encoded signals from the transmitter
to the receiver. Different types of transmission media include free-space radio transmission (i.e. all
forms of wireless transmission) and physical facilities such as metallic and optical fiber cables.

Receiver: - The receiver accepts the signal from the transmission medium and converts it into a form
that can be handled by the destination device. For example, a modem will accept an analog signal
coming from a network or transmission line and convert it into a digital bit stream.

Destination: - Takes the incoming data from the receiver and can be any kind of digital equipment
like the source.

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Serial and Parallel Data Transmission:

There are two methods of transmitting digital data namely parallel and serial transmissions.
In parallel data transmission, all bits of the binary data are transmitted simultaneously. For example,
to transmit an 8-bit binary number in parallel from one unit to another, eight transmission lines are
required. Each bit requires its own separate data path. All bits of a word are transmitted at the same
time. This method of transmission can move a significant amount of data in a given period of time.
Its disadvantage is the large number of interconnecting cables between the two units. For large binary
words, cabling becomes complex and expensive. This is particularly true if the distance between the
two units is great. Long multiwire cables are not only expensive, but also require special interfacing
to minimize noise and distortion problems. Serial data transmission is the process of transmitting
binary words a bit at a time. Since the bits time-share the transmission medium, only one
interconnecting lead is required.

While serial data transmission is much simpler and less expensive because of the use of a
single interconnecting line, it is a very slow method of data transmission. Serial data transmission is
useful in systems where high speed is not a requirement. Parallel communication is used for short-
distance Data Communication and within a computer, and serial transmission is used for long-
distance Data Communication.
Data Communication Circuit Arrangements:
A Data Communication circuit can be described in terms of circuit configuration and
transmission mode.

Circuit Configurations
Data Communication networks can be generally categorized as either two point or multipoint.
A two-point configuration involves only two locations or stations, whereas a multipoint configuration
involves three or more stations.

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A two-point circuit involves the transfer of digital information between a mainframe

computer and a personal computer, two mainframe computers or two Data Communication networks.
A multi-point network is generally used to interconnect a single mainframe computer (host) to many
personal computers or to interconnect many personal computers and capacity of the channel is either
Spatially shared: Devices can use the link simultaneously or Timeshare: Users take turns

Transmission Modes:
There are four modes of transmission for Data Communication circuits:

In simplex mode(SX), the communication is unidirectional, as on a one-way street. Only one of the
two devices on a link can transmit; the other can only receive. Commercial radio broadcasting is an
example. Simplex lines are also called receive-only, transmit-only or one-way-only lines.

In half-duplex(HDX) mode, each station can both transmit and receive, but not at the same time.
When one device is sending, the other can only receive, and vice versa. The half-duplex mode is used
in cases where there is no need for communication in both directions at the same time; the entire
capacity of the channel can be utilized for each direction. Citizens band (CB) radio is an example
where push to talk (PTT) is to be pressed or depressed while sending and transmitting.

In full-duplex mode(FDX) (called duplex), both stations can transmit and receive simultaneously.
One common example of full-duplex communication is the telephone network. The full-duplex mode
is used when communication in both directions is required all the time. The capacity of the channel
must be divided between the two directions.

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In full/full duplex (F/FDX) mode, transmission is possible in both directions at the same time but not
between the same two stations (i.e. station 1 transmitting to station 2, while receiving from station 3).
F/FDX is possible only on multipoint circuits. Postal system can be given as a person can be sending
a letter to one address and receive a letter from another address at the same time.

Data Communication Networks:

Any group of computers connected together can be called a Data Communication network,
and the process of sharing resources between computers over a Data Communication network is
called networking. The most important considerations of a Data Communication network are
performance, transmission rate, reliability and security.

Network Components, Functions, and Features

The major components of a network are end stations, applications and a network that will
support traffic between the end stations. Computer networks all share common devices, functions,
and features, including servers, clients, transmission media, shared data, shared printers and other
peripherals, hardware and software resources, network interface card (NIC), local operating system
(LOS) and the network operating system (NOS).

Servers: Servers are computers that hold shared files, programs and the network operating system.
Servers provide access to network resources to all the users of the network and different kinds of
servers are present. Examples include file servers, print servers, mail servers, communication servers
etc.

Clients: Clients are computers that access and use the network and shared network resources. Client
computers are basically the customers (users) of the network, as they request and receive service
from the servers.

Shared Data: Shared data are data that file servers provide to clients, such as data files, printer
access programs, and e-mail.

Shared Printers and other peripherals: these are hardware resources provided to the users of the
network by servers. Resources provided include data files, printers, software, or any other items used
by the clients on the network.

Network interface card: Every computer in the network has a special expansion card called network
interface card (NIS), which prepares and sends data, receives data, and controls data flow between
the computer and the network. While transmitting, NIC passes frames of data on to the physical layer
and on the receiver side, the NIC processes bits received from the physical layer and processes the
message based on its contents.

Local operating system: A local operating system allows personal computers to access files, print to
a local printer, and have and use one or more disk and CD drives that are located on the computer.
Examples are MS-DOS, PC-DOS, UNIX, Macintosh, OS/2, Windows 95, 98, XP and Linux.

Network operating system: the NOS is a program that runs on computers and servers that allows the
computers to communicate over a network. The NOS provides services to clients such as log-in
features, password authentication, printer access, network administration functions and data file
sharing.

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Network Models:
Computer networks can be represented with two basic network models: peer-to-peer
client/server and dedicated client/server. The client/server method specifies the way in which two
computers can communicate with software over a network.

Peer-to-peer client/server network:

Here, all the computers share their resources, such as hard drives, printers and so on with all
the other computers on the network. Individual resources like disk drives, CD-ROM drives, and even
printers are transformed into shared, collective resources that are accessible from every PC. Unlike
client-server networks, where network information is stored on a centralized file server PC and made
available to tens, hundreds, or thousands client PCs, the information stored across peer-to-peer
networks is uniquely decentralized. Because peer-to-peer PCs have their own hard disk drives that
are accessible by all computers, each PC acts as both a client (information requestor) and a server
(information provider). The peer-to-peer network is an appropriate choice when there are fewer than
10 users on the network, security is not an issue and all the users are located in the same general area.
The advantages of peer-to-peer over client-server NOSs include:
No need for a network administrator
Network is fast/inexpensive to setup & maintain
Each PC can make backup copies of its data to other PCs for security.
Easiest type of network to build, peer-to-peer is perfect for both home and office use.

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Dedicated client/server network:

Here, one computer is designated as server and the rest of the computers are clients.
Dedicated Server Architecture can improve the efficiency of client server systems by using one
server for each application that exists within an organization. The designated servers store all the
networks shared files and applications programs and function only as servers and are not used as a
client or workstation. Client computers can access the servers and have shared files transferred to
them over the transmission medium. In some client/server networks, client computers submit jobs to
one of the servers and once they process the jobs, the results are sent back to the client computer.
In general, the dedicated client/server model is preferable to the peer-to-peer client/server
model for general purpose data networks.

Network Topologies:

In computer networking, topology refers to the layout of connected devices, i.e. how the
computers, cables, and other components within a Data Communication network are interconnected,
both physically and logically. The physical topology describes how the network is actually laid out,
and the logical topology describes how the data actually flow through the network. Two most basic
topologies are point-to-point and multipoint. A point-to-point topology usually connects two
mainframe computers for high-speed digital information. A multipoint topology connects three or
more stations through a single transmission medium and some examples are star, bus, ring, mesh and
hybrid.

Star topology:
A star topology is designed with each node (file server, workstations, and peripherals) connected
directly to a central network hub, switch, or concentrator. Data on a star network passes through the
hub, switch, or concentrator before continuing to its destination. The hub, switch, or concentrator
manages and controls all functions of the network. It also acts as a repeater for the data flow.

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Bus topology: Bus networks use a common backbone to connect all devices. A single cable, (the
backbone) functions as a shared communication medium that devices attach or tap into with an
interface connector. A device wanting to communicate with another device on the network sends a
broadcast message onto the wire that all other devices see, but only the intended recipient actually
accepts and processes the message. The bus topology is the simplest and most common method of
interconnecting computers. The two ends of the transmission line never touch to form a complete
loop. A bus topology is also known as multi drop or linear bus or a horizontal bus.

Ring topology:
In a ring network (sometimes called a loop), every device has exactly two neighbours for
communication purposes. All messages travel through a ring in the same direction (either
"clockwise" or "counter clockwise"). All the stations are interconnected in tandem (series) to form a
closed loop or circle. Transmissions are unidirectional and must propagate through all the stations in
the loop. Each computer acts like a repeater and the ring topology is similar to bus or star topologies.

Mesh topology:
The mesh topology incorporates a unique network design in which each computer on the network
connects to every other, creating a point-to-point connection between every device on the network.
Unlike each of the previous topologies, messages sent on a mesh network can take any of several possible
paths from source to destination. mesh network in which every device connects to every other is called
a full mesh. A disadvantage is that, a mesh network with n nodes must have n(n-1)/2 links and each
node must have n-1 I/O ports (links).

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Hybrid topology: This topology (sometimes called mixed topology) is simply combining two or
more of the traditional topologies to form a larger, more complex topology. Main aim is being able to
share the advantages of different topologies.

Network Classifications:
One way to categorize the different types of computer network designs is by their scope or
scale. Common examples of area network types are:

Local Area Network - Wireless Local Area Network

WAN - Wide Area Network - Metropolitan Area Network
SAN – Storage Area Network, System Area Network, Server Area Network, or sometimes
Small Area Network.
CAN - Campus Area Network, Controller Area Network, or sometimes Cluster Area
Network
PAN - Personal Area Network
DAN - Desk Area Network

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Local area network: A local area network (LAN) is a network that connects computers and
devices in a limited geographical area such as home, school, computer laboratory, office building, or
closely positioned group of buildings. LANs use a network operating system to provide two-way
communications at bit rates in the range of 10 Mbps to 100 Mbps. In addition to operating in a
limited space, LANs are also typically owned, controlled, and managed by a single person or
organization. They also tend to use certain connectivity technologies, primarily Ethernet and Token
Ring.

Advantages of LAN:
Share Resources efficiency
Individual workstation might survive network failure if it doesn’t rely upon others
component evolution independent of system evolution
Support heterogeneous hardware/software
Access to other LANs and WANs
Higher transfer rate with low error rates

Metropolitan area network:

A MAN is optimized for a larger geographical area than a LAN, ranging from several blocks
of buildings to entire cities. Its geographic scope falls between a WAN and LAN. A MAN might be a
single network like the cable television network or it usually interconnects a number of local area
networks (LANs) using a high-capacity backbone technology, such as fiber-optical links, and
provides up-link services to wide area networks and the Internet. MANs typically operate at speeds
of 1.5 Mbps to 10 Mbps and range from five miles to a few hundred miles in length. Examples of
MANs are FDDI (fiber distributed data interface) and ATM (asynchronous transfer mode)

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Wide area network: Wide area networks are the oldest type of Data Communication network that
provide relatively slow-speed, long-distance transmission of data, voice and video information over
relatively large and widely dispersed geographical areas, such as country or entire continent. WANs
interconnect routers in different locations. A WAN differs from a LAN in several important ways.
Most WANs (like the Internet) are not owned by any one organization but rather exist under
collective or distributed ownership and management. WANs tend to use technology like ATM,
Frame Relay and X.25 for connectivity over the longer distances.

Global area network: A GAN provides connections between countries around the entire globe.
Internet is a good example and is essentially a network comprised of other networks that interconnect
virtually every country in the world. GANs operate from 1.5 Mbps to 100 Gbps and cover thousands
of miles.

Campus Area Network: A network spanning multiple LANs but smaller than a MAN, such as on a
university or local business campus.

Storage Area Network: connects servers to data storage devices through a technology like Fibre
Channel.

System Area Network: - Links high-performance computers with high-speed connections in a

cluster configuration. Also known as Cluster Area Network.

Building backbone: - It is a network connection that normally carries traffic between departmental
LANs within a single company. It consists of a switch or router to provide connectivity to other
networks such as campus backbones, enterprise backbones, MANs, WANs etc

Camus backbone: - It is a network connection used to carry traffic to and from LANs located in
various buildings on campus. It normally uses optical fiber cables for the transmission media between
buildings and operates at relatively high transmission rates.

Enterprise networks: - It includes some or all of the above networks and components connected in a
cohesive and manageable fashion.

Alternate Protocol Suites:

The protocols other than OSI that are in wide spread used are TCP?IP and the Cisco three-
layer hierarchical model.

TCP/IP Protocol Suite

The U.S. Department of Defense (DoD) created the TCP/IP reference model because it
wanted a network that could survive any conditions, even a nuclear war. Transmission Control
Protocol/Internet Protocol (TCP/IP) {commonly known as internet suite} model is a set of
communication protocols that allow communication across multiple diverse networks. TCP/IP is a
hierarchical protocol comprised of either three or four layers. The three-layer version of TCP/IP
contains the network, transport and application layers. Four layer version specifies the host to
network layer.

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The designers of TCP/IP felt that the higher level protocols should include the session and
presentation layer details. They simply created an application layer that handles high-level
protocols, issues of representation, encoding, and dialog control. The TCP/IP combines all
application-related issues into one layer, and assures this data is properly packaged for the next layer.

The TCP/IP transport layer deals with the quality-of-service issues of reliability, flow
control, and error correction. One of its protocols, the transmission control protocol (TCP), provides
excellent and flexible ways to create reliable, well-flowing, low-error network communications. TCP
is a connection-oriented protocol. The other protocol is User Datagram Protocol (UDP) which is a
connection less protocol.

Common TCP/IP Protocols

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The purpose of the Internet layer is to send source packets from any network on the
internetwork and have them arrive at the destination independent of the path and networks they took
to get there. The specific protocol that governs this layer is called the Internet protocol (IP). Best
path determination and packet switching occur at this layer.
The network access layer also called the host-to-network layer is concerned with all of the
issues of physically delivering data packets using frames or cells.
Differences between OSI and TCP/IP
TCP/IP combines the presentation and session layer issues into its application layer
TCP/IP combines the OSI data link and physical layers into one layer
TCP/IP appears simpler because it has fewer layers

TCP/IP protocols are the standards around which the Internet developed, so the TCP/IP model
gains credibility just because of its protocols. In contrast, typically networks aren't built on the OSI
protocol, even though the OSI model is used as a guide.

Cisco Three Layer Model:

Cisco has defined a hierarchical model known as the hierarchical internetworking model. This
model simplifies the task of building a reliable, scalable, and less expensive hierarchical internetwork
because rather than focusing on packet construction; it focuses on the three functional areas, or
layers, of your network.

Core layer: This layer is considered the backbone of the network and includes the high-end switches
and high-speed cables such as fiber cables. This layer of the network does not route traffic at the
LAN. In addition, no packet manipulation is done by devices in this layer. Rather, this layer is
concerned with speed and ensures reliable delivery of packets.

Distribution layer: This layer includes LAN-based routers and layer 3 switches. This layer ensures
that packets are properly routed between subnets and VLANs in your enterprise. This layer is also
called the Workgroup layer. It also provides policy-based network connectivity, including:

Packet filtering (firewalling): Processes packets and regulates the transmission of packets
based on its source and destination information to create network borders
QoS: The router or layer 3 switches can read packets and prioritize delivery, based on
policies set
Access Layer Aggregation Point: The layer serves the aggregation point for the desktop
layer switches
Control Broadcast and Multicast: The layer serves as the boundary for broadcast and
multicast domains
Application Gateways: The layer allows you to create protocol gateways to and from
different network architectures.
The distribution layer also performs queuing and provides packet manipulation of the
network traffic.

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Access layer: This layer includes hubs and switches. This layer is also called the desktop layer
because it focuses on connecting client nodes, such as workstations to the network. This layer ensures
that packets are delivered to end user computers. At the access layer, you can:
Enable MAC address filtering: It is possible to program a switch to allow only certain to
access the connected LANs.
Create separate collision domains: A switch can create separate collision domains for each
connected node to improve performance.
Share bandwidth: You can allow the same network connection to handle all data.
Handle switch bandwidth: You can move data from one network to another to perform load
balancing.

The benefits of the Cisco hierarchical model includes:

High Performance: You can design high performance networks, where only certain layers
are susceptible to congestion.
Efficient management & troubleshooting: Allows you to efficiently organize network
management and isolate causes of network trouble.
Policy creation: You can easily create policies and specify filters and rules.
Scalability: You can grow the network easily by dividing your network into functional areas.
Behavior prediction: When planning or managing a network, the model allows you
determine what will happen to the network when new stresses are placed on it.

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Signals, Noise, Modulation and Demodulation

Computers transmit data using digital signals, sequences of specified voltage levels.
Computers sometimes communicate over telephone lines using analog signals, which are formed by
continuously varying voltage levels. Electrical signals can be in analog or digital form. With analog
signals, the amplitude changes continuously with respect to time with no breaks or discontinuities. A
sine wave is the most basic analog signal.

Digital signals are described as discrete; their amplitude maintains a constant level for a
prescribed period of time and then it changes to another level. If only two levels are possible, it is
called a binary signal. All binary signals are digital, but all digital signals are not necessarily binary.
Converting information signals to a different form is called modulation and the reverse process is
called demodulation. The modulating signal is the information and the signal being modulated is the
carrier.
Two basic types of electronic communications systems are analog and digital. An analog
digital communications system is a communications system in which energy is transmitted and
received in analog form and are also propagated through the system in analog form. Digital
communications covers a broad range of communications techniques including digital transmission
and digital modulation.

Signal Analysis
Mathematical signal analysis is used to analyze and predict the performance of the circuit on
the basis of the voltage distribution and frequency composition of the information signal.

Amplitude, Frequency and Phase

A cycle is one complete variation in the signal, and the period is the time the waveform takes to
complete on cycle. One cycle constitutes 360 degrees (or 2π radians). Sine waves can be described in
terms of three parameters: amplitude, frequency and phase.

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Amplitude (A): It is analogous to magnitude or displacement. The amplitude of a signal is the

magnitude of the signal at any point on the waveform. The amplitude of electrical signal is generally
measured in voltage. The maximum voltage of a signal in respect to its average value is called its
peak amplitude or peak voltage.

Frequency (f): The time of one cycle of a waveform is its period, which is measured in seconds.
Frequency is the number of cycles completed per second. The measurement unit for frequency is the
hertz, Hz. 1 Hz = 1 cycle per second. The frequency of the signal can be calculated from T=1/f

Phase (Ø): The phase of the signal is measured in degrees or radians with respect to a reference
point. A phase shift of 180 degrees corresponds to a shift of half a cycle.

A phase shift of 360 degrees corresponds to a shift of one complete cycle. If two sine waves
have the same frequency and occur at the same time, they are said to be in phase, or they are said to
out of phase. The difference in phase can be measured in degrees, and is called the phase angle,

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Varying Sine wave with respect to frequency and phase

Periodic Signals
A signal is periodic if it completes a pattern within a measurable time and is characterized by
amplitude, frequency and phase. Mathematically, a single frequency voltage wave form is

v(t) = V sin(2πft + θ),

v(t) is time-varying voltage sine wave
V is peak amplitude in volts
f is frequency in hertz
t is time in second
θ is phase in degrees or radians

It is called a periodic wave because, it repeats at a uniform rate. A series of sine, cosine or
square waves constitute an example of periodic waves, which can be analyzed in either the time
domain or the frequency domain.

Time domain:
Time domain is a term used to describe the analysis of mathematical functions, or physical
signals, with respect to time. In the time domain, the signal or function's value is known for all real
numbers, for the case of continuous time, or at various separate instants in the case of discrete time.
An oscilloscope is a time-domain tool commonly used to visualize real-world signals in the time
domain. A time domain graph shows how a signal changes over time.

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Frequency Domain: frequency domain is a term used to describe the analysis of mathematical
functions or signals with respect to frequency, rather than time. A spectrum analyser is a frequency-
domain instrument which displays amplitude-versus frequency plot (called a frequency spectrum). The
horizontal axis represents frequency and the vertical axis amplitude showing a vertical deflection for
each frequency present in the waveform, which is proportional to the amplitude of the frequency it
represents.

Examples for time domain and frequency domain

Complex Signals:
Any repetitive waveform that is comprised of more than one harmonically related sine or
cosine wave is called a non sinusoidal, complex wave. Fourier series is used to analyze the complex
periodic waves.

Fourier series: The Fourier series is used in signal analysis to represent the sinusoidal components of
non sinusoidal periodic waveforms. A Fourier series decomposes a periodic function or periodic signal into
a sum of simple oscillating functions, namely sines and cosines. It can be expressed as:

f(t) = A0 + A1 cosα + A2 cos2α + A3 cos3α + .......An cos nα

+ A0 + B1 sinβ + B2 sin2β + B3 sin3β +.........Bn sin nβ

Where α=β

Any periodic waveform is comprised of an average dc component and a series of harmonically

related sines or cosine waves. A harmonic is an integral multiple of the fundamental frequency.
Fundamental frequency is the first harmonic and equal to the frequency (repetition rate) of the
waveform. Second multiple is called second harmonic, third multiple is called third harmonic and so
forth.

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Wave symmetry: It describes the symmetry of a waveform in the time domain, i.e., its relative
position with respect to the horizontal (time) and vertical (amplitude) axes.
Even symmetry: If a periodic voltage waveform is symmetric about the vertical axis, it is said to have
axes, or mirror, symmetry and is called an even function. For all even functions, the
β Co-efficients are zero. Even function satisfy the condition f(t) = f(-t)

Odd symmetry: If a periodic voltage waveform is symmetric about a line midway between the vertical
axis and the negative horizontal axis and passing through the coordinate origin, it is said to have to
point or skew, symmetry and is called an odd function. For all odd functions, the α coefficients are
zero. Odd function satisfies f(t) = -f(-t)

Half-wave symmetry: If a periodic voltage waveform is such that the waveform for the first half cycle
repeats itself except with the opposite sign for the second half cycle, it is called to have half-wave
symmetry. Half-wave symmetry implies that the second half of the wave is exactly opposite to the first
half. A function with half-wave symmetry does not have to be even or odd, as this property requires
only that the shifted signal is opposite. Half-wave functions satisfy the condition f(t) = -f(T+t)/2
Frequency Spectrum and Bandwidth:
The frequency spectrum of a waveform consists of all the frequencies contained in the waveform
and their respective amplitudes plotted in the frequency domain.
Bandwidth of an information signal is simply the difference between the highest and lowest
frequencies contained in the information and the bandwidth of a communication channel is the
difference between the highest and lowest frequencies that the channel will allow to pass through it.
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Frequency Spectrum and Bandwidth:

The frequency spectrum of a waveform consists of all the frequencies contained in the waveform
and their respective amplitudes plotted in the frequency domain.
Bandwidth of an information signal is simply the difference between the highest and lowest
frequencies contained in the information and the bandwidth of a communication channel is the
difference between the highest and lowest frequencies that the channel will allow to pass through it.

Electrical Noise and Signal-To-Noise Ratio:

Noise is any disturbance or distortion that comes in the process of communication. Electrical
noise is defined as any undesirable electrical energy that falls within the pass band of the signal. A
noise signal consists of a mixture of frequencies with random amplitudes. Noise can originate in
various ways. The most prevalent and most interfering to Data Communication signals are man-made
noise, thermal noise, correlated noise, and impulse noise.

Man-made noise: It is the kind of noise produced by mankind. The main sources are spark-producing
mechanisms like commutators in electric motors, automobile ignition systems, ac power-generating
and switching equipment, and fluorescent lights. It is impulsive in nature and contains a wide range of
frequencies propagated in the free space like the radio waves. Man-made noise is most intense in more
densely populated areas and sometimes is referred to as industrial noise.

Thermal noise: This is the noise generated by thermal agitation of electrons in a conductor. It is also
referred to as white noise because of its uniform distribution across the entire electromagnetic
frequency spectrum. Noise power density is the thermal noise power present in a 1-Hz bandwidth and
is given by, No = KT.

Thermal noise is independent of frequency and thus thermal noise present in any bandwidth is,
N=KTB,
where N is thermal noise power in watts, K is Boltzmann's constant in joules per Kelvin, T is the
conductor temperature in Kelvin (0K = -273oC), and B is the bandwidth in hertz. Noise power is often
measured in dBm. From the equation above, noise power in a resistor at room temperature, in dBm, is:
NdBm = -174 dBm + 10 log B

Correlated noise: this noise is correlated to the signal and cannot be present in a circuit unless there is
a signal. Correlated noise is produced by nonlinear amplification and includes harmonic distortion and
inter modulation distortion. Harmonic distortion occurs when unwanted harmonics of a signal are
produced through nonlinear amplification and is also called amplitude distortion. Inter modulation
distortion is the generation of unwanted sum and difference frequencies produced when two or more
signals are amplified in a nonlinear device.

Impulse noise: This noise is characterized by high-amplitude peaks of short duration in the total noise
spectrum. It consists of sudden bursts of irregularly shaped pulses that generally last between a few
microseconds and several milliseconds, depending on their amplitude and origin. In case of voice
communications, impulse noise is very annoying as it generates a sharp popping or crackling sound
where as it is devastating in Data Circuits.

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Signal-to-noise power ratio: Signal-to-noise ratio (often abbreviated SNR or S/N) is defined as the
ratio of signal power to the noise power corrupting the signal. A ratio higher than 1:1 indicates more
signal than noise. Signal-to-noise ratio is defined as the power ratio between signal (meaningful
information) and the background noise (unwanted signal)

where P is average power in watts. The ratio often expressed in decibels as S/N (dBm) = 10 log(PS/PN)

Analog Modulation Systems:

A sine wave has three main components: amplitude, frequency and phase and can be expressed
as v(t) = V sin(2πft + θ). If the information signal is analog and the amplitude ‘V’ of the carrier is
varied proportional to the informational signal, amplitude modulation (AM) is produced. If the
frequency (f) is varied proportional to the information signal, frequency modulation (FM) is produced
and if the phase (θ) is varied proportional to the information signal, phase modulation (PM) is
produced. Frequency and phase modulation are similar and often combined and are simply called
angle modulation.

The process of impressing relatively low-frequency information signals onto a high-frequency

carrier signal is called modulation and the reverse process is called demodulation.

Analog modulation is used for the transmission of conventional analog signals, such as voice,
music, and video and not particularly useful for Data Communication systems.

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Amplitude Modulation
Amplitude modulation is the process of changing the amplitude of a relatively high frequency
carrier signal in proportion to the instantaneous value of the modulating signal (information). AM
modulators are two-input devices, one of them is a single, relatively high frequency carrier signal of
constant amplitude and the second is the relatively low-frequency information signal. The following
figure shows generation of AM waveform when a single-frequency modulating signal acts on a high
frequency carrier signal.

AM generation

Advantages of AM are simple to implement, needs a circuit with very few components and
inexpensive. The disadvantages include inefficient power usage and use of bandwidth and also prone
to noise. The total bandwidth required for AM can be determined from the bandwidth of the audio
signal: BAM = 2B

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Angle Modulation
Angle modulation results whenever the phase angle of a sinusoidal signal is varied with respect
to time and includes both FM and PM. Whenever the frequency of a carrier signal is varied, the phase
is also varied and vice versa. If the frequency of the carrier is varied directly in accordance with the
information signal, FM results, whereas if the phase is varied directly, PM results.

The above figure shows the FM and PM of a sinusoidal carrier by a single-frequency

modulating signal. Both FM and PM waveforms are identical except for their time relationship
(phase). With FM, the maximum frequency deviation occurs during the maximum positive and
negative peaks of the modulating signal. With FM, the maximum frequency deviation occurs during
the zero crossings in the modulating signal.

Comparison of FM and PM

An important feature of FM and PM is that they can provide much better protection to
the message against channel noise when compared to AM. Also because of their constant amplitude
nature, they can withstand nonlinear distortion and amplitude fading.

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Information Capacity, Bits, Bit Rate, Baud, and M-ARY Encoding:

Information capacity is a measure of how much information can be propagated through a
communication system and a function of bandwidth and transmission time. It represents the number of
independent symbols that can be carried through a system in a given unit of time. The most basic
digital symbol used to represent information is the binary digit, or bit. Bit rate is simply the number of
bits transmitted during 1 second and is expressed as bits per second (bps).

R. Hartley developed a useful relationship among bandwidth, transmission time and

information capacity called Hartley’s law given by:
IαB×t
Where, I is the information capacity in bps, B is bandwidth in hertz and t is transmission time in sec’s
Relation between information capacity of a communication channel to a bandwidth and signal-
to-noise ratio is given by Claude E. Shannon. The higher the signal-to-noise ratio, the better the
performance and also information capacity is higher. The Shannon limit of information capacity is
I = B log2 (1 + S/N ) Or I = 3.32 B log10 (1 + S/N )
Where ‘I’ is information capacity in bps, B is bandwidth in hertz and S/N is signal to noise ratio.

M-ary Encoding
M-ary is a term derived from the word binary. M simply represents a digit that corresponds to
the number of conditions, levels, or combinations possible for a given number of binary variables. For
example, a digital signal with four possible conditions is an M-ary system where M= 4 and if there are
eight possible conditions, then M= 8. The number of bits necessary to produce a given number of
conditions is expressed mathematically as:
N = log2 M or it can be written as M = 2N, where N is no of bits necessary and M is number of
conditions, levels or combinations possible with N bits. From the equation, it can be said that if there is
one bit, only 21 or two conditions are possible. For two bits 22 or four conditions are possible.

Baud and Minimum Bandwidth

Baud, like bit rate is a rate of change. Baud refers to the rate of change of the signal on the
transmission medium after encoding and modulation have occurred. Baud is the reciprocal of the time
of one output signaling element, and a signaling element may represent several information bits. Baud
is also transmitted one at a time and a baud may represent more than one information bit. So, the baud
of the Data Communication system may be considerably less than the bit rate.
According to H.Nyquist, binary digital signals can be propagated through an ideal noiseless
medium at a rate equal to twice the bandwidth of the medium. The minimum theoretical bandwidth
necessary to propagate a signal is called the minimum Nyquist bandwidth or sometimes the Nyquist
bandwidth. Using multilevel signalling, the Nyquist formulation for channel capacity is fb = B log2 M
where, fb is channel capacity in bps, B is minimum Nyquist bandwidth in hertz and M is no of discrete
signal or voltage levels. If N is substituted, we get

B = baud = fb/N, where N is number of bits encoded into each signaling element.

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Digital Modulation:
Digital modulation is the transmission of digitally modulated analog signals between two or
more points in a communications system. Analog and Digital modulation systems use analog carriers
to transport information through the system, but digital modulation uses digital modulating
(information) signal. Analog systems use analog signal only. In, v(t) = V sin(2πft + θ), if the
information signal is digital and amplitude (V) of the carrier is varied proportional to the information
signal, a digitally modulated signal called amplitude-shift keying (ASK) is produced. If the frequency
(f) is varied proportional to the information signal, frequency-shift keying (FSK) is produced and if
the phase is varied proportional to the information signal, phase-shift keying (PSK) is produced. If
both amplitude and phase are varied proportional to the information signal, quadrature amplitude
modulation (QAM) results.

Digital modulation is ideally suited to a multitude of communications applications including

both cable and wireless systems. Applications include relatively low-speed voice-band Data
Communication systems, high-speed data transmission systems, digital satellite communication
systems and personal communication systems (PCS).

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Amplitude-Shift Keying
It is the simplest digital modulation technique where a binary information signal directly
modulates the amplitude of an analog carrier. Only two output amplitudes are possible and ASK is
sometimes called as digital amplitude modulation (DAM). Amplitude shift keying is given in
mathematical terms as:

vask(t) = [ 1 + vm(t) ][ A/2 cos(ωct)]

Where vask(t) is amplitude-shift keying wave, vm(t) is digital modulation (modulating) signal in
volts, A/2 is unmodulated carrier amplitude in volts and ω c is analog carrier radian frequency in
radians per second.

In the above equation, for the modulating signal vm(t), logic 1 is represented by +1V and
logic 0 is represented by -1V. So the modulated wave vask(t) is either Acos(ωct) or 0 i.e., the carrier is
either on or off. ASK is sometimes referred as on-off keying (OOK). The rate of change of the ASK
waveform (baud) is the same as the rate of change of the binary input making bit rate equal to baud.
With ASK, the bit rate is also equal to the minimum Nyquist bandwidth.

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Frequency Shift Keying

FSK is another simple, low-performance type of digital modulation. It is similar to FM,
except the modulating signal is a binary signal varying between two discrete voltage levels. FSK is
sometimes called as binary FSK (BFSK). FSK is generally expressed as,
vfsk(t) = Vc cos{ 2π[fc + vm(t)Δf]t }
Where vfsk(t) is binary FSK waveform, Vc is peak analog carrier amplitude in volts, fc is
analog carrier center frequency in hertz, f is peak change or shift in the analog carrier frequency and
vm(t) is binary input(modulating) signal in volts. For logic 1, vm(t) = +1 and for logic 0, vm(t) = -1
reducing the equation to vfsk(t) = Vc cos {2π[fc + f]t } and vfsk(t) = Vc cos{ 2π[fc - f]t}
As the binary signal changes from a logic 0 to a logic 1 and vice versa, the output frequency
shifts between two frequencies: a mark, or logic 1 frequency (fm) and a space or logic 0 frequency
(fs). The mark and space frequencies are separated from the carrier frequency by the peak frequency
deviation ( f ) and from each other by 2 f.

With FSK, frequency deviation is defined as the difference between either the mark or space
frequency and the center frequency or half the difference between the mark and space frequencies.
Frequency deviation can be expressed as f = |fm – fs| / 2.
The baud for BFSK is determined by placing N = 1, i.e., baud = fb/1 = fb
The minimum bandwidth for FSK is determined from;
B = |(fs - fb) - (fm - fb)| = |fs - fm| + 2fb . But |fs - fm| = 2 f,
Therefore, B = 2( f + fb), where B is minimum Nyquist bandwidth in hertz and f is frequency deviation
and fb is input bit rate.
Phase-Shift Keying (PSK) is a digital modulation scheme that conveys data by changing, or
modulating, the phase of a reference signal (the carrier wave). PSK uses a finite number of phases;
each assigned a unique pattern of binary digits. Usually, each phase encodes an equal number of bits.
PSK is not susceptible to the noise degradation that affects ASK or to the bandwidth limitations of
FSK.
Binary phase-shift Keying: The simplest PSK technique is called binary phase-shift keying
(BPSK), where N = 1 and M = 2. Therefore, with BPSK two phases are possible for the carrier. It uses
two opposite signal phases (0 and 180 degrees). The digital signal is broken up time wise into
individual bits (binary digits). The state of each bit is determined according to the state of the
preceding bit. If the phase of the wave does not change, then the signal state stays the same (0 or 1). If
the phase of the wave changes by 180 degrees -- that is, if the phase reverses -- then the signal state
changes (from 0 to 1 or from 1 to 0). Because there are two possible wave phases, BPSK is sometimes
called biphase modulation or phase-reversal keying (PRK).

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More sophisticated forms of PSK exist. In M-ary or multiple phase-shift keying (MPSK), there
are more than two phases, usually four (0, +90, -90, and 180 degrees) or eight (0, +45, -45, +90, -90,
+135, -135, and 180 degrees). If there are four phases (m = 4), the MPSK mode is called quadrature
phase-shift keying or quaternary phase-shift keying (QPSK), and each phase shift represents two
signal elements. If there are eight phases (m = 8), the MPSK mode is known as octal phase-shift
keying (OPSK), and each phase shift represents three signal elements. In MPSK, Data Can be
transmitted at a faster rate, relative to the number of phase changes per unit time, than is the case in
BPSK.

QPSK is an M-ary encoding scheme where N = 2 and M = 4, which has four output phases are
possible for a single carrier frequency needing four different input conditions. With two bits, there
are four possible conditions: 00, 01, 10, and 11. With QPSK, the binary input data are combined into
groups of two bits called dibits.

The above figure shows the output phase-versus-time relationship, truth table, and constellation
diagram for QPSK. A phase of 00 now represents 00; 900 represents 01; 1800 represents 10; and 270 0
represents 11. Data Can be transmitted twice as efficiently using 4-PSK than 2-PSK.

With 8-PSK, three bits are encoded forming tribits and producing eight different output phases.
With 8-PSK, N = 3, M = 8, and the minimum bandwidth and baud equal one third the bit rate (fb /3).
8-PSK is 3 times as efficient as 2-PSK.

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With 16-PSK, four bits called quadbits are combined, producing 16 different outputs phases. With
16-PSK, N = 4, M = 16, and the minimum bandwidth and baud equal one-fourth the bit rate (fb /4).

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Modulation Bit Rate Encoding Bandwidth Outputs Minimum Baud

Scheme Efficiency Possible Bandwidth

ASK N Single bit 1 2 fa f
b b
FSK N Single bit 1 2 >fb fb
f f
BPSK N Single bit 1 2 b b
QPSK 2N Dibits 2 4 fb /2 fb /2
8-PSK 3N Tribits 3 8 fb /3 fb /3
16-PSK 4N Quadibits 4 16 fb /4 fb /4

Quadrature Amplitude Modulation (QAM)

PSK is limited by the ability of the equipment to distinguish small differences in phase.
Bandwidth limitations make combinations of FSK with other changes practically useless. Quadrature
amplitude modulation is a combination of ASK and PSK so that a maximum contrast between each
signal unit (bit, dibit, tribit, and so on) is achieved. QAM is used extensively as a modulation scheme
for digital telecommunication systems. The primary advantage of QAM over PSK is immunity to
transmission impairments, especially phase impairments that are inherent in all communication
systems.
In 4-QAM and 8-QAM, number of amplitude shifts is fewer than the number of phase shifts.
Because amplitude changes are susceptible to noise and require greater shift differences than do
phase changes, the number of phase shifts used by a QAM system is always larger than the number
of amplitude shifts.

With 16-QAM, there are 12 phases and three amplitudes that are combined to produce 16
different output conditions. With QAM, there are always more phases possible than amplitude.

Bandwidth Efficiency
Bandwidth efficiency is often used to compare the performance of one digital modulation
technique to another. It is the ration of transmission bit rate to the minimum bandwidth required for a
particular modulation scheme. Mathematically represented as:

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Bη = transmission bit rate (bps) / minimum bandwidth (Hz)

ASK, FSK, PSK, and QAM Summary

Trellis Code Modulation

Data Transmission Rates in excess of 56 kbps can be achieved over standard telephone circuits
using an encoding scheme called trellis code modulation(TCM) developed by Dr. Ungerboeck. It
combines encoding and modulation to reduce the probability of error, thus improving the bit error
performance and it uses conventional (tree) codes.

Trellis coding defines the manner in which signal-state transitions are allowed to occur, and
transitions that do not follow this pattern are interpreted as transmission errors. TCM can improve
error performance by restricting the manner in which signals are allowed to transition. TCM
improves on standard QAM by increasing the distance between symbols on the constellation (called
Euclidean distance).

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Appendix (some additional figures):

8-QAM modulator: (a) truth table; (b) phasor diagram; (c) constellation diagram

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Assignment Questions
1. In QAM amplitude and phase of the transmitted signal are varied - Justify your answer with a
block diagram and constellation diagram.

2. a. Explain the importance of asynchronous transmission in communication.

b. Write the comparison between asynchronous and synchronous data transmission

3. (a) What is topology? Explain topologies in Data Communications?

(b) What are the various types of transmission modes and explain.

4. (a) What is Data Communications? Explain briefly Data Communication circuit.

(b) Mention some standard organizations for Data Communications?

5. Draw OS I architectural model for open system inter networking and explain.

6. (a) Explain about Analog data, Digital signal encoding technique.

(b) Differentiate between Data and Signals?

7. Sketch the binary ASK, FSK, PSK, and QPSK waveform for the following sequence 1011.
a) Explain the relationship between bits per second and baud for an FSK system.
b) Determine the bandwidth and baud for an FSK signal with a mark frequency of 24 kHz
and a bit rate of 4 kbps.
c) Explain the relationship between
i) Minimum bandwidth required for an FSK system and the bit rate
ii) Mark and space frequencies

8. a) What is a constellation diagram? How it is used with PSK?

b) Explain the minimum bandwidth required for a BPSK system and the bit rate.
c) Explain M-ary

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Module – II
METALLIC CABLE TRANSMISSION MEDIA:
Metallic Transmission Lines, Transverse Electromagnetic Waves
Characteristics of Electromagnetic Waves
Transmission Line Classifications
Metallic Transmission Line Types
Metallic Transmission Line Equivalent Circuit
Wave Propagation on Metallic Transmission Lines
Metallic Transmission Line Losses

OPTICAL FIBER TRANSMISSION MEDIA:

Advantages of Optical Fiber Cables
Disadvantages of Optical Fiber Cables
Electromagnetic spectrum
Optical Fiber Communications System Block Diagram
Optical Fiber construction
The Physics of Light
Velocity of Propagation
Propagation of Light Through an Optical fiber Cable
Optical Fiber Modes and Classifications
Optical Fiber Comparison
Losses in Optical Fiber Cables
Light sources
Light Detectors
Lasers

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METALLIC CABLE TRANSMISSION MEDIA

Introduction
The transmission medium is the physical path between transmitter and receiver in a data
transmission system. It is included in the physical layer of the OSI protocol hierarchy. The
transmission medium is usually free space, metallic cable, or fiber-optic cable. The information is
usually a signal that is the result of a conversion of data from another form.

Transmission media can be generally categorized as either unguided or guided. Guided

Transmission Media uses a "cabling" system (or some sort of conductor) 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. The conductor directs the signal propagating down it. Only
devices physically connected to the medium can receive signals propagating down a guided
transmission medium. Examples of guided transmission media are copper wire and optical fiber.

Unguided Transmission Media consists of a means for the data signals to travel but
nothing to guide them along a specific path. The data signals are not bound to a cabling media and
as such are often called Unbound Media. Unguided transmission media are wireless systems.
Signals propagating down an unguided transmission medium are available to anyone who has a
device capable of receiving them.

A physical facility is one that occupies space and has weight as opposed to wireless media
such as earth’s atmosphere or a vacuum and includes metallic cables and optical cables. Metallic
transmission lines includes open-wire, twin-lead, and twisted-pair copper wire as well as coaxial
cable, and optical fibers include plastic- and glass-core fibers encapsulated in various kinds of
cladding materials.

Metallic Transmission Lines

A transmission line is a metallic conductor system used to transfer electrical energy from
one point to another using electrical current flow. It is two or more electrical conductors separated
by a nonconductive insulator (dielectric). It can be of varied lengths varying from few inches to
several thousand miles. It can be used to propagate dc or low-frequency ac and also very high
frequencies such as microwave radio-frequency signals.

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Transverse Electromagnetic Waves

The two basic kinds of waves are longitudinal and transverse. With longitudinal waves,
the displacement is in the direction of propagation. A surface wave or sound waves can be said as
examples of longitudinal waves. With transverse waves, the direction or displacement is
perpendicular to the direction of propagation. Electromagnetic waves are transverse waves.

Propagation of electrical power along a transmission line occurs in the form of transverse
electromagnetic (TEM) waves. TEM wave propagates primarily in the non-conductor that
separates the two conductors of the transmission line. The electric field (E) and magnetic field (H)
are perpendicular to each other at all points. This is referred to as space or quadrature.
Electromagnetic waves that travel along a transmission line from the source to the load are called
incident waves and those that travel from the load back towards the source are called reflected
waves.

Characteristics of Electromagnetic waves

The three main characteristics are wave velocity, frequency and wavelength.

Wave velocity: Waves travel at different speeds depending on the type of wave and the
characteristics of the propagation medium. Sound travels at 1100 feet/second in normal
atmosphere where electromagnetic waves travel much faster. In free space i.e. in vacuum, TEM
waves travel at the speed of the light, c (approximately at 186,000 miles/sec) and slightly slower
in air and considerably slower along a transmission line.

Frequency and Wavelength: The oscillations of an electromagnetic wave are periodic and
repetitive. The rate at which the periodic wave repeats is its frequency. The distance of one cycle
occurring in space is called the wavelength and is given by

Distance = velocity × time

If the time for one cycle is substituted above, we get the length of one cycle which is called
wavelength and is given by
λ = velocity × period = v × T, where λ is wavelength, v is velocity and T is period
because T = 1/f, we can write λ = v/f

As for free space propagation, v = c; the length of one cycle is λ = c/f = 3×108 m/s/f cycles/s

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Transmission line classifications

Balanced Transmission Line

In two wire balanced lines, both conductors carry current. One conductor carries the signal
and the other conductor in the return path. This type of transmission is called differential or
balanced signal transmission. Both conductors in a balanced line carry signal currents, which are
equal in magnitude with respect to electrical ground but travel in opposite directions.
Currents that flow in opposite directions in a balanced wire pair are called metallic circuit
currents and currents that flow in same direction are called longitudinal currents. The chief
advantage of the balanced line format is good rejection of external noise. Common forms of
balanced line are twin-lead, used for radio frequency signals and twisted pair, used for lower
frequencies.

Unbalanced Transmission Line

With an unbalanced transmission line, one wire is at ground potential, whereas the other
wire is at signal potential. This type of transmission line is called single-ended or unbalanced
signal transmission. The ground wire may also be the reference for other signal-carrying wires
and must go anywhere any of the signal wires go.
Unbalanced transmission lines have the advantage of requiring only one wire for each
signal and only one ground line is required no matter how many signals are grouped into one
conductor. Balanced transmission lines can be connected to unbalanced transmission lines and
vice versa with special transformers called balums.

Metallic Transmission Line Types

All Data Communication systems and computer networks are interconnected to some
degree with cables, which form the most important part of the transmission medium transporting
signals between computers.

Parallel-Conductor Transmission Lines

Parallel-wire transmission lines are comprised of two or more metallic conductors
separated by a nonconductive insulating material called a dielectric. Common dielectric materials
include air, rubber, polyethylene, paper, mica, glass and Teflon. The most common parallel-
conductor transmission lines are open-wire, twin lead and twisted pair, including unshielded
twisted pair (UTP) and shielded twisted pair (STP).
Open-Wire Transmission Lines: These are two-wire parallel conductors, closely spaced and
separated by air. Non conductive spacers are placed at periodic intervals not only for support but
also to keep the distance between the conductors constant. TEM wave propagates in the air
between the conductors, which acts as dielectric. The main advantage is its simple construction.

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Since no shielding is present, the radiation losses are high and cable is susceptible to
picking up signals through mutual induction, which produces crosstalk. The primary usage is in
standard voice-grade telephone applications.

Twin lead: Twin-lead is essentially the same as open-wire transmission line except that the
spacers between the two conductors are replaced with a continuous solid dielectric ensuring the
uniform spacing along the entire cable.

It is mainly used to connect televisions to rooftop antennas. Common dielectric materials

used with twin-lead cable are Teflon and polyethylene.

Twisted-pair transmission lines:

A twisted-pair (TP) transmission line is formed by twisting two insulated conductors
around each other. Usually, a number of pairs of these wires are put together into a cable. The
cable may contain more than a hundred pairs of wires for long-distance communications. Twisted-
pair wires are the most common media in a telephone network. These wires support both analog
and digital signals and can transmit the signal at a speed of 10 Mbps over a short distance. The
twisting of wires with different twisting lengths reduces the effect of cross talk and low-frequency
interference.

Twisted-pair cable

Twisted-pair transmission lines are also the transmission medium of choice for most
local area networks because twisted-pair cable is simple to install and relatively independent
when compared to coaxial and optical fiber cables.
The two basic types of twisted-pair transmission lines specified are unshielded twisted
pair (UTP) and shielded twisted pair (STP).

Unshielded twisted-pair: An UTP cable consists of two copper wires where each wire is
separately encapsulated in PVC (polyvinyl chloride) insulation. Bandwidth can be improved by
controlling the number of twists per foot and also the manner in which multiple pairs are twisted
around each other. The minimum number of twists for UTP cable is two per foot.

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UTPs are cheaper, more flexible, and easier to install. They provide enough support for
telephone systems and are not covered by metal insulation. They offer acceptable performance for
a long-distance signal transmission, but as they are uninsulated, they are affected by cross talk,
atmospheric conditions, electromagnetic interference, and adjacent twisted pairs, as well as by any
noise generated nearby. The majority of the telephone twisted pairs are unshielded and can
transmit signals at a speed of 10 Mbps.
The Electronic Industries Association (EIA) has developed standard to grade UTP cable
by quality; Category 1 as the lowest quality and category 6 as the highest quality.
1. Category 1: The basic twisted-pair cabling used in telephone systems. This level of
quality is fine for voice but inadequate for data transmission.
2. Category 2: This category is suitable for voice and data transmission of up to 2Mbps.
3. Category 3: This category is suitable for data transmission of up to 10 Mbps. It is now
the standard cable for most telephone systems. At least three twist per feet
4. Category 4: This category is suitable for data transmission of up to 20 Mbps.
5. Category 5: This category is suitable for data transmission of up to 100 Mbps.
6. Category 6: CAT- 6 is recently proposed cable type comprised of four pairs of wire
capable of operating at transmission rates of up to 400Mbps.
Advantages of UTP are its easy to terminate, installation costs are less and more lines can
be run through the same wiring ducts. Disadvantages of UTP are its a bit noisy and prone to
interference.
Shielded Twisted Pair (STP) Cable:
STP cable is a parallel two-wire transmission line consisting of two copper conductors
separated by a solid dielectric material. The wires and dielectric are enclosed in a conductive-
metal sleeve called a foil. If the sleeve is woven into a mesh, it’s called braid. The metal casing
prevents the penetration of electromagnetic noise. Materials and manufacturing requirements
make STP more expensive than UTP but less susceptible to noise.

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Plenum Cable:
Plenum cables are the electrical or telecommunication cables (or wires) which are
installed in environmental air spaces in the interior of many commercial and residential buildings.
It is common practice to route communication cables and the like for computers, data devices, and
alarm systems through plenums in building constructions. If a fire occurs in a building which
includes plenums or risers, the non-fire retardant plenum construction would enable the fire to
spread very rapidly throughout the entire building. Typically plenum Data Cables have two or
more pairs of insulated conductors in a common jacket. The insulation can be made of several
types of flame retardant insulation. A plenum is defined as a compartment or chamber to which
one or more air ducts are connected and which forms part of the air distribution system of the
structure. Plenum cables have a plurality of twisted pair conductors surrounded by a jacket. The
twisted pairs generally all have the same twist or substantially the same twist. A typical and
widely used flame retardant insulation for conductors in data plenum cables is fluorinated
ethylene-propylene. Category 5 plenum cable made of jacketed twisted pairs of insulated
conductors has to satisfy a number of electrical requirements set by the EIA/TIA specification
568A.

Coaxial (Concentric) Transmission Lines

Because of the advent of modern UTP and STP twisted pair cables, coaxial cable is seen
very less in computer networks, but still has very high importance in analog systems, such as
cable television distribution networks. The basic coaxial cable consists of a center conductor
surrounded by a dielectric material (insulation), then a concentric (uniform distance from the
center) shielding, and finally a rubber environmental protection outer jacket. A coaxial cable with
one layer of foil insulation and one layer of braided shielding is referred to as dual shielded and if
two layers of foil insulation and two layers of braided metal shielding are present, it’s called quad
shielding.

Two basic types of coaxial cables are present: rigid air filled and solid flexible. Rigid air-
filled cables are relatively expensive and are tough to maintain. Coaxial cables are capable of
operating at higher bit rates than their parallel-wire counterparts, very secure than twisted-pair
cable, can be used over long distances, immune to external radiation and radiate little themselves.
Disadvantages of coaxial transmission lines are their poor cost-to-performance ratio, low
reliability, and high maintenance.
The RG numbering system used with coaxial cables refers to cables approved by U.S. Department
of Defense (DoD).

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Categories of Coaxial Cables

To connect coaxial cable to devices, it is necessary to use coaxial connectors. The most
common type of connector is the Bayone-Neill-Concelman, or BNC, connectors. BNC connectors
are sometimes referred to as bayonet mount, as they can be easily twisted on or off.

There are three types: the BNC connector, the BNC T connector, the BNC terminator.
Applications include cable TV networks, and some traditional Ethernet LANs like 10Base-2, or
10-Base5.

Metallic Transmission Line Equivalent Circuit

The characteristics of a transmission line are determined by its electrical properties like
wire conductivity, insulator dielectric constant and its physical properties like wire diameter and
conductor spacing. These properties in turn determine the primary electric constants: series
resistance (R), series inductance (L), shunt capacitance (C), and shunt conductance (G).
Resistance and inductance occur along the line, whereas capacitance and conductance occur
between the conductors.

Characteristic Impedance
For maximum power transfer from the source to load, a transmission line must be
terminated in a purely resistive load equal to the characteristic impedance of the transmission line.
Transmission line stores energy in its distributed inductance and capacitance.
Using Ohm’s law, the characteristic impedance is simply the ratio of the source voltage
(Eo) to the line current (Io), given by
Zo = Eo / I o
Where, Zo is characteristic impedance in ohms, Eo is source voltage in volts and Io is
transmission line current in amps.
Characteristic impedance of a two wire parallel transmission line with an air dielectric
can be determined from its physical dimensions Zo = 276 log D/r where D is the distance between
the centres of the two conductors and R is radius of the conductors. Characteristic impedance of a
coaxial cable can also be determined from its physical dimensions:

Where, D is inside diameter of the conductor and Ɛr is relative dielectric constant of the insulating
material.

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Wave Propagation on Metallic Transmission Lines

EM waves travel at the speed of light through vacuum and nearly the same through
air, but they travel considerably slowly in metallic transmission lines, where the conductor is
generally copper and the dielectric materials vary with cable type.

Velocity Factor and Dielectric Constant

Velocity factor is defined as the ratio of the actual velocity of propagation of an electromagnetic
wave through a given medium to the velocity of propagation through a vacuum. Mathematically,
given as:

Vf = Vp / c, where Vf is velocity factor, Vp is actual velocity of propagation and c is velocity of

propagation through a vacuum (3×108 m/s).

Dielectric constant is simply the relative permittivity of a material. The dielectric constant
depends on the type of insulating material used. The velocity at which an EM wave propagates
along a transmission line varies with the inductance and capacitance of the cable. Time can be
given as: T = √LC. Inductance, capacitance ad velocity of propagation can be given by the
formula,
velocity × Time = Distance
Therefore, Vp = Distance/ Time = D/T which can be written as Vp = D/ √LC

If the distance is normalized to 1 meter, the velocity of propagation for a lossless transmission line
is Vp = 1 / √LC

Metallic Transmission Line Losses

Signal power is lost in a transmission line through different ways: conductor loss,
radiation loss, dielectric heating loss, coupling loss and corona. All these losses are lumped
together and are specified as attenuation loss in decibels per unit length.

Conductor Losses:
As electrical current flows through a metallic transmission line, there is an inherent and
unavoidable power loss because of the finite resistance present in the line. This loss is termed as
conductor loss or conductor heating loss and is simply I 2r power loss.

Radiation Losses:
Radiation and Induction losses are similar in that both are caused by the fields
surrounding the conductors. Induction losses occur when the electromagnetic field about a
conductor cuts through any nearby metallic object and a current is induced in that object.
Radiation losses are reduced by properly shielding the cable. Therefore, STP and coaxial cables
have less radiation than UTP, twin lead and open wire.

Coupling Losses:
Coupling loss occurs whenever a connection is made to or from a transmission line or
when two sections of transmission line are connected together. Discontinuities are the locations
where dissimilar materials meet and they tend to heat up, radiate energy, and dissipate power.

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Corona:
Corona is a luminous discharge that occurs between the two conductors of a transmission
line, when the difference of potential between them exceeds the breakdown voltage of the
dielectric insulator. When corona occurs, the transmission line is destroyed.

OPTICAL FIBER TRANSMISSION MEDIA:

An optical communications system is one that uses light as the carrier of information.
They use glass or plastic fiber cables to contain the light waves and guide them in a manner
similar to the way EM waves are guided through a metallic transmission media.

Advantages of Optical Fiber Cables

Wider bandwidth and greater information capacity: The light wave occupies the frequency
range between 2×1012 Hz to 37×1012 Hz. This makes the information carrying capability of
fiber optic cables is much higher.
Immunity to crosstalk: Since fiber optic cables use glass and plastic fibers, which are non-
conductors of electrical current, no magnetic field is present. No magnetic induction means
no crosstalk.
Immunity to static interference: As optical fiber cables are non-conductors, they are immune
to electromagnetic interference (EMI) caused by lightning, electric motors, relays,
fluorescent lights and other electrical noise sources.
Environmental immunity: Optical fibers are more immune to environmental extremes. They
can operate over large temperature variations and are also not affected by corrosive liquids
and gases.
Safety and convenience: As only glass and plastic fibers are present, no electrical currents or
voltages are associated with them. Also they can be used around any volatile liquids and
gasses without worrying about their causing explosions or fires.
Lower transmission loss: Fiber optic cables offers less signal attenuation over long
distances. Typically, it is less than 1 dB/km
Security: Optical fibers are more secure as they are almost impossible to tap into because
they do not radiate signals. No ground loops exist between optical fibers hence they are
more secure.
Durability and reliability: Optical cables last longer and are more reliable than metallic
facilities because fiber cables have a higher tolerance to changes in environmental
conditions and are immune to corrosive materials.
Economics: Cost of optical fiber cables is same as metallic cables. Fiber cables have less
loss and require fewer repeaters, which in turn needs lower installation and overall system
costs.

Disadvantages of Optical Fiber Cables

Interfacing costs: As optical cables need to be connected standard electronic facilities
requiring expensive interfaces
Strength: Optical cables have lower tensile strength than coaxial cable. They need an extra
coating of Kevlar and also a protective jacket of PVC. Glass fiber is also fragile making
them less attractive in case of hardware portability is required.

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Remote electrical power: Occasionally, electrical power needs to be provided to remote

interfaces, which cannot be accomplished using optical cables.
Losses through bending: Bending the cable causes irregularities in the cable dimensions,
resulting in loss of signal power. Also, optical cables are prone to manufacture defects
causing an excessive loss of signal power.
Specialized tools, equipment and training: Special tools are required to splice and repair
cables and special test equipment are needed to make routine measurements. Technicians
working on optical cables need special skills and training.

Electromagnetic Spectrum

The electromagnetic spectrum is the range of all possible frequencies of electromagnetic

radiation. The "electromagnetic spectrum" of an object is the characteristic distribution of
electromagnetic radiation emitted or absorbed by that particular object. The frequency spectrum
extends from the subsonic frequencies (a few hertz) to cosmic rays (10 23 Hz). The light frequency
spectrum can be divided into three general bands.
1. Infrared: The band of frequencies that is too high to be seen by the human eye with
wavelengths ranging between 770nm and 106 nm. Optical fibers generally operate in infrared
band.
2. Visible: The band of light frequencies to which the human eye will respond with wave
lengths ranging between 390nm and 770nm. This band is visible to human eye.
3. Ultraviolet: The band of light frequencies, that is too low to be seen by the human eye
with wave lengths ranging between 10nm and 390nm.

Optical Fiber Communications System Block Diagram

The three primary building blocks are transmitter, receiver and the optical fiber cable.
The transmitter is comprised of a voltage-to-current converter, a light source, and source-to-fiber
interface. The fiber guide is the transmission medium, which is either an ultrapure glass or a
plastic cable. The receiver includes a fiber-to-interface, a photo detector, and a current-to-voltage
converter.

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Optical Fiber Construction

There are two basic types of fiber-optic cable. The difference is whether the fiber is free
to move inside a tube with a diameter much larger than the fiber or is inside a relatively tight-
fitting jacket. They are referred to as loose-tube and tight-buffer cables.
Both methods of construction have advantages.
• Loose-tube cables - all the stress of cable pulling is taken up by the cable’s strength
members and the fiber is free to expand and contract with temperature.
• Tight-buffer cables are cheaper and generally easier to use

Physics of Light
Albert Einstein and Max Planck showed that when light is emitted or absorbed, it behaves
like an electromagnetic wave and also like a particle called a photon, which possesses energy
proportional to its frequency. This is known as Planck’s Law. It states that “when visible light or
high-frequency electromagnetic radiation illuminates a metallic surface, electrons are emitted”. It
is expressed mathematically as, Ep = hf,
where Ep is energy of the photons in joules, h is Planck’s constant and f is frequency of
light
The process of decaying from one energy level to another energy level is called
spontaneous decay or spontaneous emission. The process of moving from one energy level to
another is called absorption.
Optical power measures the rate at which electromagnetic waves transfer light energy. It is
described as the flow of light energy past a given point in a specified time. Expressed
mathematically as,
P = d (energy)/ d(time) = dQ/ dt ,
where P is optical power in watts and dQ is instantaneous charge in joules

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Velocity of Propagation:
Refraction: Refraction is the bending of light when the light passes from one medium to another.
The angle between the light ray and the normal as it leaves a medium is called the angle of
incidence. The angle between the light ray and the normal as it enters a medium is called the
angle of refraction.

When an electromagnetic wave is reduced as it passes from one medium to another

medium of denser material, the light ray changes direction or refracts (bends) toward the normal.
When an electromagnetic wave passes from a more dense material into a less dense material, the
light ray is refracted away from the normal. The normal is simply an imaginary line drawn
perpendicular to the interface of the two materials at the point of incidence

Refractive Index: Refractive index is simply the ratio of the velocity of propagation of light ray
in free space to the velocity of propagation of a light ray in a given material. Given by,: n = c/v,
where n is refractive index and c is speed of light (m/sec) and v is speed of light in a
given material (m/sec). Typical indexes of refraction of some materials are given below:

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Snell’s Law: This relationship between the angles of incidence and refraction and the indices
of refraction of the two medium is known as Snell's Law. Snell's law applies to the refraction
of light in any situation, regardless of what the two media are.

Refractive model for Snell’s Law

Snell’s Law is stated mathematically as:
n1 sinθ1 = n2 sinθ2
Where, n1 is refractive index of material 1, n2 is refractive index of material2, θ1 is angle of
incidence and θ2 is angle of refraction.

Critical Angle: The angle of incidence is called the critical angle (θ c), which is defined as the
minimum angle of incidence at which a light ray may strike the interface of two media and result
in an angle of refraction of 90 degrees or greater. Light ray has to travel from medium of higher
refractive index to that of lower refractive index. Expressed as:
θc = sin-1 n2/n1
Acceptance angle, acceptance cone and numerical aperture: For a ray of light to
propagate down the cable, it must strike the internal core/cladding interface at an angle that is
greater than the critical angle.
θin (max) = sin -1 √(n12 – n22)
Where, θin(max) is acceptance angle or acceptance cone half angle. It defines the maximum angle
in which external light rays may strike the air/glass interface and still propagate down the fiber.
Rotating the acceptance angle around the fiber axis, a cone pattern is obtained, called as
acceptance cone of the fiber input. The cone of acceptance is the angle within which the light is
accepted into the core and is able to travel along the fiber. Launching light wave will be easier for
large acceptance cone.

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Numerical Aperture (NA) is used to describe the light-gathering or light-collecting

ability of an optical fiber. Larger the magnitude of NA, greater the amount of external light the
fiber will accept. Described as, NA = sin θin and NA = √ (n12 – n22). Therefore, it can be
written:
θin = sin -1NA

Propagation of Light through an Optical Fiber Cable:

Light can be propagated using either refraction or reflection and the way light propagates
depends on the mode of propagation and the index profile of the fiber.

Modes of propagation:
Mode simply means path. If there is only one path for light rays to take down a cable, it
is called single mode and if there is more than one path, it is called multimode. In single mode, the
light travels directly down the center of the cable, whereas for multimode, light rays propagate
down the cable in a zigzagging fashion following several paths. The number of modes possible for
a given cable can be given by:
N = [πd/λ√ (n12 – n22)]
Where N is number of modes, d is core diameter and λ is wave length and n1 is refractive index
of core and n2 is refractive index of cladding.

Index Profile

Index profile of an optical fiber is a graphical representation of the magnitude of the

refractive index across the fiber. The above figure shows the index profiles of three types of
fibers. Two basic types of index profiles are present. A step-index fiber has a central core with a
uniform refractive index. A graded-index fiber has no cladding and the refractive index of the
core is non-uniform. It is highest at the center of the core and decreases gradually with distance
towards the outer edge.

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Optical Fiber modes and Classifications

Three practical types of optical fiber configurations: single-mode step index, multimode
step index and multimode graded index.

Single-Mode Step-Index Optical Fiber:

The fiber has a central core that is sufficiently small that there is essentially only one path
for light ray through the cable. In most cases, the outside cladding is air making this fiber to have
a wide external acceptance angle making it relatively easy to couple to a light source. But, this
type of fiber is very weak and difficult to splice or terminate. A more practical approach will be
single mode step-index fiber that has a cladding other than air. This would be physically stronger
than air-clad fiber but critical angle will be higher resulting in a small acceptance angle. This
makes it difficult to couple light into the fiber from a light source.
Advantages:

Minimum dispersion: all rays take same path, same time to travel down the cable.
A pulse can be reproduced at the receiver very accurately.
Less attenuation can run over longer distance without repeaters.
Larger bandwidth and higher information rate
Difficult to couple light in and out of the tiny core
Highly directive light source (laser) is required.
Interfacing modules are more expensive

Multimode Step-Index Optical Fiber: These are similar to single mode step-index fibers except
that the center core is much larger with the multimode configuration. This type has a large light-
to-fiber aperture and therefore allows more external light to enter the cable. Light rays travel
down the cable in a zigzag fashion continuously reflecting off the interface boundary. Light rays

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travel in many paths as it propagates down the fiber. So, all light rays do not follow the same path
and do not take same amount of time to travel the length of the cable.

Advantages
These are relatively expensive and simple to manufacture
It is easier to couple light into and out of multimode step-index fiber as they have a
relatively large source-to-fiber aperture.

Disadvantages
As light rays travel in different paths, large difference in propagation times results. So,
the rays travelling down have a tendency to spread out. Consequently the pulse of light
propagating down is more distorted than other types of fibers.
Less bandwidths and lower rate of information transfer rates when compared to other
types.

Multimode Graded-Index Optical Fiber: These fibers are characterized by a central core with a
non-uniform refractive index. Cables density is maximum at centre and decreases gradually
towards the edge. Light ray is propagated through refraction. As the light propagates across the
core toward the center it intersects a less dense to more dense medium. Consequently, light rays
constantly being refracted resulting in continuous bending of light rays. The light rays take
approximately the same amount of time to travel the length of the fiber. This cable is mostly used
for long distance communication.

Losses in Optical Fiber Cables

Power loss in optical fiber cables is often called attenuation and results in reduction of power of
light wave as it travels down the cable. Generally, total power loss is expressed as:
A(dB) = 10 log (Pout / Pin) where A(dB) is total reduction in power level, attenuation and Pout is
cable output power and Pin is cable input power. Multimode fibers tend to have more attenuation
than single-mode cables because of increased scattering of light wave.
Transmission losses in optical fibers result in reduction in light power, thus reducing the system
bandwidth, information transmission rate, efficiency, and overall system capacity. The
predominant losses are:

Absorption Losses: It is analogous to power dissipation in copper cables as impurities in the

fiber absorb the light and convert it to heat. Three main factors contribute to absorption losses.

Ultraviolet absorption:- Caused by valence electrons in the silica material from which
fibers are manufactured.
Infrared absorption: - Result of photons of light that are absorbed by the atoms of the glass
core molecules.
Ion resonance absorption: - Caused by OH- ion in the material. Iron, copper and chromium
molecules also cause ion absorption.

Material or Rayleigh Scattering Losses: Rayleigh scattering of light is due to small

localized changes in the refractive index of the core and cladding material. Two main causes for
this:

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The first is due to slight fluctuation in mixing of ingredients. The random changes because
of this are impossible to eliminate completely.
The other cause is slight change in density as the silica cools and solidifies. When light ray
strikes such zones, it gets scattered in all directions. The amount of scatter depends on the
size of the discontinuity compared with the wavelength of the light.
So the shortest wavelength suffers most scattering.

Chromatic Distortion or Wavelength Dispersion: Light rays that are simultaneously

emitted from an LED and propagated down an optical fiber do not arrive at the far end of the fiber
at the same time, which results in an impairment called chromatic distortion. It occurs in only in
fibers with a single mode of transmission and can be eliminated using monochromatic light
sources like injection laser diode (ILD).

Radiation Losses: These are caused predominantly by small bends wand kinks in the fiber. The
two types of bends are: microbends and constant-radius bends. Micro bending occurs as result of
differences in the thermal contraction rates between core and cladding material and results in a
material bend along the axis of the fiber and represents a discontinuity where Rayleigh scattering
occurs. Constant-radius bends are caused by excessive pressure and tension and generally occur
when fibers are bent during installation.

Modal dispersion: Modal dispersion (called pulse spreading) is caused by the difference in the
propagation times of light rays that take different paths down a fiber and occurs only in
multimode fibers. It can be reduced considerably by using graded index fibers and almost entirely
eliminated using single-mode step-index fibers. If three rays of light are emitted into the fiber at
the same time, each ray would reach the far end at a different time resulting in a spreading out of
light energy with respect to time. This is called modal dispersion.

Coupling Losses: These losses are caused by imperfect physical connections. These occur at
three types of junctions: light source-to-fiber connections, fiber-to-fiber connections, and fiber-to-
photo detector connections. They are caused by one of the following alignment problems:

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Lateral displacement: It is the lateral or axial displacement between two pieces of adjoining
fiber
Gap displacement (misalignment): When splices are made in optical fibers, the fibers should
actually touch. The farther apart the fibers are, the greater the loss of light.
Angular displacement: It is sometimes called angular displacement and if it is less than 2
degrees, the oss will typically be less than 0.5 dB.
Imperfect surface finish: The ends of two adjoining fibers should be highly polished and fit
together squarely. If the fiber ends are less than 3 degrees off from perpendicular, the losses
will typically be less than 0.5 dB.

Light Sources
Light sources are used in fiber optic communication to generate light pulses at
wavelengths efficiently propagated by the optical fiber. They also should produce sufficient
power to allow the light to propagate through the fiber without causing distortion in the cable or
receiver. Two types of practical light sources used to generate light for optical fiber
communications systems: light-emitting diodes (LED’s) and injection laser diodes (ILD’s).

Light Emitting Diodes:

A LED is a p-n junction diode, usually made from a semiconductor material such as
aluminum-gallium-arsenide (AlGaAs) or gallium-arsenide-phosphide (GaAsP). LED’s emit light
by spontaneous emission-light is emitted as a result of the recombination of electrons and holes.
LEDs can provide light output when forward biased. The LED has a low output power, slower
switching speed and greater spectral width, hence more dispersion. These deficiencies make it not
useful for high speed and long distance communication. The output of LED is non-coherent and
coupling efficiency is very low.

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Injection Laser Diode:

ILD’s are similar to LED’s and they act similarly below a certain threshold current.
Above the threshold current, and ILD oscillates and lasing occurs. As current passes through a
forward biased p-n junction diode, light is emitted by spontaneous emission at a frequency
determined by the energy gap of the semiconductor material. The radiant output power
of ILD is more directive than LED. After lasing occurs, the optical power increases dramatically,
with small increases in drive current.

Advantage:
1. ILD’s emit coherent (orderly) light compared to incoherent (disorderly) light emitted by
LED. So ILD have a more direct radian pattern, making it easier to couple light emitted by
the ILD into an optical fiber cable. Coupling losses are reduced and also small fibers can be
used.
2. The radiant output power of ILD is greater than that for an LED. Typically the output
power for an ILD is 5 mW and only 0.5mW for LED. This allows ILD’s to provide a higher
drive power and can be used for operation over longer distances.
3. ILD’s can be used at higher bit rates than LED’s

4. ILD’s generate monochromatic light, which reduces chromatic or wavelength

dispersion

Disadvantages:
1. ILD’s are typically 10 times more expensive than LED’s
2. As ILD’s operate at higher powers, they have a short lifetime
3. ILD’s are more temperature dependent than LED’s

Light Detectors
Two devices are commonly used to detect light energy in optical fiber communications
receivers: PIN (p-type-intrinsic –n-type) diodes and APD (avalanche photodiodes). PIN diodes are
the most common device used and operate just the opposite of an LED. APD’s are more sensitive
than pin diodes and require less additional amplification. The disadvantages of APD’s are
relatively long transmit times and additional internally generated noise due to avalanche
multiplication factor.

Characteristics of light detectors

1. Responsivity: A measure of the conversion efficiency of a photodetector. It is the ratio of
the output current of a photodiode to the input optical power and has the unit of amperes/watt.
2. Dark Current: The leakage current that flows through a photodiode with no light input.

3. Transit time: The time it takes a light-induced carrier to travel across the depletion region
of a semiconductor. Determines the maximum bit-rate possible
4. Spectral Response: The range of wavelength values that a given photodiode will respond to.
5. Light Sensitivity: The minimum optical power a light detector can receive and still
produce a usable electrical output signal.

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Lasers
Laser stands for light amplification stimulated by the emission of radiation. It deals
with the concentration of light into a very small, powerful beam. There are four types of lasers:
1. Gas lasers: Gas lasers use a mixture of helium and neon enclosed in a glass tube. A flow
of coherent light waves is emitted when an electric current is discharged into the gas. The
continuous light-wave output in monochromatic (one color).
2. Liquid lasers: They use organic dyes enclosed in a glass tube for an active
medium. A powerful pulse of light excites the organic dye.
3. Solid lasers: They use a solid, cylindrical crystal such as ruby, for the active medium.
Both ends of ruby are polished and parallel and the ruby is excited by a tungsten lamp tied to
an ac power supply. It produces a continuous wave.
4. Semiconductor lasers: They are made from semiconductor p-n junctions and are
commonly called injection laser diodes. The excitation mechanism is a dc power supply that
controls the amount of current to the active medium. The output light is easily modulated
making it very useful in many electronic communication systems.

Laser Characteristics

All types of lasers use

1. an active material to convert energy into laser light.

2. a pumping source to provide power or energy.

3. optics to direct the beam through the active material to be amplified.

4. optics to direct the beam into a narrow powerful cone of divergence.

5. a feedback mechanism to provide continuous operation.

6. an output coupler to transmit power out of the laser.

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Module – III

DIGITAL TRANSMISSION:

Pulse Modulation
Pulse code Modulation
Dynamic Range
Signal Voltage–to-Quantization Noise Voltage Ratio
Linear Versus Nonlinear PCM Codes
Commanding, PCM Line Speed
Delta Modulation PCM and Differential PCM.

MULTIPLEXING AND T CARRIERS:

Time- Division Multiplexing,

T1 Digital Carrier System,
Digital Line Encoding,
T Carrier systems,
Frequency- Division Multiplexing,
Wavelength- Division Multiplexing,
Synchronous Optical Network.

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DIGITAL TRANSMISSION
Digital transmission is the transmittal of digital signals between two or more points in a
communications system. The signals can be binary or any other form of discrete-level digital pulses.
With digital transmission systems, a physical facility, such as pair of wires, a coaxial cable or an
optical fiber cable is required to interconnect the various points within the system.

Digital transmission has several advantages over analog transmission:

Important advantage is the noise immunity as digital signals are inherently less susceptible than
analog signals to interference caused by noise.
Digital signals are better suited than analog signals for processing and combining using a technique
called multiplexing.
Digital transmission systems are more resistant to analog systems to additive noise because they
use signal regeneration rather than signal amplification.
Digital signals are simpler to measure and evaluate than analog signals.

Disadvantages:

Transmitting digitally encoded analog signals requires more bandwidth than simply transmitting
the original analog signal, which makes it expensive.
Also conversion of analog signals into digital pulses prior and after transmission requires
additional encoding and decoding circuitry.
Precise time synchronization between the clocks in transmitters and receivers is required in digital
transmission
Digital transmission systems are incompatible with older analog transmission systems.

Pulse Modulation

The process of sampling analog information signals and then converting those samples into
discrete pulses and then transporting the pulses from a source to a destination over a physical
transmission medium is called Pulse Modulation. Pulse modulation involves communication using a
train of recurring pulses. There are four different types of pulse modulation techniques.

Pulse Width Modulation (PWM): The width of a constant amplitude pulse is varied proportional to
the amplitude of the analog signal at the time the signal is sampled. It is sometimes called as pulse
duration modulation (PDM) or pulse length modulation (PLM). It is very popular in digital circuits
because of its easy generation and its applications include voltage regulators and class-D audio
amplifiers.

Pulse Position Modulation (PPM): The position of a constant-width pulse within a prescribed time
slot is varied according to the amplitude of the sample of the analog signal. It is commonly used in
communications over optic fibers as multipath fading is minimal. It is also used in communications for
RC aircraft/cars etc as demodulation is easy allowing a low-cost receiver.

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Pulse modulation: (a) analog signal; (b) sample pulse; (c) PWM; (d) PPM; (e) PAM; (f) PCM

Pulse Amplitude Modulation (PAM): The amplitude of a constant width, constant-width, and
constant-position is varied according to the amplitude of the sample of the analog signal. It resembles
the original analog signal more than the wave forms for PWM or PPM. Telephone modems faster than
300 bits/sec and Ethernet use PAM.

Pulse Code Modulation (PCM): The analog signal is sampled and then converted to a serial n-bit
binary code for transmission. Each code has the same number of bits and requires the same length of
time for transmission. Applications include digital audio in computers and CDs.

Pulse Code Modulation

PCM invented by Alex H. Reeves in 1937 is the preferred method of communications within
the public switched telephone network because with PCM, it is easy to combine digitized voice and
digital data into a single, high-speed digital signal and propagate it over either metallic or optical fiber
cables. With PCM, the pulses are of fixed length and fixed amplitude. PCM is a binary system where a
pulse of lack of pulse within a prescribed time slot represents either logic 1 or logic 0 conditions.

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The above figure shows a simplified block diagram of a single-channel, simplex PCM system. The
bandpass filter limits the frequency of the analog input signal to the standard voice-band frequency
range of 300 Hz to 3000 Hz. The sample and hold circuit periodically samples the analog input signal
and converts those samples to a multiple PAM signal. The analog-to-digital converter (ADC) converts
the PAM samples to parallel PCM codes, which are converted to serial binary data in the parallel-to-
serial converter and then outputted into the transmission line as serial digital pulses. Repeaters are
placed at prescribed distances to regenerate the digital pulses.
At receiver side, the serial-to-parallel converter converts serial pulses received from the
transmission line to parallel PCM codes. The digital-to-analog converter (DAC) converts the parallel
PCM codes to multilevel PAM signals. The hold circuit is basically a low pass filter that converts the
PAM signals back to its original analog form. An integrated circuit that performs the PCM encoding
and decoding functions is called a codec (coder/decoder).

PCM Sampling
Sampling circuit in a PCM transmitter periodically samples the continually changing analog
input voltage and converts those samples to a series of constant amplitude pulses, which can be more
easily converted to binary PCM code. Two basic techniques to perform sampling exist.

Natural Sampling: In natural sampling the pulse amplitude takes the shape of the analogue waveform
for the period of the sampling pulse. The frequency spectrum of the sampled output is different from
that of an ideal sample.

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Flat-top sampling: It is accomplished in a sample-and-hold circuit. Its purpose is to periodically

sample the continually changing analog input voltage and covert those samples to a series of constant
voltage PAM voltage levels. With flat-top sampling, the input voltage is sampled with a narrow pulse
and then relatively held constant until next sample is taken.

Flat-top sampling

Flat-top sampling introduces less aperture distortion than natural sampling and can operate with a
slower analog-to-digital converter.

Sampling Rate

Nyquist sampling theorem states that for a sample to be reproduced accurately, minimum
sampling rate, fs must be twice the higher input frequency, fa. Mathematically, the minimum Nyquist
sampling rate, fs is fs ≥ 2 fa, where fs is minimum Nyquist sample rate in hertz and fa is maximum
analog input frequency in hertz.

If fs is less than twice fa, i.e. fs < 2 fa, aliasing or foldover distortion occurs. This can be
overcome by using anti-aliasing filter before sampling to suppress the component before sampling.

Quantization and the Folded Binary Code

Quantization is the process of converting a continuous range of values into a finite range of
discreet values. This is a function of analog-to-digital converters, which create a series of digital values
to represent the original analog signal. Quantization is required to convert the analog signal to a PCM
code with a limited number of combinations. Taking an example, a sine wave with peak amplitude of
5v, varying between +5V and -5V passing through every amplitude between them. A PCM code could
have only eight bits, which equates to only 28 or 256 combinations and to be converted, the sine wave
values have to be rounded off.

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The above table shows the three bit PCM code, which is a three-bit sign magnitude code with
eight possible combinations (four +ve and four –ve). The left most bit is the sign bit (1 = + and 0 = -),
and the remaining two right most bits represent magnitude. This type of code is called folded binary
code because the codes on the bottom half of the table are an exact mirror image of the codes on the
top except for the sign bit. The magnitude difference between adjacent steps is called the quantization
interval or quantum (1V for above table). For the above code, the maximum signal magnitude that can
be encoded is +3V (111) or -3V (001) and the minimum is +1V (101) or -1V (001). If the magnitude of
the sample exceeds the highest quantization interval, overload distortion (peak limiting) occurs.
Assigning PCM codes to absolute magnitudes is called quantizing. The magnitude of a
quantum is also called the resolution. It is equal to the voltage of the least significant bit (Vlsb) of the
PCM code. The smaller the magnitude of a quantum, the better the resolution and the more accurately
the quantized signal will resemble the original analog signal.

(a) Analog input signal; (b) sample pulse; (c) PAM; (d) PCM code

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Each sample voltage is rounded off to the closest available quantization level and then
converted to its corresponding PCM code. The PAM signal in the transmitter is essentially the same
PAM signal produced in the receiver. So, any round-off errors in the transmitted signal are reproduced
when the code is converted back to the analog by the DAC in the receiver. This error is called the
quantization error (Qe), which is also quantization noise (Qn). The quantized signal shown above
roughly resembles the original input signal as with three-bit PCM code, poor resolution results and
only three samples are taken from analog signal.

As shown above, the quality of PAM signal can be improved by using a PCM code with more
bits, reducing the magnitude of quantum and improving the resolution. Also, the sampling the analog
signals at a faster rate increases the quality and the PAM signal resembles the analog signal
closely Quantization error is given by,

Dynamic Range
It is the ratio of the largest possible magnitude to the smallest possible magnitude (other than
0V) that can be decoded by the digital-to-analog converter in the receiver. Mathematically,

, where, DR is dynamic range and V min is the quantum value (resolution) and
Vmax is the maximum voltage magnitude that can be discerned by the receivers DACs. Dynamic range
can be expressed in decibels as,

or ,where n is the number of PCM bits.

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Signal Voltage –To-Quantization Noise Voltage Ratio

The maximum quantization noise is half the resolution. Therefore, the worst possible signal
voltage-to-quantization noise voltage ratio (SQR) occurs when the input signal is at its minimum
amplitude. Mathematically, the worst-case voltage SQR is 2.
For linear PCM codes, the signal power-to-quantizing noise power ratio is determined by the formula:

Where, R is resistance in ohms, v is rms signal voltage in volts, q is quantization interval in volts, v 2/R
is average signal power in watts and (q2/12)/R is average quantization noise power in watts.

Linear versus Nonlinear PCM codes

• Linear codes – magnitude change between any two successive steps in uniform
– Resolution/accuracy is the same for lower and higher amplitude signal
– SQR for low amplitude signal is less than the SQR for higher amplitude signal
• Nonlinear – step size increases with the amplitude of the input signal
– More codes at the bottom
– Distance between successive codes is greater for higher amplitude signals
– Vmax/Vmin is increased

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Companding
Companding is the process of compressing and expanding and is a means of increasing the
dynamic range of a communications system. Higher-amplitude analog signals are compressed prior to
transmission and then expanded in the receiver.

An analog input signal with a dynamic range of 50dB is compressed to 25dB prior to transmission and
then expanded back at the receiver. With PCM, companding may be accomplished using analog or
digital techniques.

Analog Companding

PCM system with analog companding

In the transmitter, the dynamic range of the analog signal is compressed, sampled, and then
converted to a linear PCM code. In the receiver, the PCM code is converted to a PAM signal, filtered

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and then expanded back to its original dynamic range. Two methods of analog companding exist (also
called log-PCM codes).

µ – Law companding: µ-law is a companding scheme used in telephone network to get more
dynamics to the 8 bit samples that is available with linear coding. Compression characteristics for
µ – Law is,

Where,
Vmax = maximum uncompressed analog input amplitude (volts)
Vin = amplitude of the input signal at particular instant of time (volts)
µ= parameter used to define the amount of compression (unitless) V out = compressed output amplitude
(volts)
The graph shows the compression curves for several values of µ. The higher the µ, the more
compression and also for µ = 0, the graph is linear.

A – Law: In Europe, the ITU-T has established A-law companding to be used to approximate true
logarithmic companding. Compression characteristic for A- law companding is,

Digital Companding
Digital companding involves compression in the transmitter after the input sample has been converted
to a linear PCM code and then expansion in the receiver prior to PCM decoding.

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Recent digitally compressed PCM systems use a 12-bit linear PCM code and an eight-bit compressed
PCM code. The compression and expansion curves closely resemble analog µ-law curves with a µ =
255.
The following figure shows 12- to eight-bit digital compression curve for positive values only. For
negative values, it’s identical but just inverse. Though 16 segments are present, this scheme is called
13-segment compression because the curve for segments +0, +1, -0 and -1 is a straight line.
The digital companding algorithm for a 12-bit linear to eight-bit compressed code is quite simple and
the compressed code consists of a sign bit, a three-bit segment identifier, and a five-bit magnitude code
which specifies the quantization interval within the specified segment.

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In the encoding table shown above, the bit positions designated with an X are truncated during
compression and thereafter lost. Bits designated by A, B,C, D along with the sign bit are transmitted as
is. The analog signal is sampled and converted to a linear 12-bit sign-magnitude code. The sign bit is
transferred directly to an eight-bit compressed code. The segment number in the eight-bit code is
determined by counting the number of leading 0’s in the 11-bit magnitude portion of the linear code
beginning with the most-significant bit and then subtracting the number of leading 0s from 7, which is
the segment number. The segment number is converted to a three-bit binary number and inserted into
the eight-bit compressed code as the segment identifier. The four magnitude bits (A, B, C, D) represent
the quantization interval and are submitted into the least-significant four bits of the eight-bit
compressed code.

Segments 2 through 7 are subdivided into smaller subsegments. Each segment consists of 16
subsegments corresponding to the 16 conditions possible for bits A, B,C and D (0000 – 1111). In
segment 2, there are two codes per subsegment and in segment 3, there are four. The number of codes
per subsegment doubles with each subsequent segment. So, in segment 7, each subsegment has 64
codes. In the decoder, the most significant of the truncated bits is reinserted as logic 1. The remaining
truncated bits are reinserted as 0s. This minimizes the magnitude of error introduced by compression
and expansion process.

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Digital Compression Error

The magnitude of the compression error is not the same for all samples. However, the maximum
percentage is the same in each segment (other than segments 0 and 1, where there is no compression
error), which is calculated using:

Every function performed by a PCM encoder and decoder is now accomplished with a single
integrated-circuit chip called codec. Some of the most recent developed codecs are called combo chips,
as they include an antialiasing (band-pass) filter, a sample-and-hold circuit, and an ADC in transaction
and a DAC, a hold circuit, and a band pass filter in the receive section.

PCM Line Speed

Line speed is the data rate at which serial PCM bits are clocked out of the PCM encoder onto the
transmission line. Line speed is dependent on the sample rate and the number of bits in the compressed
PCM code. Mathematically, it is:

, where Line speed is the

transmission rate in bits per second, samples/second is sample rate (fs) and bits/sample is number of
bits in the compressed PCM code.

Delta Modulation PCM and Differential PCM

Delta modulation PCM uses a single-bit PCM code to achieve digital transmission of analog
signals. Here, only a single bit is transmitted, which simply indicates whether the present sample is
larger or smaller in magnitude than the previous sample. If the current sample is larger in magnitude
than a previous sample, a logic 1 is transmitted and if its smaller, logic 0 is transmitted.

Differential pulse code modulation (DPCM) takes advantage of the sample-to-sample

redundancies in typical speech waveforms. With DPCM, a binary code proportional to the difference
in the amplitude of two successive samples is transmitted rather than a binary code of an actual sample.
As the range of sample differences is less than the range of individual sample amplitudes, fewer bits
are required for DPCM than for conventional PCM.

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MULTIPLEXING AND T CARRIERS

Multiplexing is the transmission of information from more than one source to more than one
destination over the same transmission medium. Multiplexing is accomplished in various domains
such as space, time, phase, frequency and wavelength.

Time Division Multiplexing

With TDM system, transmission from multiple sources occurs on the same transmission
medium but not at the same time. Transmission from various sources is interleaved in time domain.
The two basic forms of TDM are: Synchronous TDM (STDM) and Asynchronous (or) Statistical TDM
(STATDM)

In synchronous TDM, time slot ‘x’ is assigned to user m alone and cannot be used by any other user or
other device. T-1 and ISDN telephone lines are common examples of synchronous time division
multiplexing. Asynchronous TDM networks assign time slots only when they are to be used and delete
them when they are idle. STATDM is used in high density and high traffic applications.

With PCM-TDM system, two or more voice channels are sampled, converted to PCM codes,
and then time-division multiplexed onto a single metallic or optical fiber cable.

The above figure shows a block diagram for a PCM carrier system comprised of two DS-0
channels that have been time-division multiplexed. Each channel’s input is alternately sampled at an 8-
kHz rte and converted to an eight-bit PCM code. While the PCM code for channel-1 is being
transmitted, channel-2 is sampled and converted to PCM code. When its turn for channel-2’s PCM
code to be transmitted, the next sample is taken from channel-1 and converted to PCM code. This is a
continuous process.

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The multiplexer is simply an electronically controlled digital switch with two inputs and one
output. One eight-bit PCM code from each channel is called a TDM frame and the time it takes to
transmit one TDM frame is called frame time and it is equal to reciprocal of sample rate. The above
figure shows the frame allocation for a two channel PCM system. The PCM code for each channel
occupies a fixed time slot within the total TDM frame.

DS-hierarchy

T1 Digital Carrier System

A digital carrier system is a communications system that uses digital pulse rather than analog
signals to encode information. The following figure shows a block diagram of for the Bell system T1
digital carrier system, which is the North American telephone standard.

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A T1 carrier system time division multiplexes PCM encoded samples from 24 voice band
channels for transmission over a single metallic wire pair or a fiber optic cable. The multiplexer has 24
independent inputs and one time-division multiplexed output. The 24 PCM output signals are
sequentially selected and connected through the multiplexer to the transmission line. To become a T1
carrier, the system has to be line encoded and placed on special conditioned cables called T1 lines.

A transmitting portion of a Channel Bank digitally encodes the 24 analog channels, adds
signalling information into each channel, and multiplexes the digital stream onto the transmission
medium. The receiving portion reverses the process. Each of the 24 channels contains an eight-bit
PCM code and is sampled 8000 times a second. Each channel is sampled at the same rate, but may not
be at the same time. The line speed is calculated as:

=>

Later, an additional bit called the framing bit is added to each frame. The framing bit occurs
once per frame and is recovered at the receiver and its main purpose is to maintain frame and sample
synchronization between TDM transmitter and receiver.

As a result of this extra bit, each frame now contains 193 bits and the line speed for a T1 digital
carrier system is 1.544 Mbps. { 193 bits × 8000 frames = 1.544 Mbps}

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D-Type Channel Banks

D type Channel Bank refers to the terms used in T1 technology. Channel Bank defines the type
of formatting that is required for transmission on T1 trunk. The purpose of a Channel Bank in the
telephone company is to form the foundation of multiplexing and demultiplexing the 24 voice channels
(DS0). D type Channel Bank is one of the types of Channel Bank which is used for digital signals.
There are five kinds of Channel Banks that are used in the System: D1, D2, D3, D4, and DCT (Digital
Carrier Trunk).

Earlier T1 carrier systems used D1 digital channel banks (PCM encoders) with a seven-bit
magnitude-only PCM code, analog companding and a µ = 100. Modern versions use digital
companded, eight-bit sign magnitude-compressed PCM codes with a µ = 255.

Super frame TDM Format

The 8-kbps signalling rate used with the early digital channel banks was excessive for
signalling on standard telephone voice circuits. Therefore, with modern channel banks, a signalling bit
is substituted only into the least-significant bit (LSB) of every sixth frame. Hence, five of every six
frames have eight-bit resolution, while one in every six frames (the signalling frame) has only seven-
bit resolution.

Because only every sixth frame includes a signalling bit, it is necessary that all the frames be
numbered so that the receiver knows when to extract the signalling bit. Also, because signaling is
accomplished with a two-bit binary word, it is necessary to identify the most- and least-significant bits
(MSB and LSB, respectively) of the signalling word. A Super frame format was devised as shown
below.

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Within each super- frame are 12 consecutively numbered frames (1 to12). The signalling bits
are substituted in frames 6 and 12, the MSB into frame 6, and the LSB into frame 12. Frames1to 6 are
called the A highway, with frame 6 designated the A channel signalling frame. Frames 7 to 12 are
called the B high way, with frame 12 designated the B channel signalling frame. Therefore, in addition
to identifying the signalling frames, the sixth and twelfth frames must also be positively identified. To
identify frames 6 and 12, a different framing bit sequence is used for the odd- and even-numbered
frames. The odd frames (frames 1, 3, 5, 7, 9, and 11) have an alternating 1/0) pattern, and the even
frames (frames 2, 4, 6, 8, 10, and 12) have a 00 1110repetitive pattern. As a result, the combined
framing bit pattern is 1000 11011100. The odd numbered frames are used for frame and sample
Synchronization and the even-numbered frames are used to identify the A and B channel signalling
frames (frames 6 and 12). Frame 6 is identified by a 0/1 transition in the framing bit between frames 4
and 6. Frame 12 is identified by a 1/0 transition in the framing bit between frames 10 and 12.

D4 channel banks time-division multiplex 48 voice-band telephone channels and operate at a

transmission rate of 3.152 Mbps, which is slightly more than twice the line speed for 24-channel D1,
D2, or D3 channel banks because with D4 channel banks, rather than transmitting a single framing bit
with each frame, a 10-bit frame synchronization pattern is used.

Line speed is calculated as: total no of bits is 8 bits/channel × 48 channels = 384 bits/frame An
additional 10 bits are added for frame: so 394 bits/frame. Therefore, line speed of DS-1C system is
394×8000 = 3.152 Mbps

Extended Super frame Format

In telecommunication, an Extended Super Frame (ESF) is a T1 framing standard, sometimes

called D5 framing because it was first used in the D5 Channel Bank, invented in the 1980s. It requires
less frequent synchronization than the earlier super frame or D-4 format, and provides on-line, real-
time testing of circuit capability and operating condition.

In ESF, a super frame is 24 frames long, and the 193rd bit of each frame is used as framing bit.
Only 6 of the 24 framing bits are used for frame synchronization. Frame synchronization bits occur in
frames 4, 8, 12, 16, 20 and 24 and have a bit sequence of 001011. Six additional framing bits in frames
1, 5, 9, 13, 17, and 21 are used for an error-detection code called CRC-6 (cyclic redundancy checking).
The 12 remaining framing bits provide for a management channel called the facilities data link (FDL).
FDL bits occur in frames 2, 3, 6, 7, 10, 11, 14, 15, 18, 19, 22, and 23.

The extended super frame format supports a four-bit signalling word with signalling bits
provided in the second least-significant bit of each channel during every sixth frame. The signalling bit
in frame 6 is called the A bit, in frame 12 is called the B bit, in frame 18 is C bit and in frame 24 is
called D bit. These signalling streams are sometimes called the A, B, C and D signalling channels (or
signalling highways).

Fractional T carrier

Fractional T carrier emerged because standard T1 earners provide a higher capacity (i.e., higher
bit rate) than most users require. Fractional T1 systems distribute the channels (i.e., bits) in a standard
T1 system among more than one user, allowing several subscribers to share one T1line. Bit rates
offered with fractional T1 carrier systems are 64kbps (1 channel), 128 kbps (2 channels), 256 kbps (4

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channels), 384 kbps (6 channels), 512 kbps (8 channels), and 768 kbps (12 channels), with 384 kbps
(1/4 T1) and 768 kbps (1/2 T1) being the most common. The minimum data rate necessary to
propagate video information is 384 kbps.

The above figure shows four subscribers combining their transmissions in a special unit called a data service
unit/channel service unit (DSU/CSU). A DSU/CSU is a digital interface that provides the physical connection to
a digital carrier network. User 1 is allocated 128 kbps, user 2 - 256 kbps, user 3 - 384 kbps, and user 4 - 768
kbps for a total of 1.536 kbps (8 kbps is reserved for the framing bit).

North American Digital Multiplexing hierarchy

The above figure shows the American Telephone and Telegraph Company’s (AT&T’s) North American
Digital Hierarchy for multiplexing digital signals into a single higher-speed pulse stream suitable for
transmission on the next higher level of the hierarchy. A special device called muldem
(multiplexers/demultiplexer) is used to upgrade from one level in the hierarchy to the next-higher level. They
handle bit-rate conversions in both directions and are designated as M12, M23 etc. which identifies the
respective input and output digital signals. As shown, an M12 muldem interfaces DS-1 and DS-2 digital signals.
Also DS-1 signals can be further multiplexed or line encoded and placed on specially conditioned cables called
T1 lines.

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Digital signals are routed at central locations called digital cross-connects (DSX), which are
convenient for making patchable interconnections and routine maintenance and troubleshooting. Each
digital signal (i.e. DS-1, DS-2, etc) has its own digital switch (DSX-1, DSX-2…).

Digital Line Encoding

It involves converting standard logic levels to a form more suitable for telephone line transmission. Six
factors must be considered
Transmission voltages and DC component: Transmission voltages or levels can be categorized as
being either unipolar (UP) or bipolar (BP). Unipolar transmission involves the transmission of only a
single nonzero voltage level (either +ve or –ve for logic 1 and a 0 V for logic 0). In bipolar
transmission, two nonzero voltages are involved (+ve voltage for logic 1 and equal –ve voltage for
logic 0).
Duty cycle: The duty cycle of a binary pulse can be used to categorize the type of transmission. If the
binary pulse is maintained for the entire bit time, this is called nonreturn to zero (NRZ). If the active
time of the binary pulse is less than 100% of the bit time, it’s called return to zero (RZ). Unipolar and
Bipolar transmission voltages can be combined with either RZ or NRZ in several ways to achieve a
particular line-encoding scheme. Alternate mark inversion (AMI) scheme involves two nonzero voltage
levels (-V and +V) but both polarities represent logic 1s and 0V represents logic 0.

With NRZ encoding, a long string of either logic 1 or logic 0’s produces a condition in which a
receiver may lose its amplitude reference for discrimination between received 1’s and 0’s. This
condition is called dc wandering.
Bandwidth Requirements: The minimum bandwidth required to propagate a line-encoded digital
signal is determined by the highest fundamental frequency, which is in turn determined by the worst-
case (fastest transition) binary sequence. For, UPNRZ and BPNRZ the worst-case is alternating 1/0
sequence making the highest fundamental frequency one-half of the bit rate (fb/2). With BPRZ, it
occurs for successive logic 1’s and 0’s making the minimum bandwidth equal to bitrate fb. With
BPRZ-AMI, the worst-case condition is two or more consecutive logic 1’s, and minimum bandwidth is
one-half of bitrate (fb/2).
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Clock and framing bit recovery: To maintain clock and framing bit synchronization, there must be
sufficient transitions in the data waveform. Among all, BPRZ is the best encoding scheme for clock
recovery as a transition occurs in each position regardless of whether the bit is a 1 or 0.

Error detection: With UPNRZ, BPNRZ, UPRZ, and BPRZ encoding, there is no way to determine if
the received data have errors. However, with BPRZ-AMJ encoding, an error in any bit will cause a
bipolar violation (BPV—the reception of two or more consecutive logic is with the same polarity).
Therefore, BPRZ-AMI has a built-in error-detection mechanism. T carriers use BPRZ-AMI, with +3V
and —3 V representing logic 1 and 0 V representing a logic 0.

Digital Biphase: Digital biphase (sometimes called the Manchester code or diphase) is a popular type
of line encoding that produces a strong timing component for clock recovery and does not dc
wandering. Biphase is a form of BPRZ encoding that uses one cycle of a square wave at 00 phase to
represent a logic 1 and one cycle of a square wave at 1800 phase to represent a logic 0.

Manchester codes always have a transition at the middle of each bit period, and depending on
the state of the signal, may have a transition at the beginning of the period as well. In addition,
assuming equal probability of 1s and 0s, the average dc voltage is 0 V. and there is no dc wandering. A
disadvantage of biphase is that it contains no means of error detection.

T Carrier Systems
T carriers are used for the transmission of PCM-encoded time-division multiplexed digital
signals. Digital signals deteriorate as they propagate along a cable and regenerative repeaters are
placed at periodic intervals. It has three functional blocks: an amplifier/equalizer, a timing clock
recovery circuit, and the regenerator itself. The amplifier/equalizer filters and shapes the incoming
digital signal and raises its power level so that the regenerator circuit can make a pulse-no pulse
decision.

The timing clock recovery circuit reproduces the clocking information from the received data
and provides the proper timing information to the regenerator so that samples can be made at the
optimum time, minimizing the chance of an error occurring. A regenerative repeater is simply a
threshold detector that compares the sampled voltage received to a reference level and determines
whether the bit is logic 1 or logic 0. Spacing of repeaters is designed to maintain an adequate signal-to-
noise ratio for error-free performance.

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T1 Carrier System
T1 carrier systems were designed to combine PCM and TDM techniques for the transmission
of 24 64-kbps channels with each channel capable of carrying digitally encoded voiceband telephone
signals or data. The transmission bit rate (line speed) for a T1 carrier is 1.544 Mbps. Using TDM, T1
divides this bandwidth into 24 individual DS-0 channels, sampling each channel 8000 times per
second. Thus 8 × 8000 samples per second give each of the 24 DS-0 channels a data rate of 64kbps.
All 24 DS-0 channels combined has a data rate of 1.544 Mbps; this digital signal level is called DS-1.
Therefore T1 lines are sometimes referred to as DS-1 lines.
Alternate mark inversion (AMI) is the type of line coding used for T1 lines. Electrically, the
signal transmitted on a T1 line is a bipolar, return-to-zero (RZ) signal. This simply means that each
logical 1 bit is transmitted as a positive or a negative pulse, after which the line voltage always returns
to zero. A logical 0 bit is transmitted as a zero voltage on the line. This format is known as AMI
because each logical 1 bit (pulse or mark) is of opposite polarity from the previous one.

One additional benefit of the AMI bipolar format is that it permits detection of line errors. If a
line problem causes a pulse to be deleted or an unintended pulse to be transmitted, two consecutive
pulses with the same polarity on the line will result, called a bipolar violation (BPV).
With modern T1 carriers, a technique called binary eight zero substitution (B8ZS) is used to
ensure that sufficient transitions occur in the data to maintain clock synchronization. Here, whenever
eight consecutive 0s are encountered, one of two special patterns is substituted for the eight 0s, either
+-0-+000 or -+0+-000. The + and – represent bipolar logic 1 conditions and a 0 indicates a logic 0
condition.

The eight-bit pattern substituted for the eight consecutive 0s is the one that purposely induces bipolar violations
in the fourth and seventh bit positions. This code is then interpreted at the remote end of the connection. A full
1.544 Mbps T1 line contains 24 fractional T1 lines (abbreviated as FT1), each with a bandwidth of 64 kbps.
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Some of the limitations of T1 services are: They are very expensive, installation cost is also very high and some
case improper utilization of bandwidth.

T2 Carrier System
T2 carriers time-division multiplex 96 64-kbps voice or data channels into a single 6.312
Mbps data signal for transmission over twisted-pair copper wire upto 500 miles over a special LOCAP
(low capacitance) metallic cable. Higher transmission rates make clock synchronization even more
critical. So, an alternative method called binary six zero substitution (B6ZS) is used to ensure that
ample transitions occur in the data.

Whenever six consecutive logic 0s occur, either 0-+0+- or 0+-0-+ is substituted, and this code
is selected to create a bipolar violations in the second and fourth bits of the substituted patterns.

T3 Carrier System
T3 carriers time-division multiplex 672 64-kbps voice or data channels for transmission over a single
3A-RDS coaxial cable. The transmission bit rate is 44.736 Mbps and coding technique used with T3
carriers is binary three zero substitution (B3ZS).

T4M Carrier System

T4M carriers time division multiplex 4032 64-kbps voice or data channels for transmitting over a
single T4M coaxial cable upto 500 miles. The transmission rate is very high (274.16 kbps) making
substituting patterns impractical. They transmit scrambled unipolar NRZ digital signals.

T5 Carrier System
T5 carriers time-division multiplex 8064 64-kbps voice or data channels and transmits them at 560.16
Mbps over a single coaxial cable.

European Time-Division Multiplexing

In Europe, a different version of T carrier lines is used called E lines. With the basic E1 system,
a 125µs frame is divided into 32 equal time slots. Time slot 0 is used for a frame alignment pattern and
for an alarm channel. Time slot 17 is used for a common signalling channel (CSC). The signalling for
all 30 voice-band channels is accomplished on the common signalling channel. Consequently, 30
voice-band channels are time-division multiplexed into each E1 frame. Every slot has eight bits. So the
number of bits per frame is given as:

And the line speed can be given as 256 bits/frame × 8000 frames/second = 2.408 Mbps

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Statistical Time Division Multiplexing

Statistical time division multiplexing, is one method for transmitting several types of data
simultaneously across a single transmission cable or line. STA-TDM is often used for managing data
being transmitted via a local area network (LAN) or a wide area network (WAN). A statistical TDM
multiplexer exploits the natural breaks in transmissions by dynamically allocating time slots on a
demand basis. As like synchronous TDM system, a statistical mux has a finite number of low-speed
data input lines with one high-speed multiplexed data output line, and each input line has its own
digital encoder and buffer. With the statistical mux, there are n input lines and k time slots available (k
>n). The multiplexer scans the input buffers, collecting data until a frame is filled, at which time the
frame is transmitted. On the receiving end, the demultiplexer removes the data from the time slots and
distributes to their appropriate output buffers. Statistical multiplexers require low data rate than
synchronous multiplexers. Also, they can support more users operating at the same transmission rate.

With Statistical multiplexing, control bits must be included in the frame. The following figure
shows the overall frame format for a statistical TDM multiplexer.

The frame includes a beginning flag and ending flag to indicate the start and end of frame, an address
field that indicates the transmitting device, a control field, a statistical TDM subframe, and a frame
check sequence field (FCS), which provides error detection.

The above figure shows the frame when only one data source is transmitting. The transmitting devise is
identified in the address field. The data field is variable and this scheme works well in times of light
loads, but inefficient for heavy loads.

The above figure shows a way to improve efficiency by allowing more than one data source to be
included within a single frame.

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Frame Synchronization

With TDM systems, it is important not only that a frame has to be identified, but also
individual timeslots within the frame be identified. There are several methods used to establish frame
synchronization, including added digit, robbed digit, added channel, statistical and unique coding.
Considerable amount of overhead is added to transmission to achieve frame synchronization.
1. Added-Digit Framing: - T1 carriers using D1, D2 or D3 channel banks use added-digit
framing. A special framing digit (framing pulse) is added to each frame. The maximum
average synchronization time is given by
Synchronization time = 2NT = 2N2tb
Where N is number of bits per frame and T is frame period of Ntb and tb is bit time.

2. Robbed-Digit Framing: - Added-digit framing is inefficient when a short frame is used in the
case of single-channel PCM systems. As an alternative, the least significant bit of every nth
frame is replaced with a framing bit. This process is called robbed-digit framing and it does
not interrupt transmission, but instead periodically replaces information bits with forced data
errors to maintain frame synchronization.

3. Added-Channel Framing: - It is essentially same as added-digit framing except that digits

are added in groups or words instead of as individual bits. The average number of bits to
acquire frame synchronization using added-channel framing is N2/2(2K – 1), where N is
number of bits per frame and K is number of bits in the synchronizing word.

4. Statistical Framing: - Here no robbing or adding digits is done. As a signal that has a
centrally peaked amplitude distribution generates a high probability of logic 1 in the second
digit, the second digit of a given channel can be used for the framing bit.

5. Unique-Line Code Framing: - Some property of the framing bit is different from the data
bits. The framing bit is either made higher or lower in amplitude or with different time
duration. The advantage is that synchronization is immediate and automatic. The disadvantage
is additional processing requirements necessary to generate and recognize the unique bit.

Frequency Division Multiplexing

Assignment of non-overlapping frequency ranges to each “user” or signal on a medium, such
that all signals are transmitted at the same time, each using different frequencies. FDM is used for
combining many relative narrowband sources into a single wideband channel, such as in public
telephone systems. Essentially FDM is taking a given bandwidth and subdividing it into narrow
segments with each segment carrying different information. FDM is an analog multiplexing scheme.

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In the above figure-b, five signal sources are fed into a multiplexer that modulates each signal
onto a different frequency (f1, f2, f3, f4 , f5). To prevent interference, the channels are separated by guard
bands, which are unused portions of the spectrum. With FDM, each user has its own modulating
circuitry, a transmitter, a receiver and a demodulator. The channel is common to all users. Since each
transmitter is using a carrier of a different frequency, there is no interference unless the sidebands or
carriers are incorrectly assigned and therefore overlap. AM, FM and cable TV broadcasting are most
common examples of FDM where each station uses a different frequency band.

Advantages of FDM:

1. In FDM system, users can be added to the system by simply adding another pair of
transmitter modulator and receiver demodulators.
2. FDM system support full duplex information flow which is required by most of the
applications
3. Noise problem for analog communication has less effect.

Disadvantages of FDM:

1. The initial cost is high, which includes the cable between the two ends and associated
connectors for the cable.
2. One users problem can sometimes affect others
3. Each user requires a precise carrier frequency.

Wavelength-Division Multiplexing
WDM involves transmission of multiple digital signals using several wavelengths without their
interfering with one another. This technology enables many optical signals to be transmitted
simultaneously by a single fiber cable. It is also referred to as wave-division multiplexing.

WDM is accomplished by modulating injection laser diodes, which are transmitting highly
concentrated light waves at different wavelengths (i.e. at different optical frequencies). Therefore
WDM is coupling light at two or more discrete wavelengths into and out of an optical fiber. Each
wavelength is capable of carrying vast amounts of information in either analog or digital form, and the
information can already be time- or frequency-division multiplexed.

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Advantages of WDM:
1. Enhanced capacity as full-duplex transmission is also possible with a single fiber.
2. WDM is inherently easier to reconfigure (i.e. adding of removing channels)
3. Usage of optical components makes it simpler, more reliable and often less costly

Disadvantages of WDM:
1. Signals cannot be placed so close in the wavelength spectrum that they interfere with each
other.
2. The overall signal strength should be approximately the same for each wavelength which may
not be possible.
3. Light waves carrying WDM are limited to a two-point circuit or a combination of many two-
point circuits that can go only where the cable goes.

Dense wavelength division multiplexing (DWDM) is a fiber-optic transmission technique that employs
light wavelengths to transmit data parallel-by-bit or serial-by-character.

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Advantages:

Protocol & Bit Rate independence

Increased overall capacity at much lower cost
Current fiber plant investment can be optimized by a factor of at least 32
Transparency
Physical layer architecture supports both TDM and data formats such as ATM, Gigabit
Ethernet, etc.
Scalability
Utilize abundance of dark fibers in metropolitan areas and enterprise networks

Disadvantages:
Dispersion
Chromatic dispersion
Polarisation mode dispersion
Attenuation
Intrinsic: Scattering, Absorption, etc.
Extrinsic: Manufacturing Stress, Environment, etc.
Four wave mixing
Non-linear nature of refractive index of optical fiber
Limits channel capacity of the DWDM System

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Synchronous Optical Network (SONET)

The synchronous optical network is a multiplexing system similar to conventional time-
division multiplexing except SONET was developed to be used with optical fibers. SONET is the
name for a standard family of interfaces for high speed optical links. These start at 51.84 Mbps, which
is referred to as synchronous transport level 1 (STS-1). It is comprised of 28 DS-1 signals. Each DS-1
signal is equivalent to a single 24-channel T1 digital carrier system. With STS-1, it is possible to
extract or add individual DS-1 signals with completely disassembling the entire frame. OC-48 is the
second level of SONET multiplexing. It has a transmission bit rate of 2.48Gbps.
SONET Applications:
1. High speed backbone networks
2. Basic architecture for B-ISDN
3. Basic architecture for ATM
4. High speed optical networks for data communications.

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Questions:

1. a) What is a T carrier system? What is a fractional T carrier system? Describe

in detail, the various T carrier systems.

b) Compare WDM and DWDM and also list the advantages and
disadvantages of WDM.

2. a) What do you understand by companding?? Compare analog anddigital

companding

b) What is SQR and give its relationship to resolution, dynamic range and
maximum no of bits in a PCM code.

3. a) What is the difference between FDMA, TDMA and CDMA

b) What is line speed and how is it determined?

4. a) What is superframe and extended superframe TDM format? Explain with

an example.

b) What is frame synchronization and how it is achieved in PCM-TDM

system.

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MODULE-IV

WIRELESS COMMUNICATIONS SYSTEMS:

Electromagnetic Polarization,
Electromagnetic Radiation,
Optical Properties of Radio Waves,
Terrestrial Propagation of Electromagnetic Waves,
Skip Distance, Free-Space Path Loss,
Microwave Communications Systems,
Satellite Communications Systems

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WIRELESS COMMUNICATIONS SYSTEMS

With wireless communication systems, electromagnetic signals are emitted from an antenna,
propagate through the earth’s atmosphere (air) or free space (a vacuum), and are then received
(captured) by another antenna. Sometimes, it is impractical to interconnect two pieces of equipment
physically. So, free space or earth’s atmosphere is often used as the transmission medium. Free space
propagation of electromagnetic waves is often called radio-frequency (RF) propagation or simply radio
propagation. Wireless communications include terrestrial and satellite microwave radio systems,
broadcast radio systems, two-way mobile radio and cellular telephone.

Electromagnetic Polarization

Electromagnetic waves are comprised of an electric and a magnetic field at 90 degrees to each
other. The polarization of a plane electromagnetic wave is simply the orientation of the electric field
vector in respect to earth’s surface. If the polarization remains constant, it is described as linear
polarization. Horizontal and vertical polarizations are two forms of linear polarization. A wave is
horizontally polarized if the electric field propagates parallel to the earth’s surface, and the wave is
vertically polarized if the electric field propagates perpendicular to the earth’s surface. The wave is
described as having circular polarization if the polarization vector rotates 360 degrees, as the wave
moves one wavelength through space and the field strength is equal at all angles of polarization. When
the field strength varies with changes in polarization, this is described as elliptical polarization. A
rotating wave can turn in either direction. If the vector rotates in a clockwise direction, it is right
handed, and if the vector rotates in a counter-clockwise direction, it is considered left handed.

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Rays and Wavefronts

Rays and wavefronts are used for analysing electromagnetic waves. A ray is a line drawn along
the direction of propagation of an electromagnetic wave. Rays are used to show the relative direction
of propagation.

A wavefront shows a surface of constant phase of electromagnetic waves. A wavefront is

formed when points of equal phase on rays propagating from the same source are joined together. The
above figure shows a wavefront with a surface that is perpendicular to the direction of propagation
(rectangle ABCD). When a surface is plane, its wavefront is perpendicular to the direction of
propagation. A point source is a single location from which rays propagate equally in all directions
(i.e. isotropic source). The wavefront generated from a point source is simply a sphere with radius R
and its center located at the point of origin of the waves.

Electromagnetic Radiation

The flow of electromagnetic waves (energy) in the direction of propagation is called

electromagnetic radiation. The rate at which energy passes through a given surface area in free space is
called power density, usually given in watts per square meter. Mathematically,

Power density P = EH, where P is power density (watt/m2), E represents rms electric field intensity
(volts/meter) and H represents rms magnetic field intensity (ampere turns/meter).

Spherical Wavefront and Inverse Square Law

A spherical wavefront is obtained by an isotopic radiator. All points at distance R(radius) from
the source lie on the surface of the sphere and have equal power densities. At an instance of time, the
total power radiated Prad is uniformly distributed over the total surface of the sphere. Therefore, the
power density at any point on the sphere is the total radiated power divided by the total area of the
sphere and can be given as,
P = Prad / (4πR2)

The power density becomes smaller as the distance from isotropic source increases. The total
radiated power is same. But as the area of the sphere increases in direct proportion to the square of
distance from source, the power density is inversely proportional to the square of the distance from the
source. This relationship is called inverse square law.

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Wave Attenuation and Absorption

When waves propagate through free space, they spread out, resulting in reduction of power
density. This is called attenuation loss and it occurs in free space as well as earth’s atmosphere.
Earth’s atmosphere contains different particles which absorb electromagnetic energy, causing
reduction in power, called as absorption loss. The reduction in power density with increase in distance
is equivalent to a power loss and is called wave attenuation. Because it’s due to spherical spreading of
wave in space, it is sometimes called space attenuation. Mathematically, wave attenuation is
γA= 10 log (P1/P2), where γA represent wave attenuation in dB, P1 is power density at point 1 and P2
is power density at point 2.

Earth’s atmosphere is not a vacuum and it consists of atoms, molecules of various substances
such as gases, liquids and solids, which are quite capable of absorbing EM waves. As the wave
propagates, energy is transferred from the wave to the atoms and molecules and this transfer is known
as wave absorption and is analogous to I2R power loss. Once absorbed, energy is lost forever and
causes reduction in the power density.

Optical Properties of Radio Waves

The free space behaviour of propagation is altered by optical effects such as refraction,
reflection, diffraction and interference.

Refraction: Electromagnetic refraction is the change in direction of an electromagnetic wave as it

passes obliquely from one medium to another medium with a different density (refractive index).

Refraction at a plane boundary between two media

Whenever a ray passes from a less dense to a more dense medium, it is effectively bent toward
the normal (imaginary line drawn perpendicular to the interface at the point of incidence). Conversely,
whenever a ray passes from a more dense to a less dense medium, it is effectively bent away from the
normal. The angle of incidence is the angle formed between the incident wave and the normal, and the
angle of refraction is the angle formed between the refracted wave and the normal. Snell’s law states
that,

where θ1 and θ2 are angles of incidence and refraction and n1 and n2 are refractive indexes of material1
and material2.

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Refraction also occurs when a wavefront propagates in a medium that has a density gradient
that is perpendicular to the direction of propagation. The following figure shows wavefront refraction
in earth’s atmosphere (which has gradient refractive index).

Wavefront refraction in a gradient medium

The medium is more dense near the bottom and less dense near the top (upper atmosphere).
Therefore, rays travelling in the upper layers of the atmosphere travel faster than rays travelling near
earth’s surface and, consequently, the wavefront tilts downward. The tilting occurs in a gradual fashion
as the wave progresses.

Reflection: Electromagnetic wave reflection occurs when an incident wave strikes a boundary of two
media and some or all of the incident power does not enter the second material (i.e., they are reflected).
The following figure shows electromagnetic wave reflection at a plane boundary between two media.

Because all the reflected waves remain in medium1, angle of reflection equals the angle of
incidence (θi = θr). The ratio of reflected to incident power is Γ, expressed as T = Pr/Pi where T is
reflection coefficient and Pr and Pi are reflected and incident power.

For perfect conductors, T = 1 and all incident power is reflected. Reflection also occurs when
the reflective surface is irregular. When an incident wavefront strikes an irregular surface, it is
randomly scattered in many directions. Such a condition is called diffuse reflection, whereas reflection
from a perfectly smooth surface is called specular (mirror like) reflection.

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Diffraction: Diffraction is defined as the modulation or redistribution of energy within a wavefront

when a density it passes near the edge of an opaque object. Diffraction is the phenomenon that allows
light or radio waves to propagate (peek) around corners. Huygen’s principle states that every point on
a given spherical wavefront can be considered as a secondary point source of electromagnetic waves
from which other secondary waves (wavelets) are radiated outward. Huygen’s principle is illustrated
below.

The first figure shows normal wave propagation considering an infinite plane. Each secondary
point source (P1, P2 and so on) radiates energy outward in all directions. But, the wavefront continues
in its original direction rather than spreading out because cancellation of the secondary wavelets occurs
in all directions except straight forward. Therefore, the wavefront remains plane. When a finite plane
wavefront is considered, as in second figure, cancellation in random directions is incomplete. So, the
wavefront spreads out or scatters. This scattering effect is called diffraction.

The third figure shows diffraction around the edge of an obstacle. It can be seen that wavelet
cancellation occurs only partially. Diffraction occurs around the edge of the obstacle, which allows
secondary waves to “sneak” around the corner of the obstacle into what is called the shadow zone.

Interference: Radio wave interference occurs when two or more electromagnetic waves combine in
such a way that system performance is degraded. Interference, on the other hand, is subject to the
principle of linear superposition of electromagnetic waves and occurs whenever two or more waves
simultaneously occupy the same point in space.

In the above figure, it can be seen that, at point X the two waves occupy the same area in space.
However, wave B has travelled a different path than wave A and, therefore, their relative phase angles
may be different. If the difference in distance travelled is an odd-integral multiple of one-half
wavelength, reinforcement takes place. If the difference is an even-integral multiple of one- half
wavelength, total cancellation occurs.

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Terrestrial Propagation of Electromagnetic Waves

Electromagnetic waves travelling within earth’s atmosphere are called terrestrial waves and
communications between two or more points on earth is called terrestrial radio communications. There
are three modes of propagating EM wave within earth’s atmosphere: ground wave propagation, space
wave propagation and sky wave propagation.

Normal modes of wave propagation

Ground Wave Propagation:

Ground waves are the electromagnetic waves that travel along the surface of earth and are also
called as surface waves. Ground waves must be vertically polarized and the changing electric field
induces voltages in earth’s surface, which cause currents to flow that are very similar to those in a
transmission line. Ground waves are attenuated as they propagate because of the presence of resistance
and dielectric losses in the earth’s surface. Ground waves propagate best over a surface that is a good
conductor, such as salt water and poorly over dry desert areas. Also losses in ground waves increase
rapidly with frequency, ground wave propagation is limited to frequencies below 2 MHz
The following figure shows ground wave propagation. Because of earth’s gradient density,
ground wave propagates around the earth, remaining close to its surface.

Surface (ground) wave propagation

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The frequency and terrain over which the ground wave propagates has to be selected carefully
to ensure that the wavefront does not tilt excessively and simply turn over, lie flat on the ground and
cease to propagate. Ground wave communication is commonly used for ship-to-ship and ship-to-shore
communications, for radio navigation and for maritime mobile communications.

Space Wave Propagation: It includes radiated energy that travels in the lower few miles of earth’s
atmosphere. Space wave include both direct and ground reflected waves.

Direct waves travels essentially in a straight line between transmit and receive antennas. And
this propagation with direct waves is commonly called line-of-sight (LOS) transmission. Direct space
wave propagation is limited by the curvature of the earth. Ground-reflected waves are waves reflected
by earth’s surface as they propagate between transmit and receive antennas. The field intensity at the
receive antenna depends on the distance between the two antennas (attenuation and absorption) and
whether the direct and ground-reflected waves are in phase (interference). The curvature of earth
presents a horizon to space wave propagation commonly called the radio horizon. Because the
conditions in earth’s lower atmosphere are subject to change, the degree of refraction can vary with
time. A special condition called duct propagation occurs when the density of the lower atmosphere is
such that electromagnetic waves can propagate within the duct for great distances, causing them to
propagate around earth following its natural curvature.

Sky Wave Propagation: Electromagnetic waves that are directed above the horizon level are called
sky waves. Sky waves are radiated toward the sky, where they are either reflected or refracted back to
earth by the ionosphere. Because of this, sky wave propagation is sometimes called ionospheric
propagation. The ionosphere is the upper portion of earth’s atmosphere and is located approximately
50km to 400km (31 mi to 248 mi) above earth’s surface. Because of the ionosphere’s non uniform
composition and its temperature and density variations, it is stratified. Essentially, three layers make
up the ionosphere (the D, E, and F layers) and are shown below:

Ionospheric Layers
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All three layers of the ionosphere vary in location and in ionization density with the time of
day. The ionosphere is most dense during times of maximum sunlight. Because the density and
location of the ionosphere vary over time, the effects it has on electromagnetic radio wave propagation
also vary.

Skip Distance
The skip distance is the distance from the transmitter to the point where the sky wave first
returns to the earth. The skip distance depends on the wave’s frequency and angle of incidence, and the
degree of ionization.

The SKIP ZONE is a zone of silence between the point where the ground wave becomes too
weak for reception and the point where the sky wave is first returned to Earth. The size of the skip
zone depends on the extent of the ground wave coverage and the skip distance. When the ground wave
coverage is great enough or the skip distance is short enough that no zone of silence occurs, there is no
skip zone.

Free-Space Path Loss

In telecommunication, free-space path loss (FSPL) is the loss in signal strength of an
electromagnetic wave that would result from a line-of-sight path through free space, with no obstacles
nearby to cause reflection or diffraction. With free-space path loss, no electromagnetic energy is
actually lost—it merely spreads out as it propagates away from the source resulting in a lower power
density. It’s also referred as spreading loss, which occurs simply because of the inverse square law.
Spreading loss is a function of distance from the source and the wavelength (frequency) of the
electromagnetic wave. Mathematically, free-space path loss is proportional to the square of the
distance between the transmitter and receiver, and also proportional to the square of the frequency of
the radio signal.

Where:
is the signal wavelength (in metres),
is the signal frequency (in hertz),
is the distance from the transmitter (in metres),
is the speed of light in a vacuum, 2.99792458 × 108 metres per second

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For typical radio applications, it is common to find measured in units of MHz and in km,
in which case the FSPL equation becomes

Microwave Communication Systems

Microwaves are generally described as electromagnetic waves with frequencies that range from
approximately 500 MHz to 300 GHz. Because of their high frequencies, microwaves have relatively
short wavelengths. Microwave systems are used for carrying long-distance voice telephone service,
metropolitan area networks, wide area networks and the Internet. There are different types of
microwave systems operating over distances that vary from 15 miles to 4000 miles in length. Intrastate
or feeder service microwave systems are generally classified as short haul because they are used to
carry information for relatively short distances, such as between cities within the same state. Long-haul
microwave systems are those used to carry information for relatively long-distances, such as interstate
and backbone route applications. Microwave radio system capacities range from less than 12 voice
grade telephone circuits to more than 22,000 voice and data channels.

Advantages of Microwave Radio Communication:

1. Radio systems do not require a right-of-way acquisition between stations.

2. Each station requires the purchase or lease of only a small area of land.
3. Because of their high operating frequencies, microwave radio systems can carry large quantities
of information.
4. High frequencies mean short wavelengths, which require relatively small antennas
5. Radio signals are more easily propagated around physical obstacles, such as water and high
mountains.
6. Microwave Systems require fewer repeaters for amplification.
7. Distances between switching centers are less.
8. Underground facilities are minimized.
9. Minimum delay times are introduced.
10. Minimal crosstalk exists between voice channels.

Disadvantages of Microwave radio systems:

1. The electronic circuits used with microwave frequencies are more difficult to analyze.
2. Conventional components, such as resistors, inductors, and capacitors, are more difficult to
manufacture and implement at microwave frequencies.
3. Microwave components are more expensive.
4. Transistor transit time is a problem with microwave devices.
5. Signal amplification is more difficult with microwave frequencies.

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Microwave Radio Link

The following figure shows a simplex microwave radio link. The transmitter includes a
modulator, mixer, and microwave generator and several stages of amplification and filtering.

The modulator may perform frequency modulation or some form of digital modulation such as PSK or
QAM. The output of modulator is an intermediate frequency (IF) carrier that has been modulated or
encoded by the baseband input signal. The baseband signal is simply the information. The mixer and
microwave generator (oscillator) combine to perform frequency up-conversion through nonlinear
mixing. The up-converter is to translate IF frequencies to RF microwave frequencies.

The receiver consists of a radio-frequency (RF) amplifier, a frequency down-converter and a

demodulator. The RF amplifier and filter increase the received signal level so that the down-converter
can convert the RF signals to IF signals. The demodulator can be for FM, PSK or QAM. The output of
demodulator is the original baseband (information) signals.

Microwave Radio Repeaters

With systems longer than 40 miles or when geographical obstructions block the transmission
path, repeaters are needed. A microwave repeater is a receiver and transmitter placed back to back in
the system.

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The repeater station receives a signal, amplifies and reshapes it, and then retransmits it to the next
repeater or terminal station down line from it. A terminal station is simply a station at the end of a
microwave system where information signals originate and terminate.

Satellite Communication Systems

A satellite is a celestial body that orbits around a planet. In other terms, a satellite is a space
vehicle launched by humans that orbits earth or another celestial body. Communication satellites are
manmade satellites that orbit earth, providing a multitude of communications services to a wide variety
of consumers, including military, governmental, private and commercial subscribers. The main
purpose of communications satellite is to relay signals between two or more earth stations. A satellite
repeater is called a transponder, and a satellite may have many transponders. Transmissions to and
from satellites are categorized as either bus or payload. The bus includes control mechanisms that
support the payload operation. The payload is the actual user information. Satellites utilize many of the
same frequency bands as terrestrial microwave radio systems.

Satellite Elevation Categories

Satellites are generally classified as having a low earth orbit (LEO), medium earth orbit (MEO), or
geosynchronous earth orbit (GEO).

LEO satellites operate in the 1.0 GHz to 2.5 GHz frequency range. Main advantage is that the
path loss between earth stations and space vehicles is much lower thereby resulting in lower
transmit powers, smaller antennas and less weight. Example is Motorola’s satellite-based mobile
telephone system, Iridium.
MEO satellites operate in the 1.2 GHz to 1.67 GHz frequency band and orbit between 6000
miles and 12,000 miles above earth. Example is DOD’s satellite based global positioning system,
NAVSTAR.
Geosynchronous or geostationary satellites operate primarily in the 2 GHz to 18 GHz frequency
spectrum with orbits 22,300 miles above the earth’s surface. They orbit in a circular pattern with
an angular velocity equal to that of earth and have an orbital time of approx 24 hours (i.e. same
as earth).

Satellite Orbits and Orbital Patterns

Satellites are classified as either synchronous or nonsynchronous. Synchronous satellites orbit

earth above the equator with the same angular velocity as earth and therefore appear to be stationary
and remain in the same location with respect to a given point on earth. Nonsynchronous satellites
rotate around earth in circular or elliptical pattern as shown below.

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In circular orbit, the speed or rotation is constant. With elliptical orbits, the velocity of a
satellite is greatest when satellite is closest to earth. The point in an elliptical orbit farthest from earth
is called the apogee, and the point on the orbit closest to earth is called perigee. If satellite rotation is
in the same direction as earth’s rotation with angular velocity greater than that of earth, the orbit is
called prograde or posigrade orbit. If it’s in opposite direction with angular velocity less than that of
earth, then it’s called a retrograde orbit. Nonsynchronous satellites revolve around in a prograde
orbit, resulting in change of position continuously in respect to a fixed position on earth. So expensive
and complicated tracking equipment is needed to locate and lock the antennas onto the satellite track.
Out of infinite number of orbital paths possible, only three are used for communication
satellites: inclined, equatorial, or polar. When satellites orbit the Earth, either in a circular or elliptical
orbit, the satellite orbit forms a plane that passes through the centre of gravity called geocentre of the
Earth.

Inclined orbits are virtually all orbits except those that travel directly above the equator or directly
above the North and South Poles.

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The angle of inclination is the angle between the earth’s equatorial plane and the orbital plane
of a satellite measured counter clockwise at the point in the orbit where it crosses the equatorial plane
from south to north and this point is called ascending node. If it’s passing from north to south, it is
called descending node. Angles of inclination vary between 0 degrees and 90 degrees. The line
joining both these nodes through the center of earth is called line of nodes.

An equatorial orbit is when the satellite rotates in an orbit directly above the equator, usually
in a circular path. With an equatorial orbit, the angle of inclination is 0 degrees. All geosynchronous
satellites are in equatorial orbits. A polar orbit is when the satellite rotates in a path that takes it over
the North and South Poles in an orbital pattern that is perpendicular to the equatorial plane. The angle
of inclination of a satellite in a polar orbit is nearly 90 degrees. 100% of earth’s surface can be covered
with a single satellite in a polar orbit. Satellites in polar orbits rotate around earth in a longitudinal
orbit while earth is rotating on its axis in a latitudinal rotation.

Geosynchronous Satellites

Also referred to as geostationary, it refers to the movement of communications satellites where

the satellite circles the globe over the equator, in a movement that is synchronized with the earth's
rotation. Because of this synchronization, the satellite appears to be stationary, and they also offer
continuous operation in the area of visibility. Geosynchronous orbits are circular. There is only one
geosynchronous earth orbit, which is occupied by a large number of satellites.

Geosynchronous orbit requirements: The most important requirement is that the orbit must have a
0-degree angle of elevation. They also must orbit in the same direction as earth’s rotation with the
same angular velocity. Using Kepler’s third law, it can be shown that geosynchronous satellites must
revolve around earth in a circular pattern 42,164 km from the center of the earth. The circumference of
a geosynchronous satellite orbit is

C = 2 π (42,164km) = 264,790 km, and the velocity (v) is v = 264,790 km/24hr = 6840 mph

Clarke orbit: Synonymous with geostationary orbit. It is so-named because noted author Arthur C.
Clarke was the first person to realize that this orbit would be useful for communication satellites. The
Clarke orbit meets the concise setoff specifications for geosynchronous satellite orbits: (1) be located
directly above the equator, (2) travel in the same direction as earth’s rotation with a velocity of 6840
mph, (3) have an altitude of 22,300 miles above earth and (4) complete one revolution in 24 hours

Geosynchronous satellite advantages and disadvantages:

Some of the advantages are,

1. Geosynchronous satellites remain almost stationary in respect to a given earth station; therefore,
expensive tracking equipment is not required at the earth stations.

2. Geosynchronous satellites are available to all earth stations within their shadow 100% of the
time. The shadow of a satellite includes all the earth stations that have a line-of-sight path to the
satellite.

3. Switching from one geosynchronous satellite to another as they orbit overhead is not necessary.
Consequently, there are no transmission breaks due to switching times
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Disadvantages are;
1. An obvious disadvantage of geosynchronous satellites is they require sophisticated and heavy
propulsion devices on board to keep them in a fixed orbit.
2. High-altitude geosynchronous satellites introduce much longer propagation delays. The
roundtrip propagation delay between two earth stations through a geosynchronous satellite is
typically between 500 ms and 600 ms.
3. Geosynchronous satellites require higher transmit power levels and more sensitive receivers
because of the longer distances and greater path losses.
4. High precision spacemanship is required to place a geosynchronous satellite into orbit and to
keep it there.
Satellite Look Angles
Two angles have to be determined to ensure the earth station antenna is pointed directly at the
satellite: the azimuth and the elevation angle. Both of them together are referred to as look angles.
With geosynchronous satellites, the look angles of earth station antennas need to be adjusted only
once, as the satellite will remain in a given position permanently except for minor variations. The pint
on the surface of earth directly below the satellite is used to identify its location is called the
subsatellite point (SSP) and for geosynchronous satellites, SSP must fall on the equator.

Satellite Antenna Radiation Patterns: Footprints

The geographical representation of the area on earth illuminated by the radiation from a
satellite’s antenna is called a footprint or sometimes a footprint map. In essence, a footprint of a
satellite is the area on earth’s surface that the satellite can receive from or transmit to. The shape of a
satellite’s footprint depends on the satellite’s orbital path, height, and the type of antenna used. The
higher the satellite, the more of the earth’s surface it can cover.
The radiation pattern from a satellite’s antenna is sometimes called a beam. The smallest and
most directive beam is called a spot beam, followed by zonal beams, hemispherical beams, and earth
(global) beams.

Satellite Multiple-Accessing Arrangements

Satellite multiple accessing implies that more than one user has access to one or more
transponders within a satellite’s bandwidth allocation. The three most commonly used multiple
accessing arrangements are frequency division multiple accessing (FDMA), time-division multiple
accessing (TDMA) and code-division multiple accessing (CDMA).

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Frequency division multiple accessing: FDMA is a method of multiple accessing where a given RF
bandwidth is divided into smaller frequency bands called subdivisions. FDMA transmissions are
separated in the frequency domain and must share the total available transponder bandwidth as well as
total transponder power. A control mechanism is used to ensure that two or more earth stations do not
transmit in the same subdivision at the same time. Essentially, the control mechanism designates a
receive station for each of the subdivisions. Thus, with FDMA, transmission can occur from more than
one station at the same time, but the transmitting stations must share the allocated power, and no two
stations can utilize the same bandwidth.

Time-division multiple accessing: TDMA is the predominant multiple-accessing method used today.
TDMA is a method of time-division multiplexing digitally modulated carriers between participating
earth stations within a satellite network using a common satellite transponder. With TDMA, each earth
stat ion transmits a short burst of information during a specific time slot within a TDMA frame. The
bursts must be synchronized so that each station’s burst arrives at the satellite at a different time, thus
avoiding a collision with another station’s carrier. TDMA transmissions are separated in the tune
domain, and with TDMA, the entire transponder bandwidth and power are used for each transmission
but for only a prescribed interval of time. Thus, with TDMA, transmission cannot occur from more
than one station at the same time. However, the transmitting station can use all the allocated power and
the entire bandwidth during its assigned time slot.

Code-division multiple accessing: CDMA is based on the use of modulation technique known as
spread spectrum. Users are separated both by frequency and time.

Because there are no limitations on bandwidth, CDMA is sometimes referred to as spread-spectrum

multiple accessing (SSMA). With CDMA, all earth stations transmit within the same frequency band
and, for all practical purposes, have no limitations on when they may transmit or on which carrier
frequency. Thus, with CDMA, the entire satellite transponder bandwidth is used by all stations on a
continuous basis. Signal separation is accomplished with envelope encryption/decryption techniques

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Comparison: In both FDMA and TDMA, only one subscriber at a time is assigned to a channel. No
other conversion can access this channel until the subscriber’s call is finished or until that original call
to be handed off to a different channel by the system. Voice data tends to be burst in nature. So much
of the time, no data is being sent over the channel. This inefficiency tends to limit the capacity of the
system. The above drawbacks are overcome in this third technique in which the users are spread across
both frequency and time in the same channel. This is a hybrid combination of FDMA and TDMA. For
example, frequency hopping may be employed to ensure during each successive time slot, the
frequency bands assigned to the users are recorded in random manner. An important advantage of
CDMA over FDMA and TDMA is that it can provide for secure communication.

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Questions
1. What is a radio wave? What are the optical properties of radio waves? Explain all
the details of how they relate to radio wave propagation?

2. What is meant by a free space path loss of an electromagnetic wave? Give the
mathematical equation in decibel form. Determine, in dB, the free space path loss
for a frequency of 6 GHz travelling a distance of 50 km.

3. What are the three modes of terrestrial propagation of electromagnetic waves?

Explain.

4. What is a satellite multiple accessing arrangement? List and describe, in detail with
neat diagrams, the three forms of satellite multiple accessing arrangements.

5. Explain the term skip distance, satellite footprint and give the advantages of
geosynchronous satellites

6. List the advantages and disadvantages of microwave communications over cable

transmission facilities.

7. Compare FDMA, TDMA

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MODULE-V

Part - A
DATA COMMUNICATION CODES, ERROR CONTROL, AND DATA
FORMATS:

Data Communication Character Codes

Bar Codes
Error Control
Error Detection
Error Correction
Character Synchronization

Part - B
DATA COMMUNICATION EQUIPMENT:

Digital Service Unit and Channel Service Unit

Voice-Band Data Communication Modems
Bell Systems- Compatible Voice- Band Modems
Voice-Band Modern Block Diagram
Voice- Band Modem Classifications
Asynchronous Voice-Band Modems
Synchronous Voice-Band Modems
Modem Synchronization, 56K Modems
Modem Control: The AT Command Set, Cable Modems

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DATA COMMUNICATION CODES

DATA CODES

This refers to the way in which data is represented. The sender and receiver must use the same code in order to
communicate properly. Here, we will briefly look at two common codes, one which was developed earlier on
and was widely used in early telegraph systems, and the other, which is in widespread use today.

THE BAUDOT CODE:

The Baudot code was used extensively in telegraph systems. It is a five bit code invented by the Frenchman
Emile Baudot in 1870. Using five bits allowed 32 different characters. To accommodate all the letters of the
alphabet and numerals, two of the 32 combinations were used to select alternate character sets.

Each character is preceded by a start bit, and followed by a stop bit. It is an asynchronous code, and thus suited
for low speed data communication.

For instance, let’s consider coding the phase “JAMES BOND 007 SAYS HI!” using the Baudot code. To switch
between the LTRs and FIGs requires the use of a LetterShift or a FigureShift. Once switched, you stay in that
mode till you want to switch back again. So, here is the phrase encoded in Baudot.

J A M E S B O N D 0 0 7 S A Y S H I !
3 2 3 2
1 2 2 2 1 7 2 1 2 2 7
3 1 5 4 9 4 7 4 3 5 4 6
1 8 5 4 2 2 2 1 0 1
1 2 5 3

ASCII (American Standard Code for Information Interchange)

The ASCII code is the most popular code for serial data communications today. It is a seven bit code (128
combinations), and thus supports upper and lowercase characters, numeric digits, punctuation symbols, and
special codes. The table below lists the values for each character in the ASCII set.

0 0 0 0 0 0 0 0 0 0 0 0 0
07 09 0E
0 1 2 3 4 5 6 8 A B C D F
N S S E E E A B T
B L V F C S
00 U O T T O N C E A SO
S F T F R I
L H X X T Q K L B
D D D D D N S E C S E
E F G U
10 L C C C C A Y T A U S RS
M S S S
E 1 2 3 4 K N B N B C
20 ! “ # $ % & ‘ ( ) * + , – . /
30 0 1 2 3 4 5 6 7 8 9 : ; < = > ?

40 @ A B C D E F G H I J K L M N O
50 P Q R S T U V W X Y Z [ \ ] ^ _
60 ` a b c d e f g h i j k l m n o
D
70 p q r s t u v w x y z { | } ~ E
L

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To work out a particular value from the table, you first determine the row value, and then add the
column value. For example, the character A has a value of 41, being a row value of 40 and a column
value of 1.

ASCII is also used as the data code for keyboards in computers. Control Codes have values between
00 and 1F (hexadecimal). Control codes are used in binary synchronous communication, and device
control codes in communicating with devices such as printers or terminals.

A control code can be generated from a keyboard by holding down the Ctrl key and pressing another
key. For instance, holding down the Ctrl key and pressing the A key generates the control code SOH.

Bar Codes:

A bar code (often seen as a single word, barcode) is the small image of lines (bars) and spaces that is
affixed to retail store items, identification cards, and postal mail to identify a particular product
number, person, or location. The code uses a sequence of vertical bars and spaces to represent numbers
and other symbols. A bar code symbol typically consists of five parts: a quiet zone, a start character,
data characters (including an optional check character), a stop character, and another quiet zone.

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Error Control

Networks must be able to transfer data from one device to another with complete accuracy.
Data can be corrupted during transmission. For reliable communication, errors must be detected and
corrected. Error control can be divided into two general categories:

1. Error Detection
2. Error Correction

Error detection and Error correction are implemented either at the data link layer or the transport
layer of the OSI model.

Types of Errors

Single-bit error:

Single bit errors are the least likely type of errors in serial data transmission because the noise must
have a very short duration which is very rare. However this kind of errors can happen in parallel
transmission.

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Burst error:

The term burst error means that two or more bits in the data unit have changed from 1 to 0 or from 0
to 1.

Burst errors does not necessarily mean that the errors occur in consecutive bits, the length of the
burst is measured from the first corrupted bit to the last corrupted bit. Some bits in between may not
have been corrupted.

Burst error is most likely to happen in serial transmission since the duration of noise is normally
longer than the duration of a bit. The number of bits affected depends on the data rate and duration of
noise.

Error detection
Error detection means to decide whether the received data is correct or not without having a copy
of the original message. Error detection uses the concept of redundancy, which means adding extra
bits for detecting errors at the destination.

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Redundancy Checks Types:

(1). VRC - Vertical Redundancy Check

(2). LRC - Longitudinal Redundancy Check
(3). CRC - Cyclic Redundancy Check
(4). Checksum

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VRC - Vertical Redundancy Check

Fig. VRC-Vertical Redundancy Check

VRC is also referred to as character parity. With character parity, each character has its own error-
detection bit called the parity bit. The parity bit is considered as a redundant bit. An n-character
message would have 'n' redundant parity bits.
• It can detect single bit error.
• It can detect burst errors only if the total number of errors is odd.

LRC - Longitudinal Redundancy Check

LRC is also referred to as message parity since it is used to check error occurred within a
message. With LRC each bit position has a parity bit. LRC is the result of XORing the bits present in
all the characters present in a message whereas VRC is the result of XORing the bits within a single
character.

• In LRC even parity is generally used, whereas with VRC odd parity is generally used.
• LCR increases the likelihood of detecting burst errors.
• If two bits in one data units are damaged and two bits in exactly the same positions in another
data unit are also damaged, the LRC checker will not detect an error.

Fig.LRC - Longitudinal Redundancy Check

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Fig. VRC and LRC

Checksum
The characters within a message are combined together to produce an error-checking character
called as checksum, which can be as simple as the arithmetic sum of the numerical values of all the
characters in the message. The checksum is appended to the end of the message.
The receiver replicates the combining operation and determines its own checksum. The receivers
checksum is compared with transmitter checksum appended with the message, and if they are the
same, it is assumed that no transmission errors have occurred.

CRC - Cyclic Redundancy Check

Fig. Cyclic Redundancy Check

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G (x )
=Q( x)+R ( x)
P ( x)
G(x) = Message Polynomial (message or Data) P(x) = Generator Polynomial
Q(x) = Quotient
R(x) = Remainder (CRC bits) For this example

G( x)=x 5 +x2
P (x )=x 3 + x2 + x0
G(x) ={1,0,0,1,0,0}

P (x )={1,1,0,1}
Here in this example CRC bits are {0, 0, 1}

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Error Correction:
Error correction is the detection of errors and reconstruction of the original, error-free data.
Error correction may generally be realized in two different ways:

• Automatic repeat request (ARQ) [Retransmission method] (sometimes also referred to as

backward error correction): This is an error control technique whereby an error detection
scheme is combined with requests for retransmission of erroneous data. Every block of data
received is checked using the error detection code used, and if the check fails, retransmission of
the data is requested – this may be done repeatedly, until the data can be verified.
• Forward error correction (FEC): The sender encodes the data using an error-correcting code
(ECC) prior to transmission. The additional information (redundancy) added by the code is used
by the receiver to recover the original data. In general, the reconstructed data is what is deemed
the "most likely" original data.

ARQ and FEC may be combined, such that minor errors are corrected without retransmission, and
major errors are corrected via a request for retransmission: this is called hybrid automatic repeat-
request (HARQ).

Forward Error Correction- Example Hamming Code.

Hamming Code:
Hamming code is an error-correcting code used for correcting transmission errors in synchronous data
streams. The Hamming code will correct only single-bit errors. It cannot correct mulitple-bit errors.
Hamming bits also sometimes called as error bits are inserted in to a character at random manner.
The combination of data bits (m bits) and Hamming bits(n bits) called as Hamming code (m+n bits).

To correct an error, the receiver reverses the value of the altered bit. To do so, it must know
which bit is in error.

Number of redundancy bits (Hamming bits) 'n' needed: Let data bits = m
Redundancy bits = n
Total message sent = m+n
The value of 'n' must satisfy the following relation:

2n ≥ m+n+1

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DATA COMMUNICATION EQUIPMENT

Digital Service Unit and Channel Service Unit:

A CSU/DSU (Channel Service Unit/Data Service Unit) is a hardware device that converts a digital
data frame from the communications technology used on a local area network (LAN) into a frame
appropriate to a wide-area network (WAN) and vice versa. Think of it as a high end modem which is
used to connect a data terminal equipment (DTE), such as a router, to a digital circuit, such as a Digital
Signal 1 (T1) line.

For example, if you have a Web business from your own home and have leased a digital line (perhaps
a T-1 or fractional T-1 line) to a phone company or a gateway at an Internet service provider, you have
a CSU/DSU at your end, and the phone company or gateway host has a CSU/DSU at its end, and the
units at both ends must be set to the same communications standard.

Voice-Band Data Communication Modems:

Modem, (from “modulator/demodulator”), any of a class of electronic devices that convert digital data
signals into modulated analog signals suitable for transmission over analog telecommunications
circuits. A modem also receives modulated signals and demodulates them, recovering the digital signal
for use by the data equipment. Modems thus make it possible for established telecommunications
media to support a wide variety of data communication, such as e-mail between personal computers,
facsimile transmission between fax machines, or the downloading of audio-video files from a World
Wide Web server to a home computer.

Most modems are “voiceband”; i.e., they enable digital terminal equipment to communicate over
telephone channels, which are designed around the narrow bandwidth requirements of the human
voice. Cable modems, on the other hand, support the transmission of data over hybrid fibre-coaxial
channels, which were originally designed to provide high-bandwidth television service. Both
voiceband and cable modems are marketed as freestanding, book-sized modules that plug into a
telephone or cable outlet and a port on a personal computer. In addition, voiceband modems are
installed as circuit boards directly into computers and fax machines. They are also available as small
card-sized units that plug into laptop computers.

Modems operate in part by communicating with each other, and to do this they must follow matching
protocols, or operating standards. Worldwide standards for voiceband modems are established by the
V-series of recommendations published by the Telecommunication Standardization sector of the
International Telecommunication Union (ITU). Among other functions, these standards establish the
signaling by which modems initiate and terminate communication, establish compatible modulation
and encoding schemes, and arrive at identical transmission speeds. Modems have the ability to “fall
back” to lower speeds in order to accommodate slower modems. “Full-duplex” standards allow
simultaneous transmission and reception, which is necessary for interactive communication. “Half-
duplex” standards also allow two-way communication, but not simultaneously; such modems are
sufficient for facsimile transmission.

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Data signals consist of multiple alternations between two values, represented by the binary digits, or
bits, 0 and 1. Analog signals, on the other hand, consist of time-varying, wavelike fluctuations in
value, much like the tones of the human voice. In order to represent binary data, the fluctuating values
of the analog wave (i.e., its frequency, amplitude, and phase) must be modified, or modulated, in such
a manner as to represent the sequences of bits that make up the data signal. Modems employ a number
of methods to do this; they are noted below in the section.

Each modified element of the modulated carrier wave (for instance, a shift from one frequency to
another or a shift between two phases) is known as a baud. In early voiceband modems beginning in
the early 1960s, one baud represented one bit, so that a modem operating, for instance, at 300 bauds
per second (or, more simply, 300 baud) transmitted data at 300 bits per second. In modern modems a
baud can represent many bits, so that the more accurate measure of transmission rate is bits or kilobits
(thousand bits) per second.
During the course of their development, modems have risen in throughput from 300 bits per second
(bps) to 56 kilobits per second (Kbps) and beyond. Cable modems achieve a throughput of several
megabits per second (Mbps; million bits per second). At the highest bit rates, channel-encoding
schemes must be employed in order to reduce transmission errors. In addition, various source-
encoding schemes can be used to “compress” the data into fewer bits, increasing the rate of
information transmission without raising the bit rate.

Development of voiceband modems

The first generation

Although not strictly related to digital data communication, early work on telephotography machines
(predecessors of modern fax machines) by the Bell System during the 1930s did lead to methods for
overcoming certain signal impairments inherent in telephone circuits. Among these developments were
equalization methods for overcoming the smearing of fax signals as well as methods for translating fax
signals to a 1,800-hertz carrier signal that could be transmitted over the telephone line.

The first development efforts on digital modems appear to have stemmed from the need to transmit
data for North American air defense during the 1950s. By the end of that decade, data was being
transmitted at 750 bits per second over conventional telephone circuits. The first modem to be made
commercially available in the United States was the Bell 103 modem, introduced in 1962 by the
American Telephone & Telegraph Company (AT&T).

The Bell 103 permitted full-duplex data transmission over conventional telephone circuits at data rates
up to 300 bits per second. In order to send and receive binary data over the telephone circuit, two pairs
of frequencies (one pair for each direction) were employed. A binary 1 was signaled by a shift to one
frequency of a pair, while a binary 0 was signaled by a shift to the other frequency of the pair. This
type of digital modulation is known as frequency-shift keying, or FSK. Another modem, known as the
Bell 212, was introduced shortly after the Bell 103.

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Transmitting data at a rate of 1,200 bits, or 1.2 kilobits, per second over full-duplex telephone circuits,
the Bell 212 made use of phase-shift keying, or PSK, to modulate a 1,800-hertz carrier signal. In PSK,
data is represented as phase shifts of a single carrier signal. Thus, a binary 1 might be sent as a zero-
degree phase shift, while a binary 0 might be sent as a 180-degree phase shift.

Between 1965 and 1980, significant efforts were put into developing modems capable of even higher
transmission rates. These efforts focused on overcoming the various telephone line impairments that
directly limited data transmission. In 1965 Robert Lucky at Bell Laboratories developed an automatic
adaptive equalizer to compensate for the smearing of data symbols into one another because of
imperfect transmission over the telephone circuit. Although the concept of equalization was well
known and had been applied to telephone lines and cables for many years, older equalizers were fixed
and often manually adjusted.

The advent of the automatic equalizer permitted the transmission of data at high rates over the public
switched telephone network (PSTN) without any human intervention. Moreover, while adaptive
equalization methods compensated for imperfections within the nominal three-kilohertz bandwidth of
the voice circuit, advanced modulation methods permitted transmission at still higher data rates over
this bandwidth.

One important modulation method was quadrature amplitude modulation, or QAM. In QAM, binary
digits are conveyed as discrete amplitudes in two phases of the electromagnetic wave, each phase
being shifted by 90 degrees with respect to the other. The frequency of the carrier signal was in the
range of 1,800 to 2,400 hertz. QAM and adaptive equalization permitted data transmission of 9.6
kilobits per second over four-wire circuits. Further improvements in modem technology followed, so
that by 1980 there existed commercially available first-generation modems that could transmit at 14.4
kilobits per second over four-wire leased lines.

In mid-1990 the CCITT began to consider the possibility of full-duplex transmission over the PSTN at
even higher rates than those allowed by the upgraded V.32 standard. This work resulted in the issuance
in 1994 of the V.34 modem standard, allowing transmission at 28.8 kilobits per second.

The second generation

Beginning in 1980, a concerted effort was made by the International Telegraph and Telephone
Consultative Committee (CCITT; a predecessor of the ITU) to define a new standard for modems that
would permit full-duplex data transmission at 9.6 kilobits per second over a single-pair circuit
operating over the PSTN.

Two breakthroughs were required in this effort. First, in order to fit high-speed full-duplex data
transmission over a single telephone circuit, echo cancellation technology was required so that the
sending modem’s transmitted signal would not be picked up by its own receiver.

Second, in order to permit operation of the new standard over unconditioned PSTN circuits, a new
form of coded modulation was developed.

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In coded modulation, error-correcting codes form an integral part of the modulation process, making
the signal less susceptible to noise. The first modem standard to incorporate both of these technology
breakthroughs was the V.32 standard, issued in 1984. This standard employed a form of coded
modulation known as trellis-coded modulation, or TCM. Seven years later an upgraded V.32 standard
was issued, permitting 14.4-kilobit-per-second full-duplex data transmission over a single PSTN
circuit.

The third generation

The engineering of modems from the Bell 103 to the V.34 standard was based on the assumption that
transmission of data over the PSTN meant analog transmission—i.e., that the PSTN was a circuit-
switched network employing analog elements. The theoretical maximum capacity of such a network
was estimated to be approximately 30 Kbps, so the V.34 standard was about the best that could be
achieved by voice band modems.

In fact, the PSTN evolved from a purely analog network using analog switches and analog
transmission methods to a hybrid network consisting of digital switches, a digital “backbone” (long-
distance trunks usually consisting of optical fibers), and an analog “local loop” (the connection from
the central office to the customer’s premises). Furthermore, many Internet service providers (ISPs) and
other data services access the PSTN over a purely digital connection, usually via a T1 or T3 wire or an
optical-fiber cable.

With analog transmission occurring in only one local loop, transmission of modem signals at rates
higher than 28.8 Kbps is possible. In the mid-1990s several researchers noted that data rates up to 56
Kbps downstream and 33.6 Kbps upstream could be supported over the PSTN without any data
compression. This rate for upstream (subscriber to central office) transmissions only required
conventional QAM using the V.34 standard. The higher rate in the downstream direction (that is, from
central office to subscriber), however, required that the signals undergo “spectral shaping” (altering the
frequency domain representation to match the frequency impairments of the channel) in order to
minimize attenuation and distortion at low frequencies.

In 1998 the ITU adopted the V.90 standard for 56-Kbps modems. Because various regulations and
channel impairments can limit actual bit rates, all V.90 modems are “rate adaptive.” Finally, in 2000
the V.92 modem standard was adopted by the ITU, offering improvements in the upstream data rate
over the V.90 standard. The V.92 standard made use of the fact that, for dial-up connections to ISPs,
the loop is essentially digital. Through the use of a concept known as preceding, which essentially
equalizes the channel at the transmitter end rather than at the receiver end, the upstream data rate was
increased to above 40 Kbps. The downstream data path in the V.92 standard remained the same 56
Kbps of the V.90 standard

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Cable modems
A cable modem connects to a cable television system at the subscriber’s premises and enables two-
way transmission of data over the cable system, generally to an Internet service provider (ISP). The
cable modem is usually connected to a personal computer or router using an Ethernet connection that
operates at line speeds of 10 or 100 Mbps. At the “head end,” or central distribution point of the cable
system, a cable modem termination system (CMTS) connects the cable television network to the
Internet. Because cable modem systems operate simultaneously with cable television systems, the
upstream (subscriber to CMTS) and downstream (CMTS to subscriber) frequencies must be selected to
prevent interference with the television signals.

Two-way capability was fairly rare in cable services until the mid-1990s, when the popularity of the
Internet increased substantially and there was significant consolidation of operators in the cable
television industry. Cable modems were introduced into the marketplace in 1995. At first all were
incompatible with one another, but with the consolidation of cable operators the need for a standard
arose. In North and South America a consortium of operators developed the Data Over Cable Service
Interface Specification (DOCSIS) in 1997. The DOCSIS 1.0 standard provided basic two-way data
service at 27–56 Mbps downstream and up to 3 Mbps upstream for a single user. The first DOCSIS 1.0
modems became available in 1999. The DOCSIS 1.1 standard released that same year added voice
over Internet protocol (VoIP) capability, thereby permitting telephone communication over cable
television systems. DOCSIS 2.0, released in 2002 and standardized by the ITU as J.122, offers
improved upstream data rates on the order of 30 Mbps.

All DOCSIS 1.0 cable modems use QAM in a six-megahertz television channel for the downstream.
Data is sent continuously and is received by all cable modems on the hybrid coaxial-fibre branch.
Upstream data is transmitted in bursts, using either QAM or quadrature phase-shift keying (QPSK)
modulation in a two-megahertz channel. In phase-shift keying (PSK), digital signals are transmitted by
changing the phase of the carrier signal in accordance with the transmitted information. In binary
phase-shift keying, the carrier takes on the phases +90° and −90° to transmit one bit of information; in
QPSK, the carrier takes on the phases +45°, +135°, −45°, and −135° to transmit two bits of
information. Because a cable branch is a shared channel, all users must share the total available
bandwidth. As a result, the actual throughput rate of a cable modem is a function of total traffic on the
branch; that is, as more subscribers use the system, total throughput per user is reduced. Cable
operators can accommodate greater amounts of data traffic on their networks by reducing the total span
of a single fibre-coaxial branch.

DSL modems
In the section Development of voice band modems, it is noted that the maximum data rate that can be
transmitted over the local telephone loop is about 56 Kbps. This assumes that the local loop is to be
used only for direct access to the long-distance PSTN. However, if digital information is intended to be
switched not through the telephone network but rather over other networks, then much higher data
rates may be transmitted over the local loop using purely digital methods. These purely digital methods
are known collectively as digital subscriber line (DSL) systems. DSL systems carry digital signals over
the twisted-pair local loop using methods analogous to those used in the T1 digital carrier system to
transmit 1.544 Mbps in one direction through the telephone network.

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The first DSL was the Integrated Services Digital Network (ISDN), developed during the 1980s. In
ISDN systems a 160-Kbps signal is transmitted over the local loop using a four-level signal format
known as 2B1Q, for “two bits per quaternary signal.” The 160-Kbps signal is broken into two “B”
channels of 64 Kbps each, one “D” channel of 16 Kbps, and one signaling channel of 16 Kbps to
permit both ends of the ISDN local loop to be initialized and synchronized. ISDN systems are
deployed in many parts of the world. In many cases they are used to provide digital telephone services,
although these systems may also provide 64-Kbps or 128-Kbps access to the Internet with the use of
an adapter card. However, because such data rates are not significantly higher than those offered by
56-Kbps V.90 voiceband modems, ISDN is not widely used for Internet access.

High-bit-rate DSL, or HDSL, was developed in about 1990, employing some of the same technology
as ISDN. HDSL uses 2B1Q modulation to transmit up to 1.544 Mbps over two twisted-pair lines. In
practice, HDSL systems are used to provide users with low-cost T1-type access to the telephone
central office. Both ISDN and HDSL systems are symmetric; i.e., the upstream and downstream data
rates are identical.

Asymmetric DSL, or ADSL, was developed in the early 1990s, originally for video-on-demand
services over the telephone local loop. Unlike HDSL or ISDN, ADSL is designed to provide higher
data rates downstream than upstream—hence the designation “asymmetric.” In general, downstream
rates range from 1.5 to 9 Mbps and upstream rates from 16 to 640 Kbps, using a single twisted-pair
wire. ADSL systems are currently most often used for high-speed access to an Internet service
provider (ISP), though regular telephone service is also provided simultaneously with the data service.

At the local telephone office, a DSL access multiplexer, or DSLAM, statistically multiplexes the data
packets transmitted over the ADSL system in order to provide a more efficient link to the Internet. At
the customer’s premises, an ADSL modem usually provides one or more Ethernet jacks capable of line
rates of either 10 Mbps or 100 Mbps.

In 1999 the ITU standardized two ADSL systems. The first system, designated G.991.1 or G.DMT,
specifies data delivery at rates up to 8 Mbps on the downstream and 864 Kbps on the upstream. The
modulation method is known as discrete multitone (DMT), a method in which data is sent over a large
number of small individual carriers, each of which uses QAM modulation (described above in
Development of voiceband modems). By varying the number of carriers actually used, DMT
modulation may be made rate-adaptive, depending upon the channel conditions. G.991.1 systems
require the use of a “splitter” at the customer’s premises to filter and separate the analog voice channel
from the high-speed data channel. Usually the splitter has to be installed by a technician; to avoid this
expense a second ADSL standard was developed, variously known as G.991.2, G.lite, or splitterless
ADSL. This second standard also uses DMT modulation to achieve the same rates as G.991.1. In place
of the splitter, user-installable filters are required for each telephone set in the home.

Unlike cable modems, ADSL modems use a dedicated telephone line between the customer and the
central office, so the delivered bandwidth equals the bandwidth actually available. However, ADSL
systems may be installed only on local loops less than 5,400 metres (18,000 feet) long and therefore
are not available to homes located farther from a central office. Other versions of DSL have been
announced to provide even higher rate services over shorter local loops. For instance, very high data
rate DSL, or VDSL, can provide up to 15 Mbps over a single twisted wire pair up to 1,500 metres
(5,000 feet) long.

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