UNIT – I

Syllabus:
Introduction to Data Communication, Network, Protocols & standards and standards
organizations – Line Configuration – Topology – Transmission mode – Classification
of Network – OSI Model – Layers of OSI Model.
Introduction to Data Communication
Data:

Data is a collection of information produced and consumed in the form of digital

signals

Communication:

The process of sharing information between one or more devices or persons.

DATA COMMUNICATION:

In its simplest form, data communication takes place between two devices
that are directly connected by some form of point-to-point transmission medium.
Data communications are the exchange of data between two devices via some
form of transmission medium such as a wire cable. For data communications to occur,
the communicating devices must be pail of a Communication system made up of a
cornbination of hardware (physical equipment) and software (programs).
The effectiveness of a data Communications system depends on four
fundamental characteristics: delivery, accuracy, timeliness, and jitter.
1. Delivery.
The system must deliver data to the correct destination. Data must be received
by the intended device or user and only by that device or user.
2. Accuracy.
The system must deliver the data accurately. Data that have been altered in
transmission and left uncorrected are unusable.
3. Timeliness.
The system must deliver data in a timely manner. Data delivered late are
useless. In the case of video and audio, timely delivery means delivering data as
they are produced, in the same order that they are produced, and without significant
delay. This kind of delivery is called real-time transmission.

4. Jitter.

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 Jitter refers to the variation in the packet arrival time. It is the uneven delay in
the delivery of audio or video packets. For exarnple, let us assume that video
packets arc sent every 30 ms. If some of the packets arrive with 30-ms delay and
others with 40-ms delay, an uneven quality in the video is the result.

Components

A data communications system has five components (see Fig 1.1).

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. Sonic examples of transmission media include twisted-pair wire,
coaxial cable, fiber-optic cable, and radio waves.
5. Protocol.
A protocol is a set of sents an agreement between the devices may be connected
but not cannot be understood by a person rules that govern data communications. It
reprecommunicating devices without a protocol, two communicating, just as a
person speaking French who speaks only Japanese.

Data Representation

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 Information today comes in different forms such as text, numbers, images,
audio, and video.

Fig 1.2

Data Flow

Communication between two devices can be simplex, half-duplex, or full-

duplex as shown in Figure 1.2.

Simplex

In simplex mode, the communication js unidirectional, as on a one-way street,

Only one of the two devices on a link can transmit; the other can only receive (see
Figure 1.2a).

Keyboards and traditional monitors are examples of simplex devices. The

keyboard can only introduce input; the monitor can only accept output. The simplex
mode can use the entire capacity of the channel to send data in one direction.

Half-Duplex

In half-duplex 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 (sec
Figure 1.2b).

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 The half-duplex mode is like a one-lane road with traffic allowed in both
threetions. When cars are traveling in one direction, cars going the other way must
wait. In a half-duplex transmission, the entire capacity of a channel is taken over by
whichever of the two devices is transmitting at the time. Walkie-talkies and CB
(citizens band) radios are both half-duplex systems.

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 tr each direction.

Full-Duplex

In full-duplex mode (also called duplex), both stations can transmit and
receive simultaneously (see Figure 1.2c).

The full-duplex mode is like a two-way street with traffic flowing in both
directions at the same time. In full-duplex mode, signals going in one direction share
the capacity of the link with signals going in the other direction. This sharing can occur
in two ways: Either the link must contain two physically separate transmission paths,
one for sending and the other for receiving; or the capacity of the channe1 is divided
between signals traveling in both directions.

One common example of full-duplex communication is the telephone network.

When two people are communicating by a telephone line, both can talk and listen at the
same time.

The full-duplex mode is used when communication in both directions is

required all the time. The capacity of the channel, however, must be divided between
the two directions.

Network

A computer network allows sharing of resources and information among

devices connected to the network. The Advanced Research Projects Agency (ARPA)
funded the design of the "Advanced Research Projects Agency Network"
(ARPANET) for the United States Department of Defense. It was the first operational
computer network in the world. Development of the network began in 1969, based on
designs developed during the 1960s.

A computer network is a group of computers that are connected to each other

for the purpose of communication. Networks may be classified according to a wide
variety of characteristics.

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 A network is a set of devices (often referred to as nodes) connected by
communication links. A node can be a computer, printer, or any other device capable
of sending and/or receiving data generated by other nodes on the network.

Distributed Processing

Most networks use distributed processing, in which a task is divided among

multiple computers. Instead of one single large machine being responsible for all
aspects of a process, separate computers (usually a personal computer or workstation)
handle a subset.

Network Criteria

A network must be able to meet a certain number of criteria. The most

important of these are performance, reliability, and security.

Performance

Performance can be measured in many ways, including transit time and

response time. Transit time is the amount of time required for a message to travel from
one device to another. Response time is the elapsed time between an inquiry and a
response. The performance of a network depends on a number of factors, including the
number of users, the type of transmission medium, the capabilities of the connected
hardware, and the efficiency of the software.

Performance is often evaluated by two networking metrics: throughput and

delay. We often need more throughput and less delay. However, these two criteria are
often contradictory. If we try to send more data to the network, we may increase
throughput but we increase the delay because of traffic congestion in the network.

Reliability

In addition to accuracy of delivery, network reliability is measured by the

frequency of failure, the time it takes a link to recover from a failure, and the network‘s
robustness in a catastrophe.

Security

Network security issues include protecting data from unauthorized access,

protecting data from damage and development, and implementing policies and
procedures for recovery from breaches and data losses.

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 Physica1 Structures

Before discussing networks, we need to define some network attributes.

Type of Connection

A network is two or more devices connected through links. A link is a

communications pathway that transfers data from one device to another. For
visualization purposes, it is simplest to imagine any link as a line drawn between two
points. For communication to occur, two devices must be connected in some way to the
same link at the same time.

There are two possible types of connections: point-to-point and multipoint,

Point-to-Point:

A point-to-point connection provides a dedicated link between two devices.

The entire capacity of the link is reserved for transmission between those two devices.
Most point-to-point connections use an actual length of wire or cable to con- fleet the
two ends, but other options, such as microwave or satellite links, are also possible (see
Figure 1 .3a). When you change television channels by infrared remote control, you are
establishing a point-to-point connection between the remote control and the
television‘s control system.

Multipoint:

A multipoint (also called multidrop) connection is one in which more than

two specific devices share a single link (see Figure 1.3b).

In a multipoint environment, the capacity of the channel is shared, either

spatially or temporally. If several devices can use the link simultaneously, it is a
spatially shared connection. If users must take turns, it is a timeshared connection.

Protocols & standards and standards organizations

Protocol (computing)

In computing, a protocol is a set of rules which is used by computers to

communicate with each other across a network. A protocol is a convention or standard
that controls or enables the connection, communication, and data transfer between
computing endpoints. In its simplest form, a protocol can be defined as the rules
governing the syntax, semantics, and synchronization of communication. Protocols
may be implemented by hardware, software, or a combination of the two. At the lowest
level, a protocol defines the behavior of a hardware connection.

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 de facto standard

A protocol that has not been approved by an organized body but adopted as a
standard through widespread use.

de jure standard

A protocol that has been legislated by an officially recognized body.

Common protocols

 IP (Internet Protocol)
 UDP (User Datagram Protocol)
 TCP (Transmission Control Protocol)
 DHCP (Dynamic Host Configuration Protocol)
 HTTP (Hypertext Transfer Protocol)
 FTP (File Transfer Protocol)
 Telnet (Telnet Remote Protocol)
 SSH (Secure Shell Remote Protocol)
 POP3 (Post Office Protocol 3)
 SMTP (Simple Mail Transfer Protocol)
 IMAP (Internet Message Access Protocol)
 SOAP (Simple Object Access Protocol)
 PPP (Point-to-Point Protocol)
 RFB (Remote Framebuffer Protocol)

Protocols
In computer networks, communication occurs between entities in different
systems. An entity is anything capable of sending or receiving information. However,
two entities cannot simply send bit streams to each other and expect to be understxxl.
For communication to occur, the entities must agree on a protocol. A protocol is a set
of rules that govern data communications. A protocol defines what is communicated,
how it is communicated, and when it is communicated. The key elements of a protocol
are syntax, semantics, and timing.

Syntax:

The term syntax refers to the structure or format of the data, meaning the order
in which they are presented. For example, a simple protocol might expect the first 8
bits of data to be the address of the sender, the second S bits to be the address of the
receiver, and the rest of the stream to be the message itself.

Semantics:

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 The word semantics refers to the meaning of each section of bits. How is a
particular pattern to be interpreted, and what action is to be taken based on that
interpretation? For example, does an address identify the route to be taken or the final
destination of the message?

Timing:

The term timing refers to two characteristics: when data should be sent and how
fast they can be sent. For example, if a sender produces data at 100 Mhps hut the
receiver can process data at only 1 Mbps. the transmission will overload the receiver
and some data will be lost.

Standards and standards organizations

 American National Standards Institute (ANSI)

 International Electro-technical Commission (IEC)
 International Telecommunication Union (ITU)
 Institute of Electrical and Electronics Engineers (IEEE)
 International Organization for Standardization (ISO)
 Internet Society (ISOC) and the associated Internet Engineering Task Force
(IETF)
 Electronic Industries Alliance (EIA) and the associated Telecommunictions
Industry Association (TIA)

 American National Standards Institute (ANSI)

ANSI is a private, non-governmental agency where members are

manufacturers, users and other interested companies. It has nearly 1000 member of the
ISO (International Standard Organization). ANSI has set up the standards for Fiber
Distributed Data Interface (FDDI) and for local are networks using optical fiber. ANSI
has also set up the American Standard Code for Information Interchanged (ASCII),
used by many computers for storing information.

 International Electro-technical Commission (IEC)

IEC is a non-governmental agency devising standards for data processing and

interconnections and safety in office equipment. It was involved in the development of
the Joint Photographic Experts Group (JPEG), a group that devised compression
standard for images.

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  International Telecommunication Union (ITU)

ITU is an agency of the United Nations and has three sectors.

1) ITU-R deals with radio communications.

2) ITU-D is a development sector.
3) ITU-T deals with telecommunications

International Telecommunications Union sets standards for modems, e-mail,

and digital telephone systems. The ITU has contributed to the following standards.

 Institute of Electrical and Electronics Engineers (IEEE)

The IEEE is the largest professional organization in the world and consists of
computing and engineering professionals. It is involved in developing standards for
computing, communication, and for processes in electrical engineering, and
electronics. It sponsored an important standard for local area networks called Project
802.

 International Organization for Standardization (ISO)

The ISO is a non-governmental organization based in Geneva, Switzerland, in

which over 100 countries participate. One of ISOs most significant activities is its work
on open systems, which define the protocols that would allow any two computers to
communicate independent of their architecture. Open Systems Interconnections (OSI)
model, contains seven layer protocols for network communications.

 Internet Society (ISOC) and the associated Internet Engineering Task Force
(IETF)

Internet Society and the associated Internet Engineering Task Force are
concerned with expediting the growth and in the evaluation of Internet
communications. The Internet Society concentrates on users issues, including
enhancements to the TCP/IP protocol suite. IETF focuses on technical Internet issues
(hardware and software). Important contributions include the development of Simple
Network Management Protocol (SNMP)

 Electronic Industries Alliance (EIA) and the associated Telecommunictions

Industry Association (TIA)

EIA is responsible to develop network cabling standards. EIA has made

significant contributions by defining physical connection interfaces and electronic
signaling specifications for data communications. TIA was created as a separate body
within the EIA to develop telecommunications and cabling standards.

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 Line Configuration

Line configuration refers to the way two or more communication devices

attached to a link. Line configuration is also referred to as connection. A link is a
communication medium through which data is communicated between devices. For
communication to occur between two devices, they must be connected to the same link
at the same time. There are two possible types of line configurations or connections.
These connections are.

1. Point-to-point connection
2. Multipoint connection

Point-to-Point Connector

The point-to-point connection provides a dedicated link between two

communication devices. The entire link or channel is reserved for two devices for data
communication, and no other devices can use the dedicated link. Usually, in this type
of connection, the two devices are connected together with a cable.

It must be noted that microwave and satellite dedicated links are also possible.
Two computers connected together (point-to-point) through microwave link.

When you change the television channel by remote control, you are establishing
a point-to-point connection between the remote control and the television‘s control
system.

Multipoint Connection

Multipoint connection is also referred to as multidrop connection. This type

of connection allows multiple devices (more than two devices) to share a single
link. The multipoint connection or line configuration is shown below.

Fig 1.3

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 Topology

In networking, the term topology is the way of connecting computers or nodes

on a network. There are many ways in which computers are connected together in a
computer network. Therefore network topology is defined as: the schemes of joining a
number of computers in the form of a network are called Network Topologies.

Computer networks may be classified according to the network topology upon

which the network is based, such as

bus network,
star network,
ring network,
mesh network,
star-bus network,
tree or hierarchical topology network.

Network topology signifies the way in which devices in the network see their
logical relations to one another. The use of the term "logical" here is significant. That
is, network topology is independent of the "physical" layout of the network. Even if
networked computers are physically placed in a linear arrangement, if they are
connected via a hub, the network has a Star topology, rather than a bus topology. In
this regard the visual and operational characteristics of a network are distinct; the
logical network topology is not necessarily the same as the physical layout. Networks
may be classified based on the method of data used to convey the data, these include
digital and analog networks.

We know that two or more devices are connected to a link for data
communication. Similarly, two or more links form a topology. The topology of a
network is the geometric representation of the relationship of all the links and the
nodes (communication devices) to one another.

There are three commonly used network topologies. These are:

1. Star topology
2. Ring topology
3. Bus topology

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 1. Star Topology

In a star network, each node (computer or other device) is directly connected to

the central computer or Hub that provides connection points for nodes on the network.
The star topology is the most common topology in use today. In star network,
information or data is communicated from one computer to another through Hub. This
form of network configuration looks like a star as shown in figure 1.4 below.

Fig 1.4
Advantages:
The main advantages of star topology are:
 It is easy to install and to maintain.
 You can easily add and remove nodes to and from the network without
affecting the network.
 If any node fails, other nodes are not affected.

Disadvantages
The main disadvantages of star topology are:
 This type of network depends upon the central Hub. If Hub fails the entire
network is failed.
 Each computer is directly connected to the Hub through a cable, so it becomes
more costly.

2. Ring Topology

In ring network, each node is connected to two adjacent nodes in the form a
closed ring or loop. In ring topology, the last node connects to the first node to
complete the ring. In ring topology, each node has a dedicated point-to-point
connection only with the two devices on either side of it.

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 In this network, data is communicated in one direction from node to node
around the entire ring. When a computer in ring network sends message to another
computer on the network, the message travels to each node or computer until it reaches
its destination. The ring network configuration is shown in figure 1.5 below.

Fig 1.5
Advantages
The main advantages of ring topology are:
 It is less expensive than star topology.
 Nodes can be easily added or removed.
Disadvantages
The main disadvantages of ring topology are:
 It is more difficult to install and maintain.
 If a node fails, it affects the entire network.

3. Bus Topology
In bus network, all nodes are connected to a common communication medium
or central cable. The central physical cable that connects the nodes is called Bus. The
data is communicated between nodes in both directions through bus. A bus topology
uses the multipoint connection. The central single cable (or bus) acts as backbone to
link all the devices to the network.
In bus network, when a computer sends a message to another computer it also
attaches the address of the destination computer. In bus topology, a special device
called a terminator is attached at the cable‘s start and end points. A terminator stops the
network signals.

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 In LAN, bus topology is mostly used. In this topology, each computer is
assigned a unique address. The bus network configuration is given in figure 1.6 below.

Fig 1.6

Complex topologies
1. Mesh Topology
In the mesh topology, separate cables are used to connect individual devices on
the network. This topology is expensive because of the number of cables used in the
network.
The mesh topology is of two types,
full-mesh and
partial-mesh.

a) Full mesh:
In this topology, each device is interconnected with all the devices on the
network, by a dedicated cable. If one device fails, the data traveling along the network
can be routed through another device attached to the active device. The structure of the
network is complex because the devices in the network are interconnected.

Fig 1.7 Full Mesh Partial Mesh

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 b) Partial Mesh
In this topology, each device on the network is not connected to other devices.
Only a few devices on the network are connected using the full-mesh topology, and the
others are connected to one or more devices on the network.

Hybrid Topology:
This topology is the combination of bus, star, and ring networks. In other
words, this topology combines multiple topologies to form a large topology. The
hybrid topology is widely implemented in WANs.

a) Hybrid Star-Bus Topology:

Fig shows two networks, A and B, on a star topology. However, the connection
between the two networks is established using the bus topology. In a star-bus topology,
the star topology of each network is linked to the bus topology.

Hybrid Star-Bus Topology

Hybrid star bus topology

Fig (1.8)

Hybrid star ring

topology

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 Transmission mode
A transmission mode defines the way in which group of bits goes from one
device to another. In transmission mode data flows in 3 ways.
1. Simplex 2. Half-duplex 3.Full-duplex.

There are 2 categories of transmission modes

1. Parallel transmission
2. Serial transmission

Parallel Transmission
Binary data 1s and 0s are organized into groups of n bits each. Bits are
transmitted simultaneously by using a separate line (wire) for each bit. Multiple bits are
sent with each clock tick.
Advantages:
 It is commonly used for data transmission
 Distances between two devices are short. (eg: communication between
computer and peripheral devices.
Disadvantages:
 Limited to short distances
 Very expensive

Fig (1.9) Parallel transmission

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 Serial Transmission:

Group of bits is transmitted one by one using line (wire) for all bits.

Advantages:
The advantage of serial over parallel transmission is that with only one
communication channel, serial transmission reduces the cost of transmission over
parallel by roughly a factor of n.
Since communication within devices is parallel, conversion devices are
required at the interface between the sender and the line (parallel-to-serial) and
between the line and the receiver (serial-to-parallel).
We can communicate to long distance and it is less expensive.
It has 2 ways to provide communications

1. Asynchronous
2. Synchronous

Fig (1.10) Serial Transmission

Asynchronous transmission mode

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 Fig 1.11

Bits are divided into small groups (bytes) and sent independently. The sender
can send the groups at any time and the receiver never knows when they will arrive.

We send one start bit (0) at the beginning and one stop bit (1) at end of each
byte.

There may be a gap between each byte. When the receiver detects a start bit, it
sets a timer and begins counting bits as they come in after receiving stop bit, it ignores
any received pulses.

Synchronous transmission mode:

Bit stream is combined into larger ―frames‖, which may contain multiple bytes.
We send bits one after another without start / stop bits or groups.
It is the responsibility of the receiver to group the bits.

Fig (1.12) synchronous transmission

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 Classification of Network

Types of networks

Below is a list of the most common types of computer networks in order of

scale.

Personal area network

A personal area network (PAN) is a computer network used for communication

among computer devices close to one person. Some examples of devices that are used
in a PAN are personal computers, printers, fax machines, telephones, scanners, and
even video game consoles. Such a PAN may include wired and wireless connections
between devices. The reach of a PAN is typically at least about 20-30 feet
(approximately 6-9 meters), but this is expected to increase with technology
improvements.

Local Area Networks

Local area networks, generally called LANs, are privately-owned networks
within a single building or campus of up to a few kilometers in size. They are widely
used to connect personal computers and workstations in company offices and factories
to share resources (e.g., printers) and exchange information.
LANs are distinguished from other kinds of networks by three characteristics:
(1) their size,
(2) their transmission technology, and
(3) their topology.
LANs are restricted in size, which means that the worst-case transmission time
is bounded and known in advance. Knowing this bound makes it possible to use certain
kinds of designs that would not otherwise be possible. It also simplifies network
management. LANs may use a transmission technology consisting of a cable to which
all the machines are attached, like the telephone company party lines once used in rural
areas. Traditional LANs run at speeds of 10 Mbps to 100 Mbps, have low delay
(microseconds or nanoseconds), and make very few errors. Newer LANs operate at up
to 10 Gbps. In this book, we will adhere to tradition and measure line speeds in
megabits/sec (1 Mbps is 1,000,000 bits/sec) and gigabits/sec (1 Gbps is 1,000,000,000
bits/sec). Various topologies are possible for broadcast LANs. Fig shows two of them.

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 In a bus (i.e., a linear cable) network, at any instant at most one machine is the master
and is allowed to transmit. All other machines are required to refrain from sending. An
arbitration mechanism is needed to resolve conflicts when two or more machines want
to transmit simultaneously. The arbitration mechanism may be centralized or
distributed. IEEE 802.3, popularly called Ethernet, for example, is a bus-based
broadcast network with decentralized control, usually operating at 10 Mbps to 10
Gbps. Computers on an Ethernet can transmit whenever they want to; if two or more
packets collide, each computer just waits a random time and tries again later.

Two broadcast networks. Fig 1.13(a) Bus. (b) Ring.

A second type of broadcast system is the ring. In a ring, each bit propagates
around on its own, not waiting for the rest of the packet to which it belongs. Typically,
each bit circumnavigates the entire ring in the time it takes to 21 transmit a few bits,
often before the complete packet has even been transmitted. As with all other broadcast
systems, some rule is needed for arbitrating simultaneous accesses to the ring. Various
methods, such as having the machines take turns, are in use. IEEE 802.5 (the IBM
token ring), is a ring-based LAN operating at 4 and 16 Mbps. FDDI is another example
of a ring network.
Broadcast networks can be further divided into static and dynamic, depending
on how the channel is allocated. A typical static allocation would be to divide time into
discrete intervals and use a round-robin algorithm, allowing each machine to broadcast
only when its time slot comes up. Static allocation wastes channel capacity when a
machine has nothing to say during its allocated slot, so most systems attempt to
allocate the channel dynamically (i.e., on demand). Dynamic allocation methods for a
common channel are either centralized or decentralized. In the centralized channel
allocation method, there is a single entity, for example, a bus arbitration unit, which
determines who goes next. It might do this by accepting requests and making a
decision according to some internal algorithm. In the decentralized channel allocation
method, there is no central entity; each machine must decide for itself whether to
transmit. You might think that this always leads to chaos, but it does not. Later we will
study many algorithms designed to bring order out of the potential chaos.

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 Fig 1.14
Metropolitan Area Networks
A metropolitan area network, or MAN, covers a city. The best-known example
of a MAN is the cable television network available in many cities. This system grew
from earlier community antenna systems used in areas with poor over-the-air television
reception. In these early systems, a large antenna was placed on top of a nearby hill and
signal was then piped to the subscribers' houses. At first, these were locally-designed,
ad hoc systems. Then companies began jumping into the business, getting contracts
from city governments to wire up an entire city. The next step was television
programming and even entire channels designed for cable only. Often these channels
were highly specialized, such as all news, all sports, all cooking, all gardening, and so
on. But from their inception until the late 1990s, they were intended for television
reception only.
Starting when the Internet attracted a mass audience, the cable TV network
operators began to realize that with some changes to the system, they could provide
two-way Internet service in unused parts of the spectrum. At that point, the cable TV
system began to morph from a way to distribute television to a metropolitan area
network. To a first approximation, a MAN might look something like the system
shown in Fig 1.14. In this figure we see both television signals and Internet being fed
into the centralized head end for subsequent distribution to people's homes.

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 Figure 1.15. A metropolitan area network based on cable TV

Campus area network

A campus area network (CAN) is a computer network made up of an

interconnection of local area networks (LANs) within a limited geographical area. It
can be considered one form of a metropolitan area network, specific to an academic
setting.

In the case of a university campus-based campus area network, the network is

likely to link a variety of campus buildings including; academic departments, the
university library and student residence halls. A campus area network is larger than a
local area network but smaller than a wide area network (WAN) (in some cases).

The main aim of a campus area network is to facilitate students accessing

internet and university resources. This is a network that connects two or more LANs
but that is limited to a specific and contiguous geographical area such as a college
campus, industrial complex, office building, or a military base. A CAN may be
considered a type of MAN (metropolitan area network), but is generally limited to a
smaller area than a typical MAN. This term is most often used to discuss the
implementation of networks for a contiguous area. This should not be confused with a
Controller Area Network. A LAN connects network devices over a relatively short
distance. A networked office building, school, or home usually contains a single LAN,
though sometimes one building will contain a few small LANs (perhaps one per room),
and occasionally a LAN will span a group of nearby buildings.

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 Metropolitan area network

A metropolitan area network (MAN) is a network that connects two or more

local area networks or campus area networks together but does not extend beyond the
boundaries of the immediate town/city. Routers, switches and hubs are connected to
create a metropolitan area network.

Wide area network

A wide area network (WAN) is a computer network that covers a broad area
(i.e. any network whose communications links cross metropolitan, regional, or national
boundaries [1]). Less formally, a WAN is a network that uses routers and public
communications links. Contrast with personal area networks (PANs), local area
networks (LANs), campus area networks (CANs), or metropolitan area networks
(MANs), which are usually limited to a room, building, campus or specific
metropolitan area (e.g., a city) respectively. The largest and most well-known example
of a WAN is the Internet. A WAN is a data communications network that covers a
relatively broad geographic area (i.e. one city to another and one country to another
country) and that often uses transmission facilities provided by common carriers, such
as telephone companies. WAN technologies generally function at the lower three
layers of the OSI reference model: the physical layer, the data link layer, and the
network layer.

Fig 1.16

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 Global area network

A global area networks (GAN) (see also IEEE 802.20) specification is in

development by several groups, and there is no common definition. In general,
however, a GAN is a model for supporting mobile communications across an arbitrary
number of wireless LANs, satellite coverage areas, etc. The key challenge in mobile
communications is "handing off" the user communications from one local coverage
area to the next. In IEEE Project 802, this involves a succession of terrestrial
WIRELESS local area networks (WLAN).

Virtual private network

A virtual private network (VPN) is a computer network in which some of the

links between nodes are carried by open connections or virtual circuits in some larger
network (e.g., the Internet) instead of by physical wires. The data link layer protocols
of the virtual network are said to be tunneled through the larger network when this is
the case. One common application is secure communications through the public
Internet, but a VPN need not have explicit security features, such as authentication or
content encryption. VPNs, for example, can be used to separate the traffic of different
user communities over an underlying network with strong security features.

A VPN may have best-effort performance, or may have a defined service level
agreement (SLA) between the VPN customer and the VPN service provider. Generally,
a VPN has a topology more complex than point-to-point.

A VPN allows computer users to appear to be editing from an IP address

location other than the one which connects the actual computer to the Internet.

Internet work

An Internetwork is the connection of two or more distinct computer networks

or network segments via a common routing technology. The result is called an
internetwork (often shortened to internet). Two or more networks or network segments
connect using devices that operate at layer 3 (the 'network' layer) of the OSI Basic
Reference Model, such as a router. Any interconnection among or between public,
private, commercial, industrial, or governmental networks may also be defined as an
internetwork.

In modern practice, interconnected networks use the Internet Protocol. There

are at least three variants of internetworks, depending on who administers and who
participates in them:

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  Intranet
 Extranet
 Internet

Intranets and extranets may or may not have connections to the Internet. If
connected to the Internet, the intranet or extranet is normally protected from being
accessed from the Internet without proper authorization. The Internet is not considered
to be a part of the intranet or extranet, although it may serve as a portal for access to
portions of an extranet.

Intranet

An intranet is a set of networks, using the Internet Protocol and IP-based tools
such as web browsers and file transfer applications, that is under the control of a single
administrative entity. That administrative entity closes the intranet to all but specific,
authorized users. Most commonly, an intranet is the internal network of an
organization. A large intranet will typically have at least one web server to provide
users with organizational information.

Extranet

An extranet is a network or internetwork that is limited in scope to a single

organization or entity and also has limited connections to the networks of one or more
other usually, but not necessarily, trusted organizations or entities (e.g., a company's
customers may be given access to some part of its intranet creating in this way an
extranet, while at the same time the customers may not be considered 'trusted' from a
security standpoint). Technically, an extranet may also be categorized as a CAN,
MAN, WAN, or other type of network, although, by definition, an extranet cannot
consist of a single LAN; it must have at least one connection with an external network.

Internet
The Internet consists of a worldwide interconnection of governmental, academic,
public, and private networks based upon the networking technologies of the Internet
Protocol Suite. It is the successor of the Advanced Research Projects Agency Network
(ARPANET) developed by DARPA of the U.S. Department of Defense. The Internet is
also the communications backbone underlying the World Wide Web (WWW). The
'Internet' is most commonly spelled with a capital 'I' as a proper noun, for historical
reasons and to distinguish it from other generic internetworks.

Participants in the Internet use a diverse array of methods of several hundred

documented, and often standardized, protocols compatible with the Internet Protocol
Suite and an addressing system (IP Addresses) administered by the Internet Assigned
Numbers Authority and address registries. Service providers and large enterprises
exchange information about the reachability of their address spaces through the Border
Gateway Protocol (BGP), forming a redundant worldwide mesh of transmission paths.

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 OSI Model
The OSI model is a reference model which most IT professionals use to describe
networks and network applications.

The OSI model was originally intended to describe a complete set of production
network protocols, but the cost and complexity of the government processes involved
in defining the OSI network made the project unviable. In the time that the OSI
designers spent arguing over who would be responsible for what, TCP/IP conquered
the world.

The Seven Layers of the OSI Model

The seven layers of the OSI model are:
Layer Name
7 Application
6 Presentation
5 Session
4 Transport
3 Network
2 Data Link
1 Physical

Layers of OSI Model

Layer Seven of the OSI Model

The Application Layer of the OSI model is responsible for providing end-user
services, such as file transfers, electronic messaging, e-mail, virtual terminal access,
and network management. This is the layer with which the user interacts.

Layer Six of the OSI Model

The Presentation Layer of the OSI model is responsible for defining the syntax
which two network hosts use to communicate. Encryption and compression should be
Presentation Layer functions.

Layer Five of the OSI Model

The Session Layer of the OSI model is responsible for establishing process-to-
process commnunications between networked hosts.

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 Layer Four of the OSI Model

The Transport Layer of the OSI model is responsible for delivering messages
between networked hosts. The Transport Layer should be responsible for fragmentation
and reassembly.

Layer Three of the OSI Model

The Network Layer of the OSI model is responsible for establishing paths for
data transfer through the network. Routers operate at the Network Layer.

Layer Two of the OSI Model

The Data Link Layer of the OSI model is responsible for communications
between adjacent network nodes. Hubs and switches operate at the Data Link Layer.

Layer One of the OSI Model

The Physical Layer of the OSI model is responsible for bit-level transmission
between network nodes. The Physical Layer defines items such as: connector types,
cable types, voltages, and pin-outs.

The OSI model is a layered framework for the design of network systems that
allows

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 communication between all types of computer systems. It Consists of seven
separate but related layers, each of which defines a part of the process of moving
information across a network (see Figure 1.17). An understanding of the fundamentals
of the OSI model provides a solid basis for exploring data communications.

Layers of OSI Model

Layers of OSI Model

Fig 1.17

Layered Architecture

The OSI model is composed of seven ordered layers: physical (layer 1), data
link (layer 2), network (layer 3), transport (layer 4), session (layer 5), presentation
(layer 6), and application (layer 7). Figure 2.3 shows the layers involved when a
message is sent from device A to device B. As the message traveLs from A to B, it
may pass through many intermediate nodes. These intermediate nodes usually involve
only the first three layers of the OSI model.

In developing the model, the designers distilled the process of transmiting data
to its most fundamental elements. They identi6ed which networking ftznctons had
related uses and collected those functions into discrete groups that became the layers.
Each layer defines a family of functions distinct from those of the other layers. By
defining and localizing functionality in this fashion, the designers created an
architecture that is both comprehensive and flexible. Most importantly, the OS1 model
allows complete i rileroperabilily between otherwise incompatible systems.

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 Within a single machine, each layer calls upon the services of the layer just
below it. Layer 3, for example, uses the services provided by layer 2 and provides
services for layer 4 Between machines, layer x on one machine communicates with
layer x on another machine. This communication is governed by an agreed-upon series
of rules and conventions called protocols. The processes on each machine that
communicate at a given layer are called peer-to-peer processes. Communication
between machines is therefore a peer-to-peer process using the protocols appropriate to
a given layer.

Physical layer
The physical layer is the first or the lowest layer in the OSI reference model.
This layer deals with the actual transmission of data using a transmission medium.
Functions:
 The physical layer is responsible for:
 Interfacing with the physical transmission medium.
 Defining the physical, electrical, and mechanical properties of he involved
components.
Services:
 The services offered by the physical layer are:
 Setting up of connection
 Ending the connection
 Transmitting data over a communication channel
 Receiving data from a communication channel

Data link layer:

 The data link layer is the second layer in the OSI reference model.
 Provides for the transfer of frames (block of information) across a transmission
link that directly connects two nodes.
 It uses error detection and correction techniques, to ensure that transmission
contains no errors.
 It uses flow control techniques.
Function:
 Interfacing with the physical layer and network layer.
 It received data from network and passes it to physical layer. Data packets are
framed by adding header and trailer.
Services:
 Correcting errors
 Controlling flow of data
 Framing

Network layer:
It is a heart of the OSI model. It deals with routing strategies, which are
responsible for delivery of a packet from source to destination.

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 Function:
 It is used to provide internetworking.
 It moves packets from source to destination.

Services:
 Routing
 Accounting
 Packetizing

Transport Layer:
It provide different types of data transmission services.
Function:
 Interfacing with network and session
 Splitting data
 Controlling transmission and sequencing.
 It ensures the packets are delivered correctly.
Services:
 Controlling errors
 Controlling flow
 Both services connection oriented and connection less
Eg: TCP, UDP, SPX.
Session layer:
This layer manages sessions between communicating entities.
Function:
 Interfacing between transport and presentation.
 It accepts data from presentation and passes to transport.
 It manages the information exchange between two communicating systems with
the help of various services.
Services:
 Authenticating the user (verifying the user information)
 Managing the dialogs
 Controls the transmission of data and determines the applications whose turn it
is to transmit the data.
 Providing synchronization service.
Eg: NetBIOS , RPC( Remote Procedure Call)

Presentation Layer:
It is responsible for data formatting and presenting data for display.
Function:
 Interfacing between the session and application layer.
 Accepts data from application and passes to session.
 Presenting data for display after formatting.
 Converts data from one format to another using Unicode or ASCII .

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 Services:
 Encrypting data
 Decrypting data
 Compressing data
Eg: HTTP, FTP

Application layer:
It is the top most layer.
Function:
 Interfacing between user and presentation
 Accepts input from user and passes to presentation.
 Defining how applications on one computer can communications with
application on other computers.
Services:
It provides underlying network related services.
E.g: FTP, Telnet.

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 UNIT – II
Syllabus:
Parallel and Serial Transmission – DTE/DCE/such as EIA-449, EIA-530 and x.21
interface – Interface standards – Modems –Guided Media – Unguided Media –
Performance – Types of Error – Error Detection – Error Corrections.

Parallel and Serial Transmission

Digital data transmission can occur in two basic modes: serial or parallel. Data
within a computer system is transmitted via parallel mode on buses with the width of
the parallel bus matched to the word size of the computer system. Data between
computer systems is usually transmitted in bit serial mode. Consequently, it is
necessary to make a parallel-to-serial conversion at a computer interface when sending
data from a computer system into a network and a serial-to-parallel conversion at a
computer interface when receiving information from a network. The type of
transmission mode used may also depend upon distance and required data rate.

Parallel Transmission

In parallel transmission, multiple bits (usually 8 bits or a byte/character) are

sent simultaneously on different channels (wires, frequency channels) within the same
cable, or radio path, and synchronized to a clock. Parallel devices have a wider data
bus than serial devices and can therefore transfer data in words of one or more bytes at
a time. As a result, there is a speedup in parallel transmission bit rate over serial
transmission bit rate. However, this speedup is a tradeoff versus cost since multiple
wires cost more than a single wire, and as a parallel cable gets longer, the
synchronization timing between multiple channels becomes more sensitive to distance.
The timing for parallel transmission is provided by a constant clocking signal sent over
a separate wire within the parallel cable; thus parallel transmission is considered
synchronous.

Examples

Examples of parallel mode transmission include connections between a

computer and a printer (parallel printer port and cable). Most printers are within 6
meters or 20 feet of the transmitting computer and the slight cost for extra wires is
offset by the added speed gained through parallel transmission of data.

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 Serial Transmission

In serial transmission, bits are sent sequentially on the same channel (wire)
which reduces costs for wire but also slows the speed of transmission. Also, for serial
transmission, some overhead time is needed since bits must be assembled and sent as a
unit and then disassembled at the receiver.

Serial transmission can be either synchronous or asynchronous. In

synchronous transmission, groups of bits are combined into frames and frames are sent
continuously with or without data to be transmitted. In asynchronous transmission,
groups of bits are sent as independent units with start/stop flags and no data link
synchronization, to allow for arbitrary size gaps between frames. However, start/stop
bits maintain physical bit level synchronization once detected.

Applications

Serial transmission is between two computers or from a computer to an external

device located some distance away. Parallel transmission either takes place within a
computer system (on a computer bus) or to an external device located a close distance
away.

A special computer chip known as a universal asynchronous receiver

transmitter (UART) acts as the interface between the parallel transmission of the
computer bus and the serial transmission of the serial port. UARTs differ in
performance capabilities based on the amount of on-chip memory they possess.

Examples

Examples of serial mode transmission include connections between a computer

and a modem using the RS-232 protocol. Although an RS-232 cable can theoretically
accommodate 25 wires, all but two of these wires are for overhead control signaling
and not data transmission; the two data wires perform simple serial transmission in
either direction. In this case, a computer may not be close to a modem, making the cost
of parallel transmission prohibitive—thus speed of transmission may be considered
less important than the economical advantage of serial transmission.

Tradeoffs

Serial transmission via RS-232 is officially limited to 20 Kbps for a distance of

15 meters or 50 feet. Depending on the type of media used and the amount of external

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 interference present, RS-232 can be transmitted at higher speeds, or over greater
distances, or both. Parallel transmission has similar distance-versus-speed tradeoffs, as
well as a clocking threshold distance. Techniques to increase the performance of serial
and parallel transmission (longer distance for same speed or higher speed for same
distance) include using better transmission media, such as fiber optics or conditioned
cables, implementing repeaters, or using shielded/multiple wires for noise immunity.

DTE/DCE/such as EIA-449, EIA-530, EIA-232 and x.21 interface

Data terminal equipment (DTE) is an end instrument that converts user

information into signals or reconverts received signals. These can also be called tail
circuits. A DTE device communicates with the data circuit-terminating equipment
(DCE). The DTE/DCE classification was introduced by IBM.

Two different types of devices are assumed on each end of the interconnecting
cable for a case of simply adding DTE to the topology (e.g. to a hub, DCE), which also
brings a less trivial case of interconnection of devices of the same type: DTE-DTE or
DCE-DCE. Such cases need crossover cables, such as for the Ethernet or null modem
for RS-232.

A DTE is the functional unit of a data station that serves as a data source or a
data sink and provides for the data communication control function to be performed in
accordance with link protocol.

The data terminal equipment may be a single piece of equipment or an

interconnected subsystem of multiple pieces of equipment that perform all the required
functions necessary to permit users to communicate. A user interacts with the DTE
(e.g. through a human-machine interface), or the DTE may be the user.

Usually, the DTE device is the terminal (or a computer emulating a terminal),
and the DCE is a modem.

DTE is usually a male connector and DCE is a female connector.

A general rule is that DCE devices provide the clock signal (internal clocking)
and the DTE device synchronizes on the provided clock (external clocking). D-sub
connectors follow another rule for pin assignment.

 25 pin DTE devices transmit on pin 2 and receive on pin 3.

 25 pin DCE devices transmit on pin 3 and receive on pin 2.
 9 pin DTE devices transmit on pin 3 and receive on pin 2.
 9 pin DCE devices transmit on pin 2 and receive on pin 3.

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 This term is also generally used in the Telco and CISCO equipment context to
designate a device unable to generate clock signals, hence a PC to PC Ethernet
connection can also be called a DTE to DTE communication. This communication is
done via an Ethernet crossover cable as opposed to a PC to DCE (hub, switch, or
bridge) communication which is done via an Ethernet straight cable.

Data circuit-terminating equipment

A Data circuit-terminating equipment (DCE) is a device that sits between

the data terminal equipment (DTE) and a data transmission circuit. It is also called
data communications equipment and data carrier equipment.

In a data station, the DCE performs functions such as signal conversion,

coding, and line clocking and may be a part of the DTE or intermediate equipment.
Interfacing equipment may be required to couple the data terminal equipment (DTE)
into a transmission circuit or channel and from a transmission circuit or channel into
the DTE.

Although the terms are most commonly used with RS-232, several data
communications standards define different types of interfaces between a DCE and a
DTE. The DCE is a device that communicates with a DTE device in these standards.
Standards that use this nomenclature include:

 Federal Standard 1037C, MIL-STD-188

 RS-232
 Certain ITU-T standards in the V series (notably V.24 and V.35)
 Certain ITU-T standards in the X series (notably X.21 and X.25)

A general rule is that DCE devices provide the clock signal (internal clocking)
and the DTE device synchronizes on the provided clock (external clocking). D-sub
connectors follow another rule for pin assignment. DTE devices usually transmit on
pin connector number 2 and receive on pin connector number 3. DCE devices are just
the opposite: pin connector number 2 receives and pin connector number 3 transmits
the signals.

Usually, the DTE device is the terminal (or computer), and the DCE is a
modem.

When two devices, that are both DTE or both DCE, must be connected together
without a modem or a similar media translator between them, a kind of crossover cable
must be used, i.e. a null modem for RS-232 or as usual for Ethernet.

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 RS- 449

The RS-449 specification, also known as EIA-449 or TIA-449, defines the

functional and mechanical characteristics of the interface between data terminal
equipment and data communications equipment.

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 EIA-530

EIA-530, or RS-530, is a balanced serial interface standard that generally uses

a 25-pin connector.

The specification defines the cable between the DTE and DCE devices. It is to
be used in conjunction with EIA-422 and EIA-423, which define the electrical
signalling characteristics. Because EIA-530 calls for the more common 25 pin
connector, it displaced the similar EIA-449, which also uses EIA-422/423, but a larger
37-pin connector.

Two types of EIA-530 are defined: the Category 1, which uses the balanced
characteristics of EIA-422, and Category 2, which is the unbalanced EIA-423.

EIA-530, or RS-530, is a balanced serial interface standard that generally uses

a 25 pin connector. The RS530 isn't an actual interfaces, but a generic connector
specification. The connector pinning can be used to support RS422, RS423,
V.36/V.37/V.10/V.11 (not V.35!) and X.21 to name the most popular ones.

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 RS530 is just like RS422 and uses a differential signaling on a DB25 - RS232
format - EIA-530 Transmit (and the other signals) use a twisted pair of wires (TD+ &
TD-) instead of TD and a ground reference as in RS232 or V.24. This interface is used
for HIGH SPEED synchronous protocols. Using a differential signaling allows for
higher speeds over long cabling. This standard is applicable for use at data signaling
rates in the range from 20,000 to a nominal upper limit of 2,000,000 bits per second.
Equipment complying with this standard, however, need not operate over this entire
data signaling rate range. They may be designed to operate over a narrower range as
appropriate for the specific application.

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 X.21

X.21, sometimes referred to as X21, interface is a specification for differential

communications introduced in the mid 1970‘s by the ITU-T. X.21 was first introduced
as a means to provide a digital signaling interface for telecommunications between
carriers and customer‘s equipment. This includes specifications for DTE/DCE physical
interface elements, alignment of call control characters and error checking, elements of
the call control phase for circuit switching services, and test loops.

When X.21 is used with V.11, it provides

synchronous data transmission at rates from 100 kbit/s to 10 Mbit/s. There is also a
variant of X.21 which is only used in select legacy applications, ―circuit switched
X.21‖. X.21 normally is found on a 15-pin D Sub connector and is capable of running
full-duplex data transmissions.

The Signal Element Timing, or clock, is provided by the carrier (your telephone
company), and is responsible for correct clocking of the data. X.21 is primarily used in
Europe and Japan, for example in the Scandinavian DATEX and German DATEX-L
circuit switched networks during the 1980s.

X.21 Overview

X.21 is a state-driven protocol running full duplex at 9600 bps to 64 Kbps with
subscriber networks. It is a circuit-switching protocol using Synchronous ASCII with
odd parity to connect and disconnect a subscriber to the public-switching network.

The data-transfer phase is transparent to the network. Any data can be

transferred through the network after Call Establishment is made successfully via the
X.21 protocol. The call-control phases which are used were defined in the CCITT (now
ITU) 1988 "Blue Book" Recommendations X.1 - X.32.

Signals Provided

The signals of the X.21 interface are presented on a 15-pin connector defined
by ISO Document 4903. The electrical characteristics are defined in CCITT
Recommendations X.26 and X.27, which refer to CCITT Recommendations V.10 and
V.11.

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 X.21 provides eight signals:

Signal Ground (G) –

This provides reference for the logic states against the other circuits. This signal
may be connected to the protective ground (earth).

DTE Common Return (Ga) -

It is used only in unbalanced-type configurations (X.26), this signal provides

reference ground for receivers in the DCE interface.

Transmit (T) -

This carries the binary signals which carry data from the DTE to the DCE. This
circuit can be used in data-transfer phases or in call-control phases from the DTE to
DCE (during Call Connect or Call Disconnect).

Receive (R) -

This carries the binary signals from DCE to DTE. It is used during the data-
transfer or Call Connect/Call Disconnect phases.

Control (C) -

It is controlled by the DTE to indicate to the DCE the meaning of the data sent
on the transmit circuit. This circuit must be ON during data-transfer phase and can be
ON or OFF during call-control phases, as defined by the protocol.

Indication (I) -

The DCE controls this circuit to indicate to the DTE the type of data sent on the
Receive line. During data phase, this circuit must be ON and it can be ON or OFF
during call control, as defined by the protocol.

Signal Element Timing (S) -

This provides the DTE or dCE with timing information for sampling the
Receive line or Transmit line. The DTE samples at the correct instant to determine if a
binary 1 or 0 is being sent by the DCE. The DCE samples to accurately recover signals
at the correct instant. This signal is always ON.

Byte Timing (B) -

This circuit is normally ON and provides the DTE with 8-bit byte element
timing. The circuit transitions to OFF when the Signal Element Timing circuit samples

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 the last bit of an 8-bit byte. Call-control characters must align with the B lead during
call-control phases. During data- transfer phase, the communicating devices bilaterally
agree to use the B lead to define the end of each transmitted or received byte. The C
and I leads then only monitor and record changes in this condition when the B lead
changes from OFF to ON, although the C and I leads may be altered by the transitions
on the S lead. This lead is frequently not used.

X.21 Protocol Operation

As stated previously, X.21 is a state protocol. Both the DTE and DCE can be in
a Ready or Not-Ready state. The Ready state for the DTE is indicated by a continuous
transmission of binary 1's on the T lead. The Ready state for the DCE is continuous
transmission of binary 1's on the R lead. During this continuous transmission of Ready
state, the control leads are OFF.

During the Not-Ready state, the DCE transmits binary 0's on the R lead with
the I lead in the OFF state.

The DTE Uncontrolled Not-Ready is indicated by transmission of binary 0's

with the C lead in the OFF state. The DTE Uncontrolled Not-Ready state signifies that
the DTE is unable to accept calls due to an abnormal condition. The DTE Controlled
Not-Ready state sends a pattern of alternating 1's and 0's on the T lead with the C lead
OFF. This state indicates that the DTE is operational, but unable to accept incoming
calls.

Sub-D15 Male Sub-D15 Female

Pin Signal abbr. DTE DCE

1 Shield - -
2 Transmit (A) Out In
3 Control (A) Out In
4 Receive (A) In Out
5 Indication (A) In Out
6 Signal Timing (A) In Out

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 7 Unassigned
8 Ground - -
9 Transmit (B) Out In
10 Control (B) Out In
11 Receive (B) In Out
12 Indication (B) In Out
13 Signal Timing (B) In Out
14 Unassigned
15 Unassigned

Functional Description

As can be seen from the pinning specifications, the Signal Element Timing
(clock) is provided by the DCE. This means that your provider (local telco office) is
responsible for the correct clocking and that X.21 is a synchronous interface. Hardware
handshaking is done by the Control and Indication lines. The Control is used by the
DTE and the Indication is the DCE one.

Modems

To get the most out of a modem, you should have a communications software
package, a program that simplifies the task of transferring data.

Short for modulator-demodulator. A modem is a device or program that

enables a computer to transmit data over, for example, telephone or cable lines.
Computer information is stored digitally, whereas information transmitted over
telephone lines is transmitted in the form of analog waves. A modem converts between
these two forms.

Fortunately, there is one standard interface for connecting external modems to

computers called RS-232. Consequently, any external modem can be attached to any
computer that has an RS-232 port, which almost all personal computers have. There are

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 also modems that come as an expansion board that you can insert into a vacant
expansion slot. These are sometimes called onboard or internal modems.

While the modem interfaces are standardized, a number of different protocols for
formatting data to be transmitted over telephone lines exist. Some, like CCITT V.34,
are official standards, while others have been developed by private companies. Most
modems have built-in support for the more common protocols -- at slow data
transmission speeds at least, most modems can communicate with each other. At high
transmission speeds, however, the protocols are less standardized.

Aside from the transmission protocols that they support, the following
characteristics distinguish one modem from another:

bps : How fast the modem can transmit and receive data. At slow rates, modems are
measured in terms of baud rates. The slowest rate is 300 baud (about 25 cps). At higher
speeds, modems are measured in terms of bits per second (bps). The fastest modems
run at 57,600 bps, although they can achieve even higher data transfer rates by
compressing the data. Obviously, the faster the transmission rate, the faster you can
send and receive data. Note, however, that you cannot receive data any faster than it is
being sent. If, for example, the device sending data to your computer is sending it at
2,400 bps, you must receive it at 2,400 bps. It does not always pay, therefore, to have a
very fast modem. In addition, some telephone lines are unable to transmit data reliably
at very high rates.

voice/data: Many modems support a switch to change between voice and data
modes. In data mode, the modem acts like a regular modem. In voice mode, the modem
acts like a regular telephone. Modems that support a voice/data switch have a built-in
loudspeaker and microphone for voice communication.

auto-answer : An auto-answer modem enables your computer to receive calls in

your absence. This is only necessary if you are offering some type of computer service
that people can call in to use.

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 data compression : Some modems perform data compression, which enables them
to send data at faster rates. However, the modem at the receiving end must be able to
decompress the data using the same compression technique.

flash memory : Some modems come with flash memory rather than conventional
ROM, which means that the communications protocols can be easily updated if
necessary.

Fax capability: Most modern modems are fax modems, which means that they can
send and receive faxes.

How Modem works?

Modem is a device that allows us to connect two computers using normal

telephone line. The sending computer‘s modem modulates the digital signal from
computer into analog signal required for transmission over the telephone line.

At the receiving end, the incoming analog signal is given to the modem. This
modem demodulates the analog signal back into digital signal and gives this signal to
the computer.

When a computer is switched on and is ready to transmit data it sends a Data

Terminal Ready (DTR) signal to the modem. When the modem is ready to receive data
or instruction from the computer, it sends a Data Set Ready (DSR) signal to the
computer. Both these signals must be present before the next step can happen.

Next, the computer sends a command to the modem to go‖ Off Hook‖, i.e to
open communication with the phone line. Next, the computer issues a command to
modem to dial a phone number.

These commands are given via ―Transmit Data TxD‖ line. The modem
acknowledges receipt of these commands by replaying to the computer on the ―Receive
Data RxD‖ line.

When the remote modem, the modem at the other end of line responds, the
local modem sends a special greeting tone to the remote modem, to inform the remote
modem that it is being called by another modem. The remote modem responds with a
higher pitched tone.

Once a communication is established between the two modems, local modem

sends a ―Carrier Detect‖ CD signal to the computer. Carrier is a steady tone signal of a
fixed frequency, which is later modulated by modem to transmit the data.

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 Next, the two modems enter a process known as ―Handshake‖. This process
decides how the two modems are going to communicate with each other. Various
things decided during this process are

 The number of bits used to represent data,

 Number of bits used to indicate start and end of data,
 Use of parity bit for error checking / correction
 Transmission speed, transmission protocol, compression method etc. being used
 If the two modems, local and remote do not use the same settings then they will
wind up sending characters that will not make sense, or they may completely refuse
to communicate with each other.

Once the handshake is done, the computer sends a Request To Send (RTS) signal
to the local modem. This signal asks the modem if the modem is free to receive data
from the computer.

If the modem is not busy, then the modem will send a Clear To Send (CTS) signal
to the computer. Once the computer receives this CIS signal from the modem, the
computer will send the data to be transmitted to the modem.

If the modem cannot transmit the data, as fast as it received from the computer,
then the modem will drop the CTS signal to the computer, this will stop any further
data from the computer. Computer will stop data transmission to the modem until it
receives CTS signal.

Data received by the local modem is transmitted as different frequency sound

signal to the remote modem, the remote modem converts these series of tones back into
digital signal and gives it to the computer, it is connected to.

Once the data transmission is over, modem sends a break communication

command. If the connection is broken by remote system, the local modem will drop the
Carrier Detect (CD) signal.

This will inform the local computer that the communication is broken.

The meaning of the indicator lights given on a modem‘s front panel.

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 TRANSMISSION MEDIA

A transmission medium can be broadly defined as anything that can carry

information from a source to a destination. In data communications the definition of
the information and the transmission medium is more specific. 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.
The use of optical fibers has increased the data rate incredibly. Free space (air,
vacuum, and water) is used more efficiently.
Computers and other telecommunication devices use signals to represent data.
These signals are transmitted from one device to another in the form of electromagnetic
energy, which is propagated through transmission media.

Electromagnetic energy, a combination of electric and magnetic fields vibrating

in relation to each other, includes power, radio waves, infrared light, visible light,
ultraviolet light, and X, gamma, and cosmic rays Each of these constitutes a portion of
the electromagnetic spectrum. Not all portions of the spectrum are currently usable for
telecommunications, however. The media to harness those that are usable are also
limited to a few types.

In telecommunications, transmission media can be divided into two broad

categories: guided and unguided. Guided media include twisted-pair cable, coaxial
cable, and fiber-optic cable. Unguided medium is free space. Figure (9) shows this
taxonomy.

Fig (9) Types of transmission media

The selection of transmission media depends upon following factors:

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 i. Bandwidth:
The maximum data rates depend upon the bandwidth of the transmission medium.

ii. Connectivity:
This factor determines, how the transmission media can be connected in
networking. For example, some media are best suited for broadcast type of
applications.

iii. Geographic coverage:

This parameter determines the geographic area covered by the transmission media.

iv. Noise immunity:

The effect of noise on the transmission medium affects the signal quality. The
transmission media should have high noise immunity.

v. Security:
This refers to data security on the transmission media.

GUIDED MEDIA

Guided media, which are those that provide a conduct from one device to
another, include twisted-pair cable, coaxial cable, and fiber-optic cable. A signal
traveling along any of these media is directed and contained by the physical limits of
the medium. Twisted-pair and coaxial cable use metallic (copper) conductors that
accept and transport signals in the form of electric current. Optical fiber is a cable that
accepts and transports signals in the form of light.

Twisted - Pair Cable

A twisted pair consists of two conductors (normally copper), each with its own
plastic insulation, twisted together. One of the wires is used to carry signals to the
receiver, and the other is used only as a ground reference. The receiver uses the
difference between the two. In addition to the signal sent by the sender on one of the
wires, interference (noise) and crosstalk may affect both wires and create unwanted
signals.
If the two wires are parallel, the effect of these unwanted signals is not the same
in both wires because they are at different locations relative to the noise or crosstalk

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 sources (e.g., one is closer and the other is farther). This results in a difference at the
receiver. By twisting the pairs, a balance is maintained.

Unshielded Versus Shielded Twisted-Pair Cable

The most common twisted-pair cable used in communications is referred to as
unshielded twisted-pair (UTP). IBM has also produced a version of twisted-pair cable
for its use called shielded twisted-pair (STP). STP cable has a metal foil or braided-
mesh covering that encases each pair of insulated conductors. Although metal casing
improves the quality of cable by preventing the penetration of noise or crosstalk, it is
bulkier and more expensive.

Applications of Twisted - Pair Cable:

i. Twisted-pair cables are used in telephone lines to provide voice and data channels.
The local loop—the line that connects subscribers to the central telephone office—
commonly consists of unshielded twisted pair cables.
ii. The DSL (Digital Subscriber Line) lines that are used by the telephone companies
to provide high-data-rate connections also use the high bandwidth capability of
unshielded twisted-pair cables.
iii. Local area networks, such as 10Base-T and 100Base-T, also use twisted-pair
cables.

Performance

One way to measure the performance of twisted-pair cable is to compare

attenuation versus frequency and distance. A twisted-pair cable can pass a wide range
of frequencies. However, Figure 7.6 shows that with increasing frequency, the
attenuation, measured in decibels per kilometer (dB/km), sharply increases with
frequencies above 100 kHz. Note that gauge is a measure of the thickness of the wire.

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 Fig Performance of UTP

Coaxial Cable

Coaxial cable (or coax) carries signals of higher frequency ranges than those in
twisted- pair cable, in part because the two media are constructed quite differently.
Instead of having two wires, coax has a central core conductor of solid or stranded wire
(usually copper) enclosed in an insulating sheath, which is, in turn, encased in an outer
conductor of metal foil, braid, or a combination of the two. The outer metallic
wrapping serves both as a shield against noise and as the second conductor, which
completes the circuit. This outer conductor is also enclosed in an insulating sheath, and
the whole cable is protected by a plastic cover.

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 Performance

In Figure 7.9 the attenuation is much higher in coaxial cables than in twisted-
pair cable. In other words, although coaxial cable has a much higher bandwidth, the
signal weakens rapidly and requires the frequent use of repeaters.

Fig Performance of Coaxial Cable

Applications of Coaxial Cable:

i. Coaxial cable was widely used in analog telephone networks where a single coaxial
network could carry 10,000 voice signals.

ii. Later it was used in digital telephone networks where a single coaxial cable could
carry digital data up to 600 Mbps. However, coaxial cable in telephone networks
has largely been replaced today with fiber-optic cable.

iii. Cable TV networks also use coaxial cables. In the traditional cable TV network, the
entire network used coaxial cable. Later, however, cable TV providers replaced
most of the media with fiber-optic cable; hybrid networks use coaxial cable only at
the network boundaries, near the consumer premises. Cable TV uses RG-59 coaxial
cable.

iv. Another common application of coaxial cable is in traditional Ethernet LANs.

Because of its high bandwidth, and consequently high data rate, coaxial cable was
chosen for digital transmission in early Ethernet LANs.

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 Fiber-Optic Cable

A fiber-optic cable is made of glass or plastic and transmits signals in the form
of light. Optical fibers use reflection to guide light through a channel. A glass or plastic
core is surrounded by a cladding of less dense glass or plastic. The difference in
density of the two materials must be such that a beam of light moving through the core
is reflected off the cladding instead of being refracted into it.

Performance

The plot of attenuation versus wavelength in Figure 7.16 shows a very

interesting phenomenon in fiber-optic cable. Attenuation is flatter than in the case of
twisted-pair cable and coaxial cable. The performance is such that we need fewer
(actually 10 times less) repeaters when we use fiber-optic cable.

Fig Performance of Fiber optic cable

Applications of Fiber-Optic Cable

i. Fiber-optic cable is often found in backbone networks because its wide bandwidth
is cost-effective. Today, with wavelength-division multiplexing (WDM), we can
transfer data at a rate of 1600 Gbps. The SONET network provides such a
backbone.

ii. Some cable TV companies use a combination of optical fiber and coaxial cable,
thus creating a hybrid network. Optical fiber provides the backbone structure while
coaxial cable provides the connection to the user premises. This is a cost-effective

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 configuration since the narrow bandwidth requirement at the user end does not
justify the use of optical fiber.

iii. Local - area networks such as 1 00B as e -FX network (Fast Ethernet) and 1 000B
as e- X also use fiber-optic cable.

Advantages of Optical Fiber:

Fiber-optic cable has several advantages over metallic cable (twisted pair or
coaxial).

 Higher bandwidth.
Fiber-optic cable can support dramatically higher bandwidths (and hence data
rates) than either twisted-pair or coaxial cable. Currently, data rates and bandwidth
utilization over fiber-optic cable are limited not by the medium but by the signal
generation and reception technology available.

 Less signal attenuation.

Fiber-optic transmission distance is significantly greater than that of other
guided media. A signal can run for 50 km without requiring regeneration. We need
repeaters every 5 km for coaxial or twisted-pair cable.

 Immunity to electromagnetic interference.

Electromagnetic noise cannot affect fiber-optic cables.

 Resistance to corrosive materials.

Glass is more resistant to corrosive materials than copper.

 Light weight.
Fiber-optic cables are much lighter than copper cables. Greater immunity to
tapping. Fibereoptic cables are more immune to tapping than copper cables. Copper
cables create antenna effects that can easily be tapped.

Disadvantages of Optical Fiber:

There are some disadvantages in the use of optical fiber.

 Installation and maintenance.

Fiber-optic cable is a relatively new technology. Its installation and

maintenance require expertise that is not yet available everywhere.

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  Unidirectional light propagation.

Propagation of light is unidirectional. If we need bidirectional communication,

two fibers are needed.

 Cost.

The cable and the interfaces are relatively more expensive than those of other
guided media. If the demand for bandwidth is not high, often the use of optical fiber
cannot be justified.

UNGUIDED MEDIA: WIRELESS

Unguided media transport electromagnetic waves without using a physical

conductor. This type of communication is often referred to as wireless communication.
Signals are normally broadcast through free space and thus are available to anyone who
has a device capable of receiving them.

The part of the electromagnetic spectrum, ranging from 3 kHz to 900 THz are
used for wireless communication.

Unguided signals can travel from the source to destination in several ways:
ground propagation, sky propagation, and line-of-sight propagation.

 In ground propagation, radio waves travel through the lowest portion of the
atmosphere, hugging the earth. These low-frequency signals emanate in all
directions from the transmitting antenna and follow the curvature of the planet.
Distance depends on the amount of power in the signal: The greater the power, the
greater the distance.

 In sky propagation, higher-frequency radio waves radiate upward into the

ionosphere (the layer of atmosphere where particles exist as ions) where they are
reflected back to earth. This type of transmission allows for greater distances with
lower output power.

 In line-of-sight propagation, very high-frequency signals are transmitted in

straight lines directly from antenna to antenna. Antennas must be directional, facing
each other and either tall enough or close enough together not to be affected by the
curvature of the earth. Line-of-sight propagation is tricky because radio
transmissions cannot be completely focused.

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 The section of the electromagnetic spectrum defined as radio waves and
microwaves is divided into eight ranges, called bands, each regulated by government
authorities. These bands are rated from very low frequency (VLF) to extremely high
frequency (EHF).

Table 7.4 lists these bands, their ranges, propagation methods, and some

applications.

We can divide wireless transmission into three broad groups: radio waves,
microwaves, and infrared waves.

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 Fig (10) Wireless transmission waves

Radio Waves

Although there is no clear-cut demarcation between radio waves and

microwaves, electromagnetic waves ranging in frequencies between 3 kHz and 1GHz
are normally called radio waves; waves ranging in frequencies between 1 and 300
GHz are called micro waves. However, the behavior of the waves, rather than the
frequencies, is a better criterion for classification.

Radio waves, for the most part, are omnidirectional. When an antenna transmits
radio waves, they are propagated in all directions. This means that the sending and
receiving antennas do not have to be aligned. A sending antenna sends waves that can
be received by any receiving antenna. The omnidirectional property has a
disadvantage, too. The radio waves transmitted by one antenna are susceptible to
interference by another antenna that may send signals using the same frequency or
band.

Radio waves, particularly those waves that propagate in the sky mode, can
travel long distances. This makes radio waves a good candidate for long distance
broadcasting such as AM radio.

Radio waves, particularly those of low and medium frequencies, can penetrate
walls. This characteristic can be both an advantage and a disadvantage. It is an
advantage because, for example, an AM radio can receive signals inside a building. It
is a disadvantage because we cannot isolate a communication to just inside or outside a

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 building.

Applications of Radio Waves

The omnidirectional characteristics of radio waves make them useful for multicasting,
in which there is one sender but many receivers. AM and FM radio, television,
maritime radio, cordless phones, and paging are examples of multicasting.

Microwaves
Electromagnetic waves having frequencies between 1 and 300 GHz are called
microwaves. Microwaves are unidirectional. When an antenna transmits microwave
waves, they can be narrowly focused. This means that the sending and receiving
antennas need to be aligned. The unidirectional property has an obvious advantage. A
pair of antennas can be aligned without interfering with another pair of aligned
antennas.

Some characteristics of microwave propagation:

 Microwave propagation is line-of-sight. Since the towers with the mounted

antennas need to be in direct sight of each other, towers that are far apart need to be
very tall. The curvature of the earth as well as other blocking obstacles do not allow
two short towers to communicate by using microwaves. Repeaters are often needed
for long- distance communication.
 Very high-frequency microwaves cannot penetrate walls. This characteristic can be
a disadvantage if receivers are inside buildings.
 The microwave band is relatively wide, almost 299 GHz. Therefore wider subbands
can be assigned, and a high data rate is possible
 Use of certain portions of the band requires permission from authorities.

Applications of Microwaves
Microwaves, due to their unidirectional properties, are very useful when unicast
(one-to-one) communication is needed between the sender and the receiver. They are
used in cellular phones, satellite networks, and wireless LANs.

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 Terrestrial Microwave

Terrestrial microwave is used for point to point links. The distance covered by
microwaves is slightly more than line of sight. A parabolic dish antenna of 3m
diameter is used for transmission. This antenna is usually mounted on tall hills or
buildings to cover large distance in line of slight. The distance covered by terrestrial
microwaves is given as,

d = 7.14  eq .1

Here d is the distance covered in km

h is the antenna height in meters
K is the factor to accommodate

more distance covered by bending of microwaves along the curvature of the earth‘s

surface. Value of K is usually taken as . Hence microwaves travel more distance than
line of sight. For terrestrial microwave 2 to 40 GHz frequencies are used. Microwaves
get attenuated with distance. Loss due to attenuation is given as,

Here L is attenuation loss in dB

d is the distance covered in km
and is wavelength of microwaves in km

In the above equation observe that loss varies as a square of the distance. But for
twisted pair cable and co-axial cable and loss varies logarithmically with distance. The
repeaters or amplifiers are places from 10 to 100km apart for terrestrial microwave
links. The communication is done on 4 – 6 GHz bands. Higher bands such as 11, 12
and 22 GHz are also used. Loss increases with frequency. Hence higher frequencies
travel very small distances. But they are highly directional.

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 Applications of Terrestrial Microwave:

 Long haul telecommunication services such as voice and TV transmission.

 Short point to point links between buildings. These are used for CCTV.
 Short haul microwaves are used for bypass applications such as connection to long
distance communication station. In this case local telephone company is bypassed.

Satellite Microwave

Satellites are widely used as repeaters. Fig (11) shows how point to point and broadcast
links can be implemented via satellite. A satellite consists of receiving antenna and
receiver circuit. The transmitted signal on uplink frequency is received by satellite. The
signal is filtered, amplified and retransmitted on downlink frequency. The transmitter
and receiver of the satellite consists of one channel. It is also called transponder. In one
satellite there are large number of channels. (transponders). Normally, all transponders
have common transmitting and receiving antenna.
Satellite

Transponder

uplink downlink

Antenna Antenna

Transmitter Receiver
Earth Station Earth Station
Fig (11) Point to Point link via satellite

Satellite

Transponder

Uplinks Downlinks

Transmitter Receiver Receiver Receiver

Earth Station Earth Station
Fig (12) Broadcast link via satellite

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 Fig (11) shows the satellite receives from one earth station and transmits to
only one earth station. It is called point to point link. In such link satellite acts as a
repeater station. Very narrow beams are used for such links.
Fig (12) shows that one earth station (transmitter) is transmitting on uplink to
the satellite. The satellite is then transmitting to multiple earth stations on downlink. In
this case, satellite acts as a broadcast station. Normally TV transmission uses such
broadcast link. Wide beams are used for this purpose.

Applications of Satellite Microwave:

 Direct broadcast satellites are used for television distribution.
 Point to point satellites are used for long distance telephone transmission,
connecting the exchange offices, connecting data links between gateways networks.
 VSATs are used private business networks.
 Satellites are used for other applications such as weather forecasting, military
mobile radio transmission etc.

Broadcast Radio:
Broadcast radio is another type of wireless transmission. It uses the frequencies from
30 MHz to 1 GHz for transmission. These frequencies are also called radio waves. The
transmission is mainly omnidirectional. Hence it doesnot require directional antennas.
Broadcast radio also uses line of sight transmission. But attenuation at these
frequencies is relatively low. The distance covered is,

The loss due to attenuation is given as,

Since wavelength is relatively higher for radio waves, they have less
attenuation.

Applications of Broadcast Radio:

 Broadcasting of FM radio.
 Broadcasting of UHF and VHF television.

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 Infrared

Infrared waves, with frequencies from 300 GHz to 400 THz (wavelengths from
1 mm to 770 nm), can be used for short-range communication. Infrared waves, having
high frequencies, cannot penetrate walls. This advantageous characteristic prevents
interference between one system and another; a short-range communication system in
one room cannot be affected by another system in the next room. When we use our
infrared remote control, we do not interfere with the use of the remote by our
neighbors. However, this same characteristic makes infrared signals useless for long-
range communication. In addition, we cannot use infrared waves outside a building
because the sun‘s rays contain infrared waves that can interfere with the
communication.

Applications of Infrared

The infrared band, almost 400 THz, has an excellent potential for data
transmission. Such a wide bandwidth can be used to transmit digital data with a very
high data rate. The Infrared Data Association (IrDA), an association for sponsoring
the use of infrared waves, has established standards for using these signals for
communication between devices such as keyboards, mice, PCs, and printers. For
example, some manufacturers provide a special port called the IrDA port that allows a
wireless keyboard to communicate with a PC. The standard originally defined a data
rate of 75 kbps for a distance up to 8 m. The recent standard defines a data rate of 4
Mbps.

Infrared signals defined by IrDA transmit through line of sight; the IrDA port
on the keyboard needs to point to the PC for transmission to occur.

Types of Errors
Whenever bits flow from one point to another, they are subject to unpredictable
changes because of interference. This interference can change the shape of the signal.
In a single-bit error, a 0 is changed to a I or a I to a 0. In a burst error, multiple bits are
changed. For example, a 1/100s burst of impulse noise on a transmission with a data
rate of 1200 bps might change all or some of the 12 bits of information.

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 Single-Bit Error
The term single-bit error means that only 1 bit of a given data unit (such as a
byte, character, or packet) is changed from 1 to 0 or from 0 to 1.
Figure (2) shows the effect of a single-bit error on a data unit. To understand
the impact of the change, imagine that each group of 8 bits is an ASCII character with
a 0 bit added to the left. In Figure 10.1, 00000010 (ASCII STX) was sent, meaning start
of text, but 00001010 (ASCII LF) was received, meaning line feed.

Fig (2) Single bit error

Single-bit errors are the least likely type of error in serial data transmission. To
understand why, imagine data sent at 1 Mbps. This means that each bit lasts only
1/1,000,000 s, or 1 µs. For a single-bit error to occur, the noise must have a duration of
only 1 µs, which is very rare; noise normally lasts much longer than this.

Burst Error

The term burst error means that 2 or more bits in the data unit have changed
from 1 to 0 or from 0to 1.

Fig (3) Burst error of length 8

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 Figure (3) shows the effect of a burst error on a data unit. In this case,
0100010001000011 was sent, but 0101110101100011 was received. Note that a burst
error does not necessarily mean that the errors occur 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.
A burst error is more likely to occur than a single-bit error. The duration of
noise is normally longer than the duration of 1 bit, which means that when noise affects
data, it affects a set of bits. The number of bits affected depends on the data rate and
duration of noise. For example, if we are sending data at 1 kbps, a noise of 1/100 s can
affect 10 bits; if we are sending data at I Mbps, the same noise can affect 10,000 bits.

Redundancy
The central concept in detecting or correcting errors is redundancy. To be able
to detect or correct errors, we need to send some extra bits with our data. These
redundant bits are added by the sender and removed by the receiver. Their presence
allows the receiver to detect or correct corrupted bits.

Detection versus Correction

The correction of errors is more difficult than the detection. In error detection,
we are looking only to see if any error has occurred. The answer is a simple yes or no.
We are not even interested in the number of errors. A single-bit error is the same for us
as a burst error.
In error correction, we need to know the exact number of bits that are corrupted
and more importantly, their location in the message. The number of the errors and the
size of the message are important factors. If we need to correct one single error in an 8-
bit data unit, we need to consider eight possible error locations; if we need to correct
two errors in a data unit of the same size, we need to consider 28 possibilities. You can
imagine the receiver‘s difficulty in finding 10 errors in a data unit of 1000 bits.
Forward Error Correction versus Retransmission
There are two main methods of error correction. Forward error correction is the
process in which the receiver tries to guess the message by using redundant bits. This is
possible, as we see later, if the number of errors is small. Correction by
retransmission is a technique in which the receiver detects the occurrence of an error
and asks the sender to resend the message. Resending is repeated until a message
arrives that the receiver believes is error-free (usually, not all errors can be detected).

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 Coding
Redundancy is achieved through various coding schemes. The sender adds redundant
bits through a process that creates a relationship between the redundant bits and the
actual data bits. The receiver checks the relationships between the two sets of bits to
detect or correct the errors. The ratio of redundant bits to the data bits and the
robustness of the process are important factors in any coding scheme. Figure (4) shows
the general idea of coding.

Fig (4) the structure of encoder and decoder

We can divide coding schemes into two broad categories:

Block coding and
Convolution coding.

Block coding
In block coding, we divide our message into blocks, each of k bits, called datawords.
We add r redundant bits to each block to make the length n = k + r. The resulting n-bit
blocks are called codewords. How the extra r bits is chosen or calculated is something
we will discuss later. For the moment, it is important to bow that we have a set of
datawords, each of size k, and a set of codewords, each of size of n. With k bits, we can
create a combination of 2k datawords; with n bits, we can create a combination of 2n
codewords. Since n > k, the number of possible codewords is larger than the number of
possible data- words. The block coding process is one-to-one; the same dataword is
always encoded as the same codeword. This means that we have 2n – 2k codewords that
are not used. We call these codewords invalid or illegal. Figure (5) shows the situation.

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 Fig (5) Datawords and codewords in block coding
Error Detection
How can errors be detected by using block coding? If the following two conditions
are met, the receiver can detect a change in the original codeword.
 The receiver has (or can find) a list of valid codewords.
 The original codeword has changed to an invalid one.
Figure (6) shows the role of block coding in error detection.

Fig (6) Process of error detection in block coding

The sender creates codewords out of datawords by using a generator that

applies the rules and procedures of encoding (discussed later). Each codeword sent to
the receiver may change during transmission. If the received codeword is the same as
one of the valid codewords, the word is accepted; the corresponding dataword is
extracted for use. If the received codeword is not valid, it is discarded. However, if the
codeword is corrupted during transmission but the received word still matches a valid
codeword d, the error remains undetected. This type of coding can detect only single
errors. Two or more errors may remain undetected.

Example 10.2
Let us assume that k = 2 and n = 3. Table (1) shows the list of datawords and
codewords. Later, we will see how to derive a codeword from a dataword.

Table (1) A code for error

detection

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 Assume the sender encodes the dataword 01 as 011 and sends it to the receiver.
Consider the following cases:
The receiver receives 011. It is a valid codeword. The receiver extracts the
dataword 01 from it.
The codeword is corrupted during transmission, and 111 is received (the leftmost
bit is corrupted). This is not a valid codeword and is discarded.
The codeword is corrupted during transmission, and 000 is received (the right two
bits are corrupted). This is a valid codeword. The receiver incorrectly extracts the
dataword 00. Two corrupted bits have made the error undetectable.

Error Correction

As we said before, error correction is much more difficult than error detection. In error
detection, the receiver needs to know only that the received codeword is invalid; in
error correction the receiver needs to find (or guess) the original codeword sent. We
can say that we need more redundant bits for error correction than for error detection.
Figure (7) shows the role of block coding in error correction. We can see that the idea
is the same as error detection but the checker functions are much more complex.

Fig (7)
Structure of encoder and decoder in error correction
Example 10.3
Let us add more redundant bits to Example 101 to see if the receiver can correct
an error without knowing what was actually sent. We add 3 redundant bits to the 2-bit
dataword to make 5-bit codewords. Again, later we will show how we chose the
redundant bits. For the moment let us concentrate on the error correction concept.
Table 10.2 shows the datawords and codewords.
Assume the dataword is 01. The sender consults the table (or uses an algorithm)
to create the codeword 01011. The codeword is corrupted during transmission, and
01001 is received (error in the second bit from the right). First, the receiver finds that

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 the received codeword is not in the table. This means an error has occurred. (Detection
must come before correction.) The receiver, assuming that there is only 1 bit corrupted,
uses the following strategy to guess the correct dataword.

 Comparing the received codeword with the first codeword in the table (01001
versus 00000), the receiver decides that the first codeword is not the one that was
sent because there are two different bits.
 By the same reasoning, the original codeword cannot be the third or fourth one in
the table.
The original codeword must be the second one in the table because this is the
only one that differs from the received codeword by 1 bit. The receiver replaces 01001
with 01011 and consults the table to find the data word 01.

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 UNIT – III
Syllabus:
Multiplexing – Types of Multiplexing – Multiplexing Application – Telephone system
– Project 802 – Ethernet – Token bus – Token Ring – FDDI – IEEE 802.6 – SMDS –
Circuit Switching – Packet Switching – Message switching – Connection Oriented and
Connectionless services.

Multiplexing

Whenever the bandwidth of a medium linking two devices is greater than the
bandwidth needs of the devices, the link can be shared. Multiplexing is the set of
techniques that allows the simultaneous transmission of multiple signals across a single
data link. As data and telecommunications use increases, so does traffic. We can
accommodate this increase by continuing to add individual links each time a new
channel is needed; or we can install higher-bandwidth links and use each to carry
multiple signals. Today‘s technology includes high-bandwidth media such as optical
fiber and terrestrial and satellite microwaves. Each has a bandwidth far in excess of
that needed for the average transmission signal. If the bandwidth of a link is greater
than the bandwidth needs of the devices connected to it, the bandwidth is wasted. An
efficient system maximizes the utilization of all resources; bandwidth is one of the
most precious resources we have in data communications.

Fig 3.1 Dividing a link into channels

In a multiplexed system, n lines share the bandwidth of one link. Figure 3.1
shows the basic format of a multiplexed system. The lines on the left direct their
transmission streams to a multiplexer (MUX), which combines them into a single
stream (many-to-one). At the receiving end, that stream is fed into a demultiplexer
(DEMUX), which separates the stream back into its component transmissions (one-to-
many) and directs them to their corresponding lines. In the figure, the word link refers

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 to the physical path. The word channel refers to the portion of a link that carries a
transmission between a given pair of lines. One link can have many (n) channels.

There are three basic multiplexing techniques:

 Frequency-division multiplexing,
 Wavelength-division multiplexing, and
 Time-division multiplexing.

The first two are techniques designed for analog signals, the third, for digital signals.

Fig 3.2 Categories of Multiplexing

Frequency-Division Multiplexing

Frequency-division multiplexing (FDM) is an analog technique that can be

applied when the bandwidth of a link (in hertz) is greater than the combined
bandwidths of the signals to be transmitted. In FDM, signals generated by each sending
device modulate different carrier frequencies. These modulated signals are then
combined into a single composite signal that can be transported by the link. Carrier
frequencies are separated by sufficient bandwidth to accommodate the modulated
signal. These bandwidth ranges are the channels through which the various signals
travel. Channels can be separated by strips of unused bandwidth guard bands to prevent
signals from overlapping. In addition, carrier frequencies must not interfere with the
original data frequencies.

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 Fig 3.3 Frequency – Division Multiplexing

Figure 3.3 gives a conceptual view of FDM. In this illustration, the transmission
path is divided into three parts, each representing a channel that carries one
transmission.

We consider FDM to be an analog multiplexing technique; however, this does not

mean that FDM cannot be used to combine sources sending digital signals. A
digital signal can be converted to an analog signal before FDM is used to

multiplex them.

Multiplexing Process

Figure 3.4 is a conceptual illustration of the multiplexing process. Each source

generates a signal of a similar frequency range. Inside the multiplexer, these similar
signals modulates different carrier frequencies (f1 f2 and f3). The resulting modulated
signals are then combined into a single composite signal that is sent out over a media

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 Fig 3.4 FDM Process

link that has enough bandwidth to accommodate it.

Demultiplexing Process

The demultiplexer uses a series of filters to decompose the multiplexed signal

into its constituent component signals. The individual signals are then passed to a
demodulator that separates them from their carriers and passes them to the output lines.
Figure 3.5 is a conceptual illustration of demultiplexing process.

Fig 3.5 FDM Demultiplexing example

Analog Hierarchy:

One of these hierarchical systems used by AT&T is made up of groups, super groups,
master groups, and jumbo groups (see Figure 3.6).

Fig 3.6 Analog Hierarchy

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 In this analog hierarchy, 12 voice channels are multiplexed onto a higher-
bandwidth line to create a group. A group has 48 kHz of bandwidth and supports 12
voice channels.

At the next level, up to five groups can be multiplexed to create a composite

signal called a supergroup. A supergroup has a bandwidth of 240 kHz and supports up
to 60 voice channels. Supergroups can be made up of either five groups or 60
independent voice channels.

At the next level, 10 supergroups are multiplexed to create a master group. A

master group must have 2.40 MHz of bandwidth, but the need for guard bands between
the supergroups increases the necessary bandwidth to 2.52 MHz. Master groups
support up to 600 voice channels.

Finally, six master groups can be combined into a jumbo group. A jumbo group
must have 15.12 MHz (6 x 2.52 MHz) but is augmented to 16.984 MHz to allow for
guard bands between the master groups.

The Analog Carrier System

To maximize the efficiency of their infrastructure, telephone companies have

traditionally multiplexed signals from lower-bandwidth lines onto higher-bandwidth
lines. In this way, many switched or leased lines can be combined into fewer but bigger
channels. For analog lines, FDM is used.

Other Applications of FDM

A very common application of FDM is AM and FM radio broadcasting.

Radio uses the air as the transmission medium. A special band from 530 to 1700 kHz is
assigned to AM radio. All radio stations need to share this band. As discussed in
Chapter 5, each AM station needs 10 kHz of bandwidth. Each station uses a different
carrier frequency, which means it is shifting its signal and multiplexing. The signal that
goes to the air is a combination of signals. A receiver receives all these signals, but
filters (by tuning) only the one which is desired. Without multiplexing, only one AM
station could broadcast to the common link, the air. However, we need to know that
there is physical multiplexer or demultiplexer here. Multiplexing is done at the data
link layer.

The situation is similar in FM broadcasting. However, FM has a wider band of

88 to 108 MHz because each station needs a bandwidth of 200 kHz.

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 Another common use of FDM is in television broadcasting. Each TV channel
has its own bandwidth of 6 MHz.

The first generation of cellular telephones (still in operation) also uses FDM.
Each user is assigned two 30-kflz channels, one for sending voice and the other for
receiving. The voice signal, which has a bandwidth of 3 kHz (from 300 to 3300 Hz), is
modulated by using FM. Remember that an FM signal has a bandwidth 10 times that of
the modulating signal, which means each channel has 30 kHz (10 x 3) of bandwidth.
Therefore, each user is given, by the base station, a 60-kHz bandwidth in a range
available at the time of the call.

Wavelength-Division Multiplexing

Wavelength-division multiplexing (WDM) is designed to use the high-data-rate

capability of fiber-optic cable. The optical fiber data rate is higher than the data rate of
metallic transmission cable. Using a fiber optic cable for one single line wastes the
available bandwidth. Multiplexing allows us to combine several lines into one.

WDM is conceptually the same as FDM, except that the multiplexing and
demultiplexing involve optical signals transmitted through fiber-optic channels. The
idea is the same: We are combining different signals of different frequencies. The
difference is that the frequencies are very high.

Figure 3.7 gives a conceptual view of a WDM multiplexer and demultiplexer.

Very narrow bands of light from different sources are combined to make a wider band
of light. At the receiver, the signals are separated by the demultiplexer.

Fig 3.7 Wavelength Division Multiplexing

Although WDM technology is very complex, the basic idea is very simple. We
want to combine multiple light sources into one single light at the multiplexer and do
the reverse at the demultiplexer. The combining and splitting of light sources are easily
handled by a prism. Recall from basic physics that a prism bends a beam of light based
on the angle of incidence and the frequency. Using this technique, a multiplexer can be
made to combine several input beams of light, each containing a narrow band of

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 frequencies, into one output beam of a wider band of frequencies. A demultiplexer can
also be made to reverse the process. Figure 6.11 shows the concept.

One application of WDM is the SONET network in which multiple optical fiber
lines are multiplexed and demultiplexed.

A new method, called dense WDM (DWDM), can multiplex a very large
number of channels by spacing channels very close to one another. It achieves even
greater efficiency.

Time-Division Multiplexing

Time-division multiplexing (TDM) is a digital process that allows several

connections to share the high bandwidth of a link. Instead of sharing a portion of the
bandwidth as in FDM, time is shared. Each connection occupies a portion of time in
the link. Figure 6.12 gives a conceptual view of TDM. Note that the same link is used
as in FDM; here, however, the link is shown sectioned by time rather than by
frequency. In the figure, portions of signals 1, 2, 3, and 4 occupy the link sequentially.

Note that in Figure 3.8 we are concerned with only multiplexing, not switching.
This means that all the data in a message from source 1 always go to one specific
destination, be it 1, 2, 3, or 4. The delivery is fixed and unvarying, unlike switching.

Fig 3.8 Time Division Multiplexing

We also need to remember that TDM is, in principle, a digital multiplexing

technique. Digital data from different sources are combined into one time shared link.
However, this does not mean that the sources cannot produce analog data; analog data
can be sampled, changed to digital data, and then multiplexed by using TDM.

TDM is a digital multiplexing technique for combining several low-rate

channels into one high-rate one.

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 We can divide TDM into two different schemes: synchronous and statistical.

Telephone System

Major Components

The telephone network, as shown in Figure 3.9, is made of three major

components:
local loops, trunks, and switching offices. The telephone network has several levels of
switching offices such as end offices, tandem offices, and regional offices.

Fig 3.9 Telephone System

Local Loops

One component of the telephone network is the local loop, a twisted-pair

cable that connects the subscriber telephone to the nearest end office or local central
office. The local loop, when used for voice, has a bandwidth of 4000Hz (4 kHz). It is
interesting to examine the telephone number associated with each local loop. The first
three digits of a local telephone number define the office, and the next four digits
define the local loop number.

Trunks

Trunks are transmission media that handle the communication between offices.
A trunk normally handles hundreds or thousands of connections through multiplexing.
Transmission is usually through optical fibers or satellite links.

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 Switching Offices

To avoid having a permanent physical link between any two subscribers, the
telephone company has switches located in a switching office. A switch connects
several local loops or trunks and allows a connection between different subscribers.
As telephone networks evolved into a complex network, the functionality of the
signaling system increased. The signaling system was required to perform other tasks
such as
i. Providing dial tone, ring tone, and busy tone
ii. Transferring telephone numbers between offices
iii. Maintaining and monitoring the call
iv. Keeping billing information
v. Maintaining and monitoring the status of the telephone network equipment
vi. Providing other functions such as caller ID, voice mail, and so on.

Project 802

In 1985, the Computer Society of the IEEE started a project, called Project
802, to set standards to enable intercommunication among equipment from a variety of
manufacturers. Project 802 does not seek to replace any part of the OSI or the Internet
model. Instead, it is a way of specifying functions of the physical layer and the data
link layer of major LAN protocols.

The standard was adopted by the American National Standards Institute

(ANSI). In 1987, the International Organization for Standardization (ISO) also
approved it as an international standard under the designation ISO 8802.

The relationship of the 802 Standard to the traditional OSI model is shown in
Figure 13.1. The IEEE has subdivided the data link layer into two sublayers: logical
link control (LLC) and media access control (MAC). IEEE has also created several
physical layer standards for different LAN protocols.

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 Fig 3.10 IEEE standard for LANs

The IEEE (Institute of Electrical and Electronic Engineers) is a technical

association of industry professionals with a common interest in advancing all
communications technologies. The previous topic discusses the IEEE organization.
This topic describes the standards developed by the LAN/MAN Standards Committee
(LMSC), which develops LAN (local area network) and MAN (metropolitan area
network) standards, mainly for the lowest two layers in the OSI reference model.
LMSC is also called the IEEE Project 802, so the standards it develops are referenced
as IEEE 802 standards, described next. In general, IEEE 802 standards define physical
network interfaces such as network interface cards, bridges, routers, connectors, cables,
and all the signaling and access methods associated with physical network connections.

Data Link Layer

The data link layer in the IEEE standard is divided into two sublayers:

i. LLC and
ii. MAC.

Logical Link Control (LLC)

That data link control handles framing, flow control, and error control. In
IEEE Project 802, flow control, error control, and part of the framing duties are
collected into one sublayer called the logical link control. Framing is handled in both
the LLC sublayer and the MAC sublayer.

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 The LLC provides one single data link control protocol for all IEEE LANs. In
this way, the LLC is different from the media access control sublayer, which provides
different protocols for different LANs. A single LLC protocol can provide
interconnectivity between different LANs because it makes the MAC sublayer
transparent. Figure 13.1 shows one single LLC protocol serving several MAC
protocols.

Need for LLC

The purpose of the LLC is to provide flow and error control for the upper-layer
protocols that actually demand these services. For example, if a LAN or several LANs
are used in an isolated system, LLC may be needed to provide flow and error control
for the application layer protocols. However, most upper-layer protocols such as IP
(discussed in Chapter 20), do not use the services of LLC. For this reason, we end our
discussion of LLC.

i. Media Access Control (MAC)

Multiple access methods includes random access, controlled access, and

channelization. IEEE Project 802 has created a sublayer called media access control
that defines the specific access method for each LAN. For example, it defines
CSMA/CD as the media access method for Ethernet LANs and the tokena passing
method for Token Ring and Token Bus LANs. As we discussed in the previous section,
part of the framing function is also handled by the MAC layer.

In contrast to the LLC sublayer, the MAC sublayer contains a number of

distinct modules; each defines the access method and the framing format specific to the
corresponding LAN protocol.

Physical Layer

The physical layer is dependent on the implementation and type of physical

media used. IEEE defines detailed specifications for each LAN implementation. For
example, although there is only one MAC sublayer for Standard Ethernet, there is a
different physical layer specifications for each Ethernet implementations as we will see
later.

Ethernet

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 The original Ethernet was created in 1976 at Xerox‘s Palo Alto Research
Center (PARC). it has gone through four generations: Standard Ethernet (10 s), Fast 00
Mbps), Gigabit Ethernet (1 Gbps), and Ten-Gigabit Ethernet (10 0bps), Figure 3.11.
We briefly discuss all these generations starting with the first, traditional) Ethernet.

Fig 3.11 Ethernet evolution through four generation

1. Standard Ethernet

MAC Sublayer

In Standard Ethernet, the MAC sublayer governs the operation of the access
method. It also frames data received from the upper layer and passes them to the
physical layer.

Physical Layer
The Standard Ethernet defines several physical layer implementations; four of
the most common, are shown in Figure 3.12.

Fig 3.12 Categories of Standard Ethernet:

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 10Base5: Thick Ethernet

The first implementation is called 10Base5, thick Ethernet, or Thicknet. The

nickname derives from the size of the cable, which is roughly the size of a garden hose
and too stiff to bend with your hands. 10Base5 was the first Ethernet specification to
use a bus topology with an external transceiver (transmitter/receiver) connected via a
tap to a thick coaxial cable. Figure 3.13 shows a schematic diagram of a 10Base5
implementation.

Fig 3.13 10Base5: Thick Ethernet

10Base2: Thick Ethernet

The second implementation is called 10Base2, thin Ethernet, or Cheapernet.

10Base2 also uses a bus topology, but the cable is much thinner and more flexible. The
cable can be bent to pass very close to the stations. In this case, the transceiver is
normally part of the network interface card (NIC), which is installed inside the station.
Figure 3. 14 shows the schematic diagram of a 10Base2 implementation.

Fig. 3.14 10Base2 implementation

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 10Base- T: Twisted-Pair Ethernet

The third implementation is called 10Base-T or twisted-pair Ethernet. 10Base-T

uses a physical star topology. The stations are connected to a hub via two pairs of
twisted cable, as shown in Figure 3.15. Note that two pairs of twisted cable create two
paths (one for sending and one for receiving) between the station and the hub. Any
collision here happens in the hub. Compared to 10Base5 or 10Base2, we can see that
the hub actually replaces the coaxial cable as far as a collision is concerned. The
maximum length of the twisted cable here is defined as 100 m, to minimize the effect
of attenuation in the twisted cable.

Fig 3.15 10Base-T implementation

10Base-F: Fiber Ethernet

Although there are several types of optical fiber 10-Mbps Ethernet, the most
common is called 10Base-E 10Base-F uses a star topology to connect stations to a hub.
The stations are connected to the hub using two fiber-optic cables, as shown in Figure
3.16,

Fig 3.16 10Base-F implementation

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 3. Fast Ethernet

Fast Ethernet was designed to compete with LAN protocols such as FDDI or
Fiber Channel (or Fibre Channel, as it is sometimes spelled). IEEE created Fast
Ethernet under the name 802.3u. Fast Ethernet is backward-compatible with Standard
Ethernet, but it can transmit data 10 times faster at a rate of 100 Mbps. The goals of
Fast Ethernet can be summarized as follows:

1. Upgrade the data rate to 100 Mbps.

2. Make it compatible with Standard Ethernet.
3. Keep the same 48-bit address.
4. Keep the same frame format.
5. Keep the same minimum and maximum frame lengths.

MAC Sublayer

A main consideration in the evolution of Ethernet from 10 to 100 Mbps was to

keep the MAC sublayer untouched. However, a decision was made to drop the bus
topologies and keep only the star topology. For the star topology, there are two
choices, as we saw before: half duplex and full duplex. In the half-duplex approach, the
stations are connected via a hub; in the full-duplex approach, the connection is made
via a switch with buffers at each port.

The access method is the same (CSMA/CD) for the half-duplex approach; for
full- duplex Fast Ethernet, there is no need for CSMAICD. However, the
implementations keep CSMA/CD for backward compatibility with Standard Ethernet.

Autonegotiation

A new feature added to Fast Ethernet is called autonegotiation. It allows a

station or a hub a range of capabilities. Autonegotiation allows two devices to negotiate
the mode or data rate of operation. It was designed particularly for the following
purposes:

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 1. To allow incompatible devices to connect to one another. For example, a device
with a maximum capacity of 10 Mbps can communicate with a device with a
100 Mbps capacity (but can work at a lower rate).
2. To allow one device to have multiple capabilities.
3. To allow a station to check a hub‘s capabilities.

Physical Layer

The physical layer in Fast Ethernet is more complicated than the one in
Standard Ethernet. We briefly discuss some features of this layer.

Topology

Fast Ethernet is designed to connect two or more stations together. If there are
only two stations, they can be connected point-to-point. Three or more stations need to
be connected in a star topology with a hub or a switch at the center, as shown in Figure
3.17.

Fig 3.17 Fast Ethernet Topology

Implementation

Fast Ethernet implementation at the physical layer can be categorized as either

two-wire or four-wire. The two-wire implementation can be either category 5 UTP
(100BaseTX) or fiber-optic cable (100Base-FX). The four-wire implementation is
designed only for category 3 UTP (100Base-T4). See Figure 3.18.

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 Fig 3.18 Fast Ethernet implementation

3. Gigabit Ethernet

The need for an even higher data rate resulted in the design of the Gigabit
Ethernet protocol (1000 Mbps). The IEEE committee calls the Standard 8O23z. The
goals of the Gigabit Ethernet design can be summarized as follows:

1. Upgrade the data rate to 1 Gbps.

2. Make it compatible with Standard or Fast Ethernet.
3. Use the same 48-bit address.
4. Use the same frame format.
5. Keep the same minimum and maximum frame lengths.
6. To support autonegotiation as defined in Fast Ethernet.

MAC Sublayer

A main consideration in the evolution of Ethernet was to keep the MAC

sublayer untouched. However, to achieve a data rate 1 Gbps, this was no longer
possible. Gigabit Ethernet has two distinctive approaches for medium access: half-
duplex and full duplex.

Almost all implementations of Gigabit Ethernet follow the full-duplex

approach. How- ever, we briefly discuss the half-duplex approach to show that Gigabit
Ethernet can be compatible with the previous generations.

Full-Duplex Mode

In full-duplex mode, there is a central switch connected to all computers or

other switches. In this mode, each switch has buffers for each input port in which data
are stored until they are transmitted. There is no collision in this mode, as we discussed
before. This means that CSMA/CD is not used. Lack of collision implies that the
maximum length of the cable is determined by the signal attenuation in the cable, not
by the collision detection process.

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 In the full-duplex mode of Gigabit Ethernet, there is no collision;
the maximum length of the cable is determined by the signal attenuation in the
cable.

Half-Duplex Mode

Gigabit Ethernet can also be used in half-duplex mode, although it is rare. In

this case, a switch can be replaced by a hub, which acts as the common cable in which
a collision might occur. The half-duplex approach uses CSMAICD. However, as we
saw before, the maximum length of the network in this approach is totally dependent
on the minimum frame size. Three methods have been defined: traditional, carrier
extension, and frame bursting.

Physical Layer

The physical layer in Gigabit Ethernet is more complicated than that in

Standard or Fast Ethernet. We briefly discuss some features of this layer.

Topology

Gigabit Ethernet is designed to connect two or more stations. If there are only
two stations they can be connected point-to-point. Three or more stations need to be
connecting in a star topology with a hub or a switch at the center. Another possible
configuration is to connect several star topologies or let a star topology be part of
another as shown Figure 3.19.

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 Fig 3.19 Topologies of Gigabit Ethernet

Implementation

Gigabit Ethernet can be categorized as either a two-wire or a four-wire

implementation. The two-wire implementations use fiber-optic cable (l000Base-SX,
short-wave, or l000Base-LX, long-wave), or STP (l000Base-CX). The four—wire
version uses category 5 twisted—pair cable (l000Base-T). In other words, we have four
implementations, as shown in Figure 3.20. I000Base-T was designed in response to
those users who had already installed this wiring for other purposes such as Fast
Ethernet or telephone services.

Fig 3.20

4. Ten-Gigabit Ethernet

The IEEE committee created Ten-Gigabit Ethernet and called it Standard

802.3ae. The. goals of the Ten-Gigabit Ethernet design can be summarized as follows:

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 1. Upgrade the data rate to 10 Gbps.
2. Make it compatible with Standard, Fast, and Gigabit Ethernet.
3. Use the same 48-bit address.
4. Use the same frame format.
5. Keep the same minimum and maximum frame lengths.
6. Allow the interconnection of existing LANs into a metropolitan area network
(MAN) or a wide area network (WAN).

MAC Sublayer

Ten -Gigabit Ethernet operates only in full duplex mode which means there is
no need for contention; CSMA/CD is not used in Ten-Gigabit Ethernet.

Physical Layer

The physical layer in Ten-Gigabit Ethernet is designed for using fiber-optic

cable over long distances. Three implementations are the most common: 10GBase-S,
10GBase-L, and 10GBase-E.

Token bus

Fig 3.21

Token passing in a Token bus network

Token bus is a network implementing the token ring protocol over a "virtual
ring" on a coaxial cable. A token is passed around the network nodes and only the node
possessing the token may transmit. If a node doesn't have anything to send, the token is
passed on to the next node on the virtual ring. Each node must know the address of its

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 neighbour in the ring, so a special protocol is needed to notify the other nodes of
connections to, and disconnections from, the ring.

Token bus was standardized by IEEE standard 802.4. It is mainly used for
industrial applications. Token bus was used by GM (General Motors) for their
Manufacturing Automation Protocol (MAP) standardization effort. This is an
application of the concepts used in token ring networks. The main difference is that the
endpoints of the bus do not meet to form a physical ring. The IEEE 802.4 Working
Group is disbanded. In order to guarantee the packet delay and transmission in Token
bus protocol, a modified Token bus was proposed in Manufacturing Automation
Systems and flexible manufacturing system (FMS) [1].

Token Ring

Fig 3.22

A Token Ring network is a local area network (LAN) in which all computers
are connected in a ring or star topology and a bit- or token-passing scheme is used in
order to prevent the collision of data between two computers that want to send
messages at the same time. The Token Ring protocol is the second most widely-used
protocol on local area networks after Ethernet. The IBM Token Ring protocol led to
a standard version, specified as IEEE 802.5. Both protocols are used and are very

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 similar. The IEEE 802.5 Token Ring technology provides for data transfer rates of
either 4 or 16 megabits per second. Very briefly, here is how it works:

1. Empty information frames are continuously circulated on the ring.

2. When a computer has a message to send, it inserts a token in an empty frame
(this may consist of simply changing a 0 to a 1 in the token bit part of the
frame) and inserts a message and a destination identifier in the frame.
3. The frame is then examined by each successive workstation. If the workstation
sees that it is the destination for the message, it copies the message from the
frame and changes the token back to 0.
4. When the frame gets back to the originator, it sees that the token has been
changed to 0 and that the message has been copied and received. It removes the
message from the frame.
5. The frame continues to circulate as an "empty" frame, ready to be taken by a
workstation when it has a message to send.

At the start, a free Token is circulating on the ring, this is a data frame which to
all intents and purposes is an empty vessel for transporting data. To use the network, a
machine first has to capture the free Token and replace the data with its own message.
In the example above, machine 1 wants to send some data to machine 4, so it
first has to capture the free Token. It then writes its data and the recipient's address
onto the Token (represented by the yellow flashing screen).
The packet of data is then sent to machine 2 who reads the address, realizes it is
not its own, so passes it on to machine 3. Machine 3 does the same and passes the
Token on to machine 4.

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 Fig 3.23

This time it is the correct address and so number 4 reads the message
(represented by the yellow flashing screen). It cannot, however, release a free Token on
to the ring, it must first send the message back to number 1 with an acknowledgement
to say that it has received the data (represented by the purple flashing screen).

The receipt is then sent to machine 5 who checks the address, realizes that it is
not its own and so forwards it on to the next machine in the ring, number 6.

Machine 6 does the same and forwards the data to number 1, who sent the
original message.

Machine 1 recognizes the address, reads the acknowledgement from number 4

(represented by the purple flashing screen) and then releases the free Token back on to
the ring ready for the next machine to use.

That's the basics of Token Ring and it shows how data is sent, received and
acknowledged, but Token Ring also has a built in management and recovery system
which makes it very fault tolerant. Below is a brief outline of Token Ring's self
maintenance system.

FDDI

FDDI stands for Fiber Distributed Data Interface. The FDDI standard is ANSI
X3T9.5 . The FDDI topology is ring with two counter rotating rings for reliability
with no hubs. Cable type is fiber-optic. Connectors are specialized. The media access
method is token passing. Multiple tokens may be used by the system. The maximum
length is 100 kilometers. The maximum number of nodes on the network is 500. Speed
is 100 Mbps. FDDI is normally used as a backbone to link other networks. A typical
FDDI network can include servers, concentrators, and links to other networks.
CDDI is a copper version of FDDI which uses category 5 cable. Obviously the distance
is more limited than FDDI.

Devices called concentrators provide functions similar to hubs. Most

concentrators use dual attachment station network cards but single attachment
concentrators may be used to attach more workstations to the network.

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 FDDI token passing allows multiple frames to circulate around the ring at the
same time. Priority levels of a data frame and token can be set to allow servers to send
more data frames. Time sensitive data may also be given higher priority. The second
ring in a FDDI network is a method of adjusting when there are breaks in the cable.
The primary ring is normally used, but if the nearest downstream neighbor stops
responding the data is sent on the secondary ring in attempt to reach the computer.
Therefore a break in the cable will result in the secondary ring being used. There
are two network cards which are:

1. Dual attachment stations (DAS) used for servers and concentrators are attached
to both rings.
2. Single Attachment stations (SAS) attached to one ring and used to attach
workstations to concentrators.

A router or switch can link an FDDI network to a local area network (LAN).
Normally FDDI is used to link LANs together since it covers long distances.

The development of this service has paralleled the emerging Asynchronous

Transfer Mode (ATM) standards. Like ATM, SMDS uses cell relay TRANSPORT.
Both services use 53 octet cells for transport and can accommodate packet lengths of
9188 octets (However, the maximum length for SMDS is 9188 octets and the
maximum length for ATM is 65535 octets.) Because of this, SMDS is considered to be
an intermediate between the packet-switched services offered today and the ATM
service of the future.

Abbreviation of Fiber Distributed Data Interface, a set of ANSI protocols for

sending digital data over fiber optic cable. FDDI networks are token-passing networks,
and support data rates of up to 100 Mbps (100 million bits) per second. FDDI networks
are typically used as backbones for wide-area networks.

An extension to FDDI, called FDDI-2, supports the transmission of voice and

video information as well as data. Another variation of FDDI, called FDDI Full Duplex
Technology (FFDT) uses the same network infrastructure but can potentially support
data rates up to 200 Mbps.

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 IEEE 802.6

IEEE 802.6 is a standard governed by the ANSI for Metropolitan Area

Networks (MAN). It is an improvement of an older standard (also created by ANSI)
which used the Fiber distributed data interface (FDDI) network structure. The FDDI-
based standard failed due to its expensive implementation and lack of compatibility
with current LAN standards. The IEEE 802.6 standard uses the Distributed Queue Dual
Bus (DQDB) network form. This form supports 150 Mbit/s transfer rates. It consists of
two unconnected unidirectional buses. DQDB is rated for a maximum of 160 km
before significant signal degradation over fiberoptic cable with an optical wavelength
of 1310 nm.

This standard has also failed, mostly due to the same reasons that the FDDI
standard failed. Most MANs now use Synchronous Optical Network (SONET) or
Asynchronous Transfer Mode (ATM) network designs, with recent designs using
native Ethernet or MPLS.

SMDS

Switched multimegabit data service (SMDS) was a connectionless service

used to connect LANs, MANs and WANs to exchange data. SMDS was based on the
IEEE 802.6 DQDB standard. SMDS fragmented its datagrams into smaller "cells" for
transport, and can be viewed as a technological precursor of ATM.

Increases in raw data rates removed the need for fragmentation into cells, and
SMDS' niche market position ensured that it remained a high-priced service. As a
result, SMDS has been supplanted by IP-based and Ethernet-based services and MPLS.

Switched Multimegabit Data Service (SMDS) is a telecommunications

service that provides connectionless, high- performance, packet-switched data
transport. Being neither a protocol nor a technology, it supports standard protocols and
communications interfaces using current (and future) technology.

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 SMDS allows users to transparently extend their data communications
capabilities over a wider geographical area. Since it is a service offered by the
telephone companies, SMDS permits this expansion using existing Customer-premises
equipment (CPE) and protocols, with minimal investment in dedicated leased lines as
the number of line terminations increases.

SMDS (Switched Multimegabit Data Service) is a public, packet-switched

service aimed at enterprises that need to exchange large amounts of data with other
enterprises over the wide-area network on a nonconstant or "bursty" basis. SMDS
provides an architecture for this kind of data exchange and a set of services. In general,
SMDS extends the performance and efficiencies of a company's local area network
(LANs) over a wide area on a switched, as-needed basis.

SMDS is connectionless, meaning that there is no need to set up a connection

through the network before sending data. This provides bandwidth on demand for the
"bursty" data transmission typically found on LANs.

Switching Techniques

In its simplest form, data communication takes place between two devices,
which are directly connected by some form of transmission medium- twisted wires,
coaxial cables, microwave and satellite links. Instead, communication is achieved by
transmitting data from source to destination through a network of intermediate nodes.
These nodes provide a switching facility, which moves data from node to node until the
destination is reached. There are three different methods of establishing
communication links between the sender and receiver in a communication network,
namely, circuit switching, message switching and packet switching.

Circuit Switching

It is the simplest method of data communication in which a dedicated physical

path is established between the sending and receiving stations through the nodes of the
network. This method is used to connect two subscribers for a telephone conversation.
Computers and terminals connected through a telephone network also use this method
of establishing communication path among them.

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 Fig 3.23 Circuit Switching

The method of circuit switching is illustrated in fig. each rectangle represents a

switching node of the communication network. When a message to be communicated,
a physical path is established between the two stations, it is exclusively used by the two
parties, and the dedicated physical link between both ends continues to exist, until the
connection is terminated either by the sender or the receiver. As soon as the connection
is terminated by one of the two stations, the dedicated resources are deallocated and
can now be used by other stations also.

Hence, circuit switching involves three phases – circuit establishing, data

transfer and circuit disconnection. It is used in the public Switched Telephone Network
(PSTN).

Message Switching

A message is a logical unit of information and can be of any length. In this

method, if a station wishes to send a message to another station, it first appends the
destination address to the message. After this, the message is transmitted from the
source to its destination either by store-and –forward or broadcast method.

Fig 3.24 Packet Switching

As shown in fig. in the store and forward method, the message is transmitted from the
source node to an intermediate node. The intermediate node stores the complete
message temporarily, inspects it for errors, and transmits it to the next node, based on
an available free channel and its routing information. The actual path taken by the
message to its destination is dynamic, because the path is established as it travels
along. When the message reaches a node, the channel on which it came is released for
use by another message. In fig, if a message is to be transmitted from station A to
station B, it may take either path 1-2-3-4 or 1-5-4, depending on the availability of a
free output path at that particular moment.

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 Fig 3.25

Packet Switching

This method works in a similar fashion as message switching. However, it

overcomes the disadvantages of message switching technique, because in this method
routing is done on ‗packet‘ basis, not on ‗message‘ basis.

A message is split up into ‗packets‘ of a fixed size. Besides the block of data to
be sent, a packet has a header, which contains the destination and source addresses,
control information, message number, number of current and last packet,
synchronization bits, acknowledgement and error checking bytes, etc. like message
switching, the packets may be routed from the sender node to the destination node
either by store-and-forward or broadcast method.

Connection Oriented and Connectionless services

Two distinct techniques are used in data communications to transfer data. Each has
its own advantages and disadvantages. They are the connection-oriented method and
the connectionless method:

 Connection-oriented Requires a session connection (analogous to a phone call)

be established before any data can be sent. This method is often called a "reliable"
network service. It can guarantee that data will arrive in the same order.
Connection-oriented services set up virtual links between end systems through a
network, as shown in Figure 1. Note that the packet on the left is assigned the

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 virtual circuit number 01. As it moves through the network, routers quickly send it
through virtual circuit 01.

 Connectionless Does not require a session connection between sender and

receiver. The sender simply starts sending packets (called datagrams) to the
destination. This service does not have the reliability of the connection-oriented
method, but it is useful for periodic burst transfers. Neither system must maintain
state information for the systems that they send transmission to or receive
transmission from. A connectionless network provides minimal services.

Fig 3.26

Connection-oriented methods may be implemented in the data link layers of the

protocol stack and/or in the transport layers of the protocol stack, depending on the
physical connections in place and the services required by the systems that are
communicating. TCP (Transmission Control Protocol) is a connection-oriented
transport protocol, while UDP (User Datagram Protocol) is a connectionless network
protocol. Both operate over IP.

The physical, data link, and network layer protocols have been used to
implement guaranteed data delivery. For example, X.25 packet-switching networks
perform extensive error checking and packet acknowledgment because the services
were originally implemented on poor-quality telephone connections. Today, networks
are more reliable. It is generally believed that the underlying network should do what it
does best, which is deliver data bits as quickly as possible. Therefore, connection-
oriented services are now primarily handled in the transport layer by end systems, not
the network. This allows lower-layer networks to be optimized for speed.

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 LANs operate as connectionless systems. A computer attached to a network can
start transmitting frames as soon as it has access to the network. It does not need to set
up a connection with the destination system ahead of time. However, a transport-level
protocol such as TCP may set up a connection-oriented session when necessary.

The Internet is one big connectionless packet network in which all packet
deliveries are handled by IP. However, TCP adds connection-oriented services on top
of IP. TCP provides all the upper-level connection-oriented session requirements to
ensure that data is delivered properly. MPLS is a relatively new connection-oriented
networking scheme for IP networks that sets up fast label-switched paths across routed
or layer 2 networks.

A WAN service that uses the connection-oriented model is frame relay. The
service provider sets up PVCs (permanent virtual circuits) through the network as
required or requested by the customer. ATM is another networking technology that
uses the connection-oriented virtual circuit approach.

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 UNIT - IV
Syllabus:

History of Analog and Digital Network – Access to ISDN – ISDN Layers – Broadband
ISDN – X.25 Layers – Packet Layer Protocol – ATM – ATM Topology – ATM
Protocol.

History of Analog and Digital Network

In transmitting data from a source to a destination, one must be concerned with

the nature of the data, the actual physical means used to propagate the data, and what
processing or adjustments may be required along the way to assure that the received
data are intelligible. For all of these considerations, the crucial question is whether we
are dealing with analog or digital entities.
The terms analog and digital correspond, roughly, to continuous and discrete,
respectively. These two terms are used frequently in data communications in at least
three contexts:

Data
Signaling
Transmission

Data can be defined as entities that convey meaning.

Signals are electric or electromagnetic encoding of data.
Signaling is the act of propagating the signal along a suitable medium.
Transmission is the communication of data by the propagation and processing of
signals.

Data
The concepts of analog and digital data are simple enough. Analog data take
continuous values on some interval. For example, voice and video are continuously
varying patterns of intensity. Most data collected by sensors, such as temperature and
pressure, are continuous-valued. The most familiar example of analog data is audio or
acoustic data, which, in the form of sound waves, can be perceived directly by human
beings. Another common example of analog data is video. Here it is easier to
characterize the data in terms of the viewer (destination) of the TV screen rather than
the original scene (source) that is recorded by the TV camera.
Digital data take discrete values; examples are text and integers. Text or
character strings generated from type writer or keyboard is discrete in nature. Such
discrete data is assigned binary (digital) values as per some code.
Signals

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 In a communications system, data are propagated from one point to another by
means of electric signals. For example, the speech data is converted to electric signal
through microphone. Video data is converted to electric signal by video camera.
Depending upon the data, its signal can be analog or digital. For example video and
speech generate analog signals. Text strings generate digital signal.
An analog signal is a continuously varying electromagnetic wave that may be
propagated over a variety of media, depending on spectrum; examples are wire media,
such as twisted pair and coaxial cable, fiber optic cable, and atmosphere or space
propagation. A digital signal is a sequence of voltage pulses that may be transmitted
over a wire medium; for example, a constant positive voltage level may represent
binary 1, and a constant negative voltage level may represent binary 0.

Data and Signals

Analog signals used to represent analog data and digital signals used to
represent digital data. Generally, analog data are a function of time and occupy a
limited frequency spectrum; such data can be represented by an electromagnetic
signal occupying the same spectrum. Digital data can be represented by digital
signals, with a different voltage level for each of the two binary digits.
Analog data – Analog signal:
Analog data such as voice or video can be represented by analog electric
signal. Fig 4.1 shows an example of telephone. Voice sound waves are analog in
nature. They are converted to analog electric signal by telephone instrument.

Fig (4.1) conversion of analog data to analog signal

Digital data – Analog signal:

It is also possible to represent digital data by analog isgnals. An analog carrier
can be modulated by digital data. Modem converts digital data to analog telephone
carriers. Fig 4.2 shows such frequency modulated carrier. The technique used for this
type is binary frequency shift keying (BFSK). Analog representation of digital data
is commonly used for long distance transmission. Normally analog carrier has high
frequency. Hence it travels long distance.

Fig (4.2) conversion of digital data to analog signal

Analog data – Digital Signal:

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 Analog data such as voice, video, temperature etc can also be represented by
digital signals. Normally pulse code modulation, delta modulation or differential
pulse code modulation can be used to convert analog data to digital signal. By means
of this digital transmission of analog signals is possible. It is possible to digitally
multiplex voice, video etc signals. Digital storage of analog signals on compact discs
is also possible due to this conversion.

Fig (4.3) conversion of analog data to analog signal

Digital data Digital signal:

Binary data can be represented by discrete amplitude levels. Binary ‘1’ can be
represented by +5V and ‘0’ can be represented by -5V. This is called data encoding.
Fig shows the digital transmitter. This transmitter encodes digital data into digital
(discrete amplitude) signal. Such digital transmitter is LAN cards, output/input ports
of microprocessors display devices, printers etc.

Fig (4.4) conversion of analog data to analog signal

Signals can be analog or digital. Analog signals can have an infinite number
of values in a range; digital signals can have only a limited number of values.

Transmission
Both analog and digital signals may be transmitted on suitable transmission
media. The way these signals are treated is a function of the transmission system.

Analog transmission
Analog transmission is a means of transmitting analog signals without regard
to their content; the signals may represent analog data (e.g., voice) or digital data
(e.g., binary data that pass through a modem). In either case, the analog signal will
become weaker (attenuated) after a certain distance. To achieve longer distances, the
analog transmission system includes amplifiers that boost the energy in the signal.
Unfortunately, the amplifier also boosts the noise components. With amplifiers
cascaded to achieve long distances, the signal becomes more and more distorted.

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 For analog data, such as voice, quite a bit of distortion can be tolerated and
the data remain intelligible. However, for digital data, cascaded amplifiers will
introduce errors.

Digital transmission
Digital transmission, in contrast, is concerned with the content of the signal. A
digital signal can be transmitted only a limited distance before attenuation endangers
the integrity of the data. To achieve greater distances, repeaters are used. A repeater
receives the digital signal, recovers the pattern of 1s and 0s, and retransmits a new
signal, thereby overcoming the attenuation.
The same technique may be used with an analog signal if it is assumed that the
signal carries digital data. At appropriately spaced points, the transmission system has
repeaters rather than amplifiers. The repeater recovers the digital data from the analog
signal and generates a new, clean analog signal. Thus, noise is not cumulative.
The preferred method of transmission – both supplied by the
telecommunications industry and its customers is digital, this despite an enormous
investment in analog communications facilities. Both long-haul telecommunications
facilities and intrabuilding services are gradually being converted to digital
transmission and, where possible, digital signaling techniques.
The most important reasons are (advantages of digital transmission):
Digital technology.
The advent of large-scale integration (LSI) and very large- scale integration
(VLSI) technology has caused a continuing drop in the cost and size of digital
circuitry. Analog equipment has not shown a similar drop.
Data integrity.
With the use of repeaters rather than amplifiers, the effects of noise and other
signal impairments are not cumulative. It is possible, then, to transmit data longer
distances and over lesser quality lines by digital means while maintaining the integrity
of the data.
Capacity utilization.
It has become economical to build transmission links of very high
bandwidth, including satellite channels and connections involving optical fiber. A
high degree of multiplexing is needed to effectively utilize such capacity, and this is
more easily and cheaply achieved with digital (time- division) rather than analog
(frequency-division) techniques.
Security and privacy.
Encryption techniques can be readily applied to digital data and to analog data
that have been digitized.
Integration.

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 By treating both analog and digital data digitally, all signals have the same
form and can be treated similarly. Thus, economies of scale and convenience can be
achieved by integrating voice, video, and digital data.

ISDN

Basics of ISDN

ISDN (Integrated Services Digital Network) is an all digital communications

line that allows for the transmission of voice, data, video and graphics, at very high
speeds, over standard communication lines. ISDN provides a single, common
interface with which to access digital communications services that are required by
varying devices, while remaining transparent to the user. Due to the large amounts of
information that ISDN lines can carry, ISDN applications are revolutionizing the way
business communicate.

ISDN is based on a number of fundamental building blocks. First, there are two
types of ISDN "channels" or communication paths:
 B-channel
The Bearer ("B") channel is a 64 kbps channel which can be used for voice,
video, data, or multimedia calls. B-channels can be aggregated together for even
higher bandwidth applications.
 D-channel
The Delta ("D") channel can be either a 16 kbps or 64 kbps channel used
primarily for communications (or "signaling") between switching equipment in the
ISDN network and the ISDN equipment at your site.
These ISDN channels are delivered to the user in one of two pre-defined
configurations:
 Basic Rate Interface (BRI)
BRI is the ISDN service most people use to connect to the Internet. An ISDN
BRI connection supports two 64 kbps B-channels and one 16 kbps D-channel over a
standard phone line. BRI is often called "2B+D" referring to its two B-channels and

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 one D-channel. It is used by single line business customers for typical desktop type
applications.

 Primary Rate Interface (PRI)

ISDN PRI service is used primarily by large organizations with intensive
communications needs. An ISDN PRI connection supports 23 64 kbps B-channels
and one 64 kbps D-channel (or 23B+D) over a high speed DS1 (or T-1) circuit. T
BRI is the most common ISDN service for Internet access. A single BRI line
can support up to three calls at the same time because it is comprised of three channels
(2B+D). Two voice, fax or data "conversations," and one packet switched data
"conversation" can take place at the same time. Multiple channels or even multiple BRI
lines can be combined into a single faster connection depending on the ISDN
equipment you have. Channels can be combined as needed for a specific application (a
large multimedia file transfer, for example), then broken down and reassembled into
individual channels for different applications (normal voice or data transmissions).
What Do We Use It For?
ISDN offers the speed and quality that previously was only available to people
who bought expensive, point-to-point digital leased lines. Combined with its flexibility
as a dial-up service, ISDN has become the service of choice for many communications
applications.
Popular ISDN applications include:
 Internet access
 Telecommuting/remote access to corporate computing
 Video conferencing
 Small and home office data networking

Why Should We Use ISDN to Access the Internet?

More and more people are discovering that ISDN is the right Internet answer.
As the Internet becomes more and more information-intensive with graphics,
sound, video and multimedia, our ability to take advantage of these new resources
depends on the speed of our Internet connection.
With ISDN, your Internet access is:
 Even faster
By combining your two B-channels you have access to up to 128 kbps -- more
than four times as fast as a 28.8 kbps modem on a standard phone line. And ISDN's
digital technology assures you the cleanest connection to the Internet so you won't be
slowed down by re-transmissions because of old analog technology.
 More efficient and economical

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 ISDN brings increased capabilities, reduced costs and improved
productivity to organizations both large and small. When you're looking for something
on the Internet, you can get there faster. You can be more productive because you
aren't waiting as long to get to that next website or download that large file.

Access to ISDN

To enhance your network connection speed, you can use an Integrated Services
Digital Network (ISDN) line. Whereas standard phone lines typically transmit at up to
56 kilobits per second (Kbps), ISDN lines can transmit at speeds of 64 or 128 Kbps.
An ISDN line must be installed by the phone company at both the server and
at the remote site. ISDN also requires that you install an ISDN adapter on the server
and your computer. Costs for ISDN equipment and lines may be higher than standard
modems and phone lines. However, the speed of communication reduces the
duration of the connection, possibly saving toll charges.
An ISDN line comes with two B channels that transmit data at 64 Kbps, and
one D channel for signaling that transmits data at 16 Kbps. We can configure each
B channel to operate as a port. With some ISDN drivers, we can aggregate the
channels. This means we can statically assign a higher bandwidth by configuring
both B channels to act as a single port. With this configuration, line speed increases
to 128 Kbps.
The Multilink feature performs channel aggregation for ISDN. Multilink
combines multiple physical links into a logical bundle. This aggregate link increases
the bandwidth of a connection. In addition, you can allocate multiple links
dynamically, which means ISDN lines are used only as they are required. This
eliminates excess bandwidth, representing a significant efficiency advantage to users.

ISDN Layers

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 Fig (4.5) layers of ISDN
Table 1
OSI Layer B Channel D Channel
3 IP DSSI (Q.931)
2 HDLC/PPP LAP – D (Q.921)
1 I.430/I.431 or ANSI T1.601

Layer 1 (Physical Layer)

ISDN physical layer (Layer 1) frame formats depends upon whether the frame
is outbound (from terminal to network) or inbound (from network to terminal).
Multiple ISDN devices can be attached physically to one circuit. In this
configuration, collisions occur when two terminals transmit simultaneously. Therefore,
ISDN avoids collision by using link contention.
At the physical layer the ITU has defined the user network interface standard as
I.430 for Basic Rate Access and I.431 for Primary Rate Access. ANSI has defined
the user network interface standard as T1.601. The physical layer uses the normal
telephone cabling as its physical cabling structure.
Layer 2 (Data Link Layer)
ISDN Layer 2 uses signaling protocol called as Link Access Procedure D
channel (LAPD). LAPD is just like High-Level Data Link Control (HDLC) and
Link Access Procedure Balanced (LAPB). The LAPD frame format is as of HDLC.
Like HDLC, LAPD makes use of information, supervisory and unnumbered frames.
The ISDN B channels will typically utilize a Point-to-Point protocol such as
HDLC (High-Level Data Link Control) or PPP frames at Layer 2.

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 Layer 3 (Network Layer)
ISDN Layer 3 is used for ISDN signaling: ITU-T I.450 and ITU-T I.451. Both
these protocols carry end-to-end, circuit-switched and packet-switched connections. A
variety of call-establishment, call-termination, information, and miscellaneous
messages are described by using SETUP, CONNECT, RELEASE, USER
INFORMATION, CANCEL, STATUS, and DISCONNECT.
At layer 3 we typically see IP packets. ISDN operates in Full-Duplex which
means that traffic can be received and transmitted at the same time.
The ISDN D channel will utilize different signaling protocols at Layer 3 and
Layer 2 of the OSI Model. Typically at Layer 2, LAP-D (Link Access Procedure – D
Channel) is the Q.921 signaling used and DSS1 (Digital Subscriber Signaling System
No.1) is the Q.931 signaling that is used at Layer 3. It is easy to remember which one
is used at which layer by simply remembering that the middle number corresponds
to the layer it operates at.

Different ISDN Components

ISDN components include terminals, terminal adapters (TAs), network-

termination devices, line-termination equipment, and exchange-termination
equipment.

ISDN terminals come in two types. Specialized ISDN terminals are referred to
as terminal equipment type 1 (TEl). Non-ISDN terminals, such as DTE, that predate
the ISDN standards are referred to as terminal equipment type 2 (TE2). TE1 is
connected to the ISDN network through a four-wire, twisted-pair digital link. TE2 is
connected to the ISDN network through a TA. The ISDN TA can be either a
standalone device or a board inside the TE2. If the TE2 is implemented as a
standalone device, it connects to the TA via a standard physical-layer interface.
Examples include EIA/TIA-232-C (formerly RS-232C), V.24, and V.35.

Beyond the TEl and TE2 devices, the next connection point in the ISDN
network is the network termination type 1 (NT 1) or network termination type 2
(NT2) device. These are network-termination devices that connect the four-wire
subscriber wiring to the conventional two-wire local loop. In North America, the NT1
is a customer premises equipment (CPE) device. In most other parts of the world, the
NT1 is part of the network provided by the carrier. The NT2 is a more complicated
device that typically is found in digital private branch exchanges (PBXs) and that
performs Layer 2 and 3 protocol functions and concentration services. An NT1/2
device also exists as a single device that combines the functions of an NT1 and an
NT2.

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 ISDN specifies a number of reference points that define logical interfaces
between functional groupings, such as TAs and NTs. ISDN reference points include
the following:

R—The reference point between non-ISDN equipment and a TA.

S—The reference point between user terminals and the NT2.

T—The reference point between NT1 and NT2 devices.

U—The reference point between NT1 devices and line-termination equipment in

the carrier network. The U reference point is relevant only in North America, where
the NT 1 function is not provided by the carrier network.

Figure 12-1 illustrates a sample ISDN configuration and shows three devices
attached to an ISDN switch at the central office. Two of these devices are ISDN-
cornpatible, so they can be attached through an S reference point to NT2 devices. The
third device (a standard, non-ISDN telephone) attaches through the reference point to
a TA. Any of these devices also could attach to an NT 1/2 device, which would replace
both the NT1 and the NT2. In addition, although they are not shown, similar user
stations are attached to the far right ISDN switch.

Fig (4.6) ISDN components

Broadband ISDN

Broadband - A service or a system requiring transmission channels capable of

supporting rates greater than the primary rate. Any service inquiry with a speed greater

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 than1.544 Mbps is defined as broadband, and any communications based on this
speed are called broadband communications.

Broadband ISDN is based

on cell switching instead of circuit switching. It uses fiber optic cables for
transmission and provides data transfer rates upto 155 Mbps. B-ISDN uses ATM and
the ATM reference model. B-ISDN can be used form video telephony, audio and
video data, high speed data transfer, in addition to broadcasting television programs.

Asynchronous transfer mode (ATM) is the transfer mode for implementing

B-ISDN and is independent of the means of transport at the Physical layer.

The goal of B-ISDN is to achieve complete integration of services, ranging

from low-bit-rate bursty signals to high-bit-rate continuous real-time signals.

Broadband ISDN Architecture

B-ISDN differs from a narrowband ISDN in a number of ways. To meet the

requirement for high-resolution video, an upper channel rate of approximately 150
Mbps is needed. To simultaneously support one or more interactive and distributive
series, a total subscriber line rate of about 600 Mbps is needed. The only appropriate
technology for widespread support of such data rates is optical fiber. Hence, the
introduction of B-ISDN depends on the pace of introduction of fiber subscriber loops.

Functional Architecture

Figure 4.7 depicts the functional architecture of B-ISDN.

Fig (4.7) Architecture of Broadband ISDN

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 The control of B-ISDN is based on common-channel signaling. B-ISDN must
support all of the 64-kbps transmission services, both circuit-switching and packet-
switching. This protects the user‘s investment and facilitates migration from
narrowband to broadband ISDN. In addition, broadband capabilities arc provided for
higher data- rate transmission services. At the user-network interface, these
capabilities will be provided with the connection-oriented asynchronous transfer mode
(ATM) facility.

User-Network Interface

Fig (4.8) Components of B - ISDN

Figure A.4, shows the reference configuration for B-ISDN. In order to clearly
illustrate the broadband aspects, the notations for reference points and functional
groupings are appended with the letter B (e.g.. B-NT1, B-TE). The broadband
functional groups are equivalent to the functional groups defined for narrowband

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 1SDN, and are discussed below. Interfaces at (he R reference point may or may not
have broadband capabilities.

Transmission Structure

In terms of data rates available to B-ISDN subscribers, three new transmission

services are defined. The first of these consists of a full-duplex 155.52-Mbps service.
The second service defined is asymmetrical, providing transmission from the
subscriber to the network at 155.52 Mbps, and in the other direction at 622.08
Mbps; and the highest-capacity service yet defined is a full-duplex, 622.08-Mbps
service.

A data rate of 155.52 Mbps can certainly support all of the narrowband ISDN
services. That is, such a rate readily supports one or more basic- or primary-rate
interfaces; in addition, it can support most of the B-ISDN services. At that rate, one or
several video channels can be supported, depending on the video resolution and the
coding technique used. Thus, the full-duplex 155.52-Mbps service will probably be
the most common B-ISDN service.

The higher data rate of 622.08 Mbps is needed to handle multiple video
distribution, such as might be required when a business conducts multiple simultaneous
videoconferences. This data rate makes sense in the network-to-subscriber direction.
The typical subscriber will not initiate distribution services and thus would still be able
to use the lower, 155.52-Mbps service. The full-duplex, 622.08- Mbps service would
be appropriate for a video-distribution provider.

Broadband ISDN Protocols

The protocol architecture for B-ISDN introduces some new elements not found
in the ISDN architecture, as depicted in Figure 4.8. For B-ISDN, it is assumed that the
transfer of information across the user-network interface will use ATM.

The decision to use ATM for B-ISDN is a remarkable one; it implies that B-
ISDN will be a packet-based network, certainly at the interface, and almost certainly
in terms of its internal switching. Although the recommendation also states that B-
ISDN will support circuit-mode applications, this will be done over a packet- based
transport mechanism. Thus, ISDN, which began as an evolution from the circuit-
switching telephone network, will transform itself into a packet-switching network
as it takes on broadband services.

The protocol reference model makes reference to three separate planes:

 User Plane. Provides for user-information transfer, along with associated

controls (e.g., flow control, error control).
 Control Plane. Performs call-control and connection-control functions.

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  Management Plane.
1. Includes plane management, which performs management functions related
to a system as a whole and provides coordination between all the planes, and
2. layer management, which performs management functions relating to
resources and parameters residing in its protocol entities.

Fig (4.9) Protocols of B - ISDN

Table A.6 highlights the functions to be performed at each sub layer.

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 Fig (4.10) Functions of ISDN layers

1. Physical Layer Functions

It is divided into two sublayers:

1. Physical medium: It is the lowest layer of the B-ISDN protocol, and it

includes the functions that are only physical-medium-dependent. It itself
provides line coding
and if necessary, electrical to optical conversion.

2. Transmission convergence: The main functions of this sub layer are cell rate
decoupling, HEC (Header Error Control) header sequence generation, cell

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 delineation, transmission frame adaptation, transmission frame
generation.

2. ATM Layer functions

 generic flow control
 cell header generation
 virtual channel identifier
 cell multiplexing and demultiplexing

3. ATM adaptation layer function:

 The basic function of the AAL is the enhanced adaptation of the services
provided by the ATM layer until the requirement of the higher layer‘s services
are met.
 In this layer, the higher layer protocol data units are mapped onto the
information field of the ATM cell, which is 48 bytes long.
X.25 Layers

X.25 is an International Telecommunication Union–Telecommunication

Standardization Sector (ITU-T) protocol standard for WAN communications that
defines how connections between user devices and network devices are established
and maintained. X.25 is designed to operate effectively regardless of the type of
systems connected to the network. It is typically used in the packet-switched
networks (PSNs) of common carriers, such as the telephone companies. Subscribers are
charged based on their use of the network. The development of the X.25 standard was
initiated by the common carriers in the 1970s. At that time, there was a need for WAN
protocols capable of providing connectivity across public data networks (PDNs). X.25
is now administered as an international standard by the ITU-T.

X.25 Devices and Protocol Operation

X.25 network devices fall into three general categories:

 Data Terminal Equipment (DTE),

 Data Circuit-terminating Equipment (DCE), and
 Packet Switching Exchange (PSE).

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 Data terminal equipment devices (DCEs) are end systems that communicate
across the X.25 network. They are usually terminals, personal computers, or
network hosts, and are located on the premises of individual subscribers. DCE devices
are communications devices, such as modems and packet switches that provide the
interface between DTE devices and a PSE and are generally located in the carrier‘s
facilities. PSEs are switches that compose the bulk of the carrier‘s network. They
transfer data from one DTE device to another through the X.25 PSN.

Figure 4.11 illustrates the relationships between the three types of X.25 network
devices.

Fig (4.11) DTEs, DCEs, PSEs make up an X.25 network

Packet Assembler/Disassembler (PAD)

The packet assembler/disassembler (PAD) is a device commonly found in

X.25 networks. PADs are used when a DTE device, such as a character-mode
terminal, is too simple to implement the full X.25 functionality.
The PAD is located between a DTE device and a DCE device, and it performs
three primary functions:

buffering,

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 packet assembly, and
packet disassembly.

The PAD buffers data sent to or from the DTE device.

It also assembles outgoing data into packets and forwards them to the DCE
device. (This includes adding an X.25 header.)
Finally, the PAD disassembles incoming packets before forwarding the
data to the DTE. (This includes removing the X.25 header.)
Figure 4.12 illustrates the basic operation of the PAD when receiving packets
from the X.25 WAN.

Fig (4.12) The PAD buffers assembles, disassembles the data packets.
X.25 Session Establishment
X.25 sessions are established when one DTE device contacts another to
request a communication session. The DTE device that receives the request can
either accept or refuse the connection. If the request is accepted, the two systems
begin full-duplex information transfer. Either DTE device can terminate the
connection. After the session is terminated, any further communication requires the
establishment of a new session.

X.25 Virtual Circuits

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 A virtual circuit is a logical connection created to ensure reliable
communication between two network devices. A virtual circuit denotes the existence
of a logical, bidirectional path from one DTE device to another across an X.25
network. Physically, the connection can pass through any number of intermediate
nodes, such as DCE devices and PSEs. Multiple virtual circuits (logical connections)
can be multiplexed onto a single physical circuit (a physical connection). Virtual
circuits are demultiplexed at the remote end, and data is sent to the appropriate
destinations. Figure 4.13 illustrates four separate virtual circuits being multiplexed onto
a single physical circuit.

Fig (4.13) Virtual circuits can be multiplexed onto a single physical circuit.

Two types of X.25 virtual circuits exist: switched and permanent.

1. Switched virtual circuits (SVCs) are temporary connections used for sporadic
data transfers. They require that two DTE devices establish, maintain, and
terminate a session each time the devices need to communicate.
2. Permanent virtual circuits (PVCs) are permanently established connections
used for frequent and consistent data transfers. PVCs do not require that
sessions be established and terminated. Therefore, DTEs can begin transferring
data whenever necessary, because the session is always active.
The basic operation of an X.25 virtual circuit begins when the source DTE
device specifies the virtual circuit to be used (in the packet headers) and then sends
the packets to a locally connected DCE device. At this point, the local DCE device
examines the packet headers to determine which virtual circuit to use and then
sends the packets to the closest PSE in the path of that virtual circuit. PSEs
(switches) pass the traffic to the next intermediate node in the path, which may be
another switch or the remote DCE device. When the traffic arrives at the remote
DCE device, the packet headers are examined and the destination address is
determined. The packets are then sent to the destination DTE device. If

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 communication occurs over an SVC and neither device has additional data to
transfer, the virtual circuit is terminated.

The X.25 layers:

X.25 originally defined three basic protocol levels or architectural layers. In the
original specifications these were referred to as levels and also had a level number,
whereas all ITU-T X.25 recommendations and ISO 8208 standards released after 1984
refer to them as layers. The layer numbers were dropped to avoid confusion with the
OSI Model layers.

 Physical layer:
This layer specifies the physical, electrical, functional and procedural
characteristics to control the physical link between a DTE and a DCE. Common
implementations use X.21, EIA-232, EIA-449 or other serial protocols.
 Data link layer:
The data link layer consists of the link access procedure for data
interchange on the link between a DTE and a DCE. In its implementation, the
Link Access Procedure, Balanced (LAPB) is a data link protocol that manages a
communication session and controls the packet framing. It is a bit-oriented
protocol that provides error correction and orderly delivery.
 Packet layer:
This layer defined a packet-layer protocol for exchanging control and user
data packets to form a packet-switching network based on virtual calls,
according to the Packet Layer Protocol.

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 Fig (4.14)

Packet Layer Protocol

Packet Layer Protocol or PLP is the Network Layer protocol for the X.25
protocol suite. PLP manages the packet exchanges between DTE (data terminal)
devices across VCs (virtual calls). PLP also can be used on ISDN using Link Access
Procedures, D channel (LAPD).

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 There are 5 modes of PLP:
 call setup,
 data transfer,
 idle,
 call clearing and
 restarting.
 Call setup mode is used to create VCs (virtual calls) between DTE devices. A
PLP uses the 14-digit X.121 addressing scheme to set up the virtual call.
 Data transfer mode is used to send data between DTE devices across a virtual
call. At this level PLP handles segmentation and reassembly, bit padding, error
control and flow control.
 Idle mode is used when a virtual call is established but there is no data transfer
happening.
 Call clearing mode is used to end sessions between DTE devices and to terminate
VCs.
 Restarting mode is used to synchronize the transmission between a DTE device
and its locally connected DCE (data communications) device.

Packet layer protocol fields

Fig (4.15) Packet Layer Protocol fields

There are 4 types of PLP packet fields:

1. General Format Identifier (GFI)

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 It identifies the packet's parameters such as whether the packet contains Data
or Control information, type of windowing performed, and whether to use Delivery
Confirmation.

2. Logical Channel Identifier (LCI)

It identifies the virtual circuit across the local DTE/DCE interface.

3. Packet Type Identifier (PTI)

It identifies one of the following 17 PLP packet types:

1. CALL ACC Call Accept

2. CALL REQ Call Request

3. CLR CNF Clear Confirmation

4. CLR REQ Clear Request

5. DATA Data Packet

6. DIAG Diagnostic

7. INT CNF Interrupt Confirmation

8. INT REQ Interrupt Request

9. REJ Reject

10. RES CNF Reset Confirmation

11. RES REQ Reset Request

12. RNR Receive Not Ready

13. RR Receive Ready

14. RSTR CNF Restart Confirmation

15. RSTR REQ Restart Request

16. REG REQ Registration Request

17. REG CNF Registration Confirmation

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 4. User Data

It encapsulated data from an upper-layer protocol such as TCP/IP. This

field only exists in data packets.

ATM

The asynchronous transfer mode (ATM) is a reference model that uses

asynchronous transmission techniques for data transmission.

Introduction to ATM

ATM is connection-oriented, cell-switching technology. As that word

asynchronous suggests, data in ATM can be sent by any time as opposed to
synchronous systems in which data can only be sent at regular intervals. Data in ATM
is transmitted in chunks of 53 bytes. These fixed-size chunks of data are known as
cells.

Cell switching provides various advantages over packet switching.

Those advantages are:

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  Variable data rates can be handled easily because of cell switching. This is
important for transmitting video and audio data. Cell switching provides ATM
networks flexibility to handle data arriving at variable data rates from various
sources.
 Cell switching provides high speeds of operation. It is easy to switch the packets
because their size is fixed. Speeds of the order of 10 Gbps are possible with
ATM.
 Cell switching in ATM enables broadcast of data. In circuit switching, the
broadcast of data is not possible.
 In addition, ATM provides Quality of Service (QoS) feature that allows you to get
a guaranteed bandwidth. You can give precedence to voice and video data over
other data. This results in better management of bandwidth.

ATM logical Connections:

Connection between two endpoints is accomplished through transmission

paths TPs), virtual paths (VPs), and virtual circuits (VCs).

1. A transmission path (TP) is the physical connection (wire, cable, satellite, and
so on) between an endpoint and a switch or between two switches.
Think of two switches as two cities. A transmission path is the Set of all
highways that directly connect the two cities. A transmission path is divided into
several virtual paths.

2. A virtual path (VP) provides a connection or a set of connections between two

switches.
Think of a virtual path as a highway that connects two cities. Each highway
is a virtual path; the set of all highways is the transmission path.

3. Cell networks are based on virtual circuits (VCs). All cells belonging to a single
message follow the same virtual circuit and remains in their original order until
they reach their destination.
Think of a virtual circuit as the lanes of a highway (virtual path).

Figure 18.10 shows the relationship between a transmission path (a physical

connection), virtual paths (a combination of virtual circuits that are bundled together
because parts of their paths are the same), and virtual circuits that logically connect two
points.

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 Fig (4.16) Relationship between TP, VP and VC

Multiple VCs are combined together to form a virtual path (VP). Virtual
paths are identified by a value known as the virtual path identifier (VPI). A
transmission path is a collection of VPs.

ATM TOPOLOGY

Architecture
ATM is a cell-switched network. The user access devices, called the
endpoints, are connected through a user-to-network interface (UNI) to the switches
inside the network. The switches are connected through network-to-network
interfaces (NNIs). Figure 18.9 shows an example of an ATM network.

Fig (4.17) ATM network

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 ATM Reference Model

The ATM reference model is a three-dimensional model. It consists of layers

and an additional dimension of planes.

Figure depicts the ATM three-dimensional reference model.

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 Fig (4.18) ATM reference model

The following three planes exist in the ATM reference model:

 Control plane: Responsible for controlling the operation of all the layers by
generating appropriate signals.
 User plane: Responsible for managing the transfer data.
 Management plane: Responsible for managing layer-specific issues such as
error control.
 Plane management plane: Responsible for managing the entire system and
coordinating various functions of the planes, such as resource reservation.

The ATM reference model has three layers. These are:

1. Physical layer
2. ATM layer
3. ATM adaptation layer

1. Physical layer

The ATM physical layer corresponds to the OSI physical layer. The ATM
physical layer is responsible for defining the electric and physical characteristics of
physical media.

Functions of the ATM physical layer:

 Interfacing with the physical media to send and receive data

 Converting received bits into cells
 Packaging cells into frames for the transmission medium

The ATM physical layer consists of the following two sublayers:

 Physical medium-dependant (PMD):

This sublayer defines the standards for physical media, connectors, and
cables. ATM can be used on a variety of physical media. Synchronous Optical

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 Network (SONET) is one of the most widely used physical media standard for TAM.
Some of the other standards are DS-3, FDDI, and shielded-twisted pair (STP).
 Transmission-convergence (TC):
This sublayer is responsible for packaging the cells into appropriate frames as
required by a particular transmission medium. In addition, the TC sublayer is
responsible for error control and cell-rate decoupling. Cell-rate decoupling refers to the
adaption of the rate of transmission of cell to the capacity of transmission media.
2. ATM Layer
The ATM layer, along with the ATM adaption layer, corresponds to the data
link layer of the OSI model.

Functions of the ATM layer:

 Defining the cell layout and the header information for the cell
 Establishing and managing the connections

First five bytes contain the header information and remaining 48 bytes contain the
data.

There are two types of ATM cells:

 User-to-network interface (UNI): UNI refers to the interface between the

network terminating equipment, such as routers and hosts to the ATM switches.
ATM switches control the flow of cells through the ATM network.
 Node-network interface (NNI): NNI refers to the connection between two
ATM switches.

3. ATM Adaptation Layer (AAL):

The function of the ATM adaption layer is to accept packets from the upper
layers, and pass them to the ATM layer.

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 AAL consists of two sublayers:

i. Convergence sublayer (CS):

Data from upper layer is passed into the convergence sublayer. The
convergence sublayer has two parts:

 Service Specific Convergence sublayer (SSCS):

SSCS has two functions, to provide clock recovery and to identify
messages.
 Common Part Convergence sublayer (CPCS):
CPCS enable sequence error detection.

ii. Segmentation and Reassembly sublayer (SAR):

This sublayer is responsible for the segmentation of data received from
upper layers. The SAR sublayer is also responsible for the reassembly of segments
received for AAL.

There are three types of AAL:

i. AAL1: It provides connection-oriented service. Connection-oriented services

are required for transferring audio and video data. This layer requires a
synchronization mechanism to indicate the start and end of data between the
sender and receiver. SONET is one of the physical layer standards that supports
synchronization.
ii. AAL3/4: It provides both connection-oriented and connectionless services.
iii. AAL5: Provides both connection-oriented and connectionless service. This
layer is used for transmitting IP traffic over ATM networks.

ATM Protocol

Protocol Structure – ATM:

ATM Cell Format:

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 Fig (4.19a)

Figure shows the two basic types of cell.

Fig (4.19b)
(a) ATM cells at the user-network interface; (b) ATM cells at the network-node
interface

Each ATM cell consists of 53 bytes: the header is five bytes long and the
remaining 48 bytes (the cell payload) carry information from higher layers. The only
difference between the two types of ATM cell is that the cells at the user-network
interface carry a data field for the flow control of data from users. This means that

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 only eight bits are available for virtual path identifiers, rather than 12 bits at the
network-node interface.
The virtual connections set up in ATM networks are identified by the
combination of the virtual path identifier and virtual channel identifier fields shown
in Fig (4.19a). These two fields provide a hierarchy in the numbering of virtual
connections, whereby a virtual path contains a number of virtual channels as is
illustrated in Figure 26. An advantage of this hierarchy is that in some cases the
switching of ATM cells may be based on the virtual path identifier alone.
The payload type field Fig (4.19b) identifies the type of cell. The cell loss
priority (CLP) field is a single bit; if the bit is 0 that cell has a high priority, and if the
bit is 1 the cell has a low priority. This information may influence the decision
whether to discard cells if a network becomes congested.
The header error control field contains a cyclic redundancy check on the
other bytes in the header.

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 UNIT - V
Syllabus:

Repeaters – Bridges – Routers – Gateway – Routing algorithms – TCP/IP network,

Transport and Application layers of TCP/IP – World Wide Web.

INTRODUCTION

CONNECTING DEVICES

We divide connecting devices into five different categories based on the layer
in which they operate in a network, as shown in Fig (5.1).

Fig (5.1) Five categories of connecting devices

The five categories contain devices which can be defined as

1. Those which operate below the physical layer such as a passive hub.
2. Those which operate at the physical layer (a repeater or an active hub).
3. Those which operate at the physical and data link layers (a bridge or a two-layer
switch).
4. Those which operate at the physical, data link, and network layers (a router or a
three-layer switch).
5. Those which can operate at all five layers (a gateway).

Passive Hubs
A passive hub is just a connector. It connects the wires coming from different
branches. In a star-topology Ethernet LAN, a passive hub is just a point where the
signals coming from different stations collide the hub is the collision point. This type
of a hub is part of the media; its location in the Internet model is below the physical
layer.

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 Repeaters

The following are the applications of Repeaters:

1. In digital communication systems:
A repeater is a device that receives a digital signal on an electromagnetic or
optical transmission medium and regenerates the signal along the next leg of the
medium. In electromagnetic media, repeaters overcome the attenuation caused by free-
space electromagnetic-field divergence or cable loss. A series of repeaters make
possible the extension of a signal over a distance.

2. In wireless communications system:

A repeater consists of a radio receiver, an amplifier, a transmitter, an

isolator, and two antennas. The transmitter produces a signal on a frequency that
differs from the received signal. This so-called frequency offset is necessary to
prevent the strong transmitted signal from disabling the receiver. The isolator provides
additional protection in this respect. A repeater, when strategically located on top of a
high building or a mountain, can greatly enhance the performance of a wireless
network by allowing communications over distances much greater than would be
possible without it.

3. In satellite wireless communication:

A repeater (more frequently called a transponder) receives uplink signals and

retransmits them, often on different frequencies, to destination locations.

4. In cellular telephone system:

A repeater is one of a group of transceivers in a geographic area that

collectively serve a system user.

5. In fiber optic network:

A repeater consists of a photocell, an amplifier, and a light-emitting diode

(LED) or infrared-emitting diode (IRED) for each light or IR signal that requires
amplification. Fiber optic repeaters operate at power levels much lower than wireless
repeaters, and are also much simpler and cheaper. However, their design requires
careful attention to ensure that internal circuit noise is minimized.

6. Repeaters are commonly used by commercial and amateur radio operators to

extend signals in the radio frequency range from one receiver to another. These

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 consist of drop repeaters, similar to the cells in cellular radio, and hub repeaters,
which receive and retransmit signals from and to a number of directions.
7. A bus repeater links one computer bus to a bus in another computer chassis,
essentially chaining one computer to another.
8. Network repeaters regenerate incoming electrical, wireless or optical signals.
With physical media like Ethernet or Wi-Fi, data transmissions can only span a
limited distance before the quality of the signal degrades. Repeaters attempt to
preserve signal integrity and extend the distance over which data can safely travel.

Actual network devices that serve as repeaters usually have some other name.
Active hubs, for example, are repeaters. Active hubs are sometimes also called
"multiport repeaters," but more commonly they are just "hubs." Other types of
"passive hubs" are not repeaters. In Wi-Fi, access points function as repeaters only
when operating in so-called "repeater mode."

Higher-level devices in the OSI model like switches and routers generally do
not incorporate the functions of a repeater. All repeaters are technically OSI physical
layer devices. A repeater is an electronic device that receives a signal and retransmits
it at a higher level and/or higher power, or onto the other side of an obstruction, so that
the signal can cover longer distances.

In telecommunication, the term repeater has the following standardized meanings:

1. An analog device that amplifies an input signal regardless of its nature (analog
or digital).
2. A digital device that amplifies, reshapes, retimes, or performs a combination of
any of these functions on a digital input signal for retransmission.
Because repeaters work with the actual physical signal, and do not attempt to
interpret the data being transmitted, they operate on the Physical layer, the first layer of
the OSI model.
A repeater is a device that operates only in the physical layer. Signals that
carry information within a network can travel a fixed distance before attenuation
endangers the integrity of the data. A repeater receives a signal and, before it becomes
too weak or corrupted, regenerates the original bit pattern. The repeater then sends the
refreshed signal. A repeater can extend the physical length of a LAN, as shown in Fig
(5.2).

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 Fig (5.2) A repeater connecting two segments of a LAN

A repeater does not actually connect two LANs; it connects two segments of
the same LAN. The segments connected are still part of one single LAN. A repeater is
not a device that can connect two LANs of different protocols.

A repeater connects segments of a LAN.

A repeater can overcome the 10Base5 Ethernet length restriction. In this

standard, the length of the cable is limited to 500m. To extend this length, we divide
the cable into segments and install repeaters between segments. Note that the whole
network is still considered one LAN, but the portions of the network separated by
repeaters are called segments. The repeater acts as a two-port node, but operates only
in the physical layer. When it receives a frame from any of the ports, it regenerates and
forwards it to the other port.

A repeater forwards every frame; it has no filtering capability.

It is tempting to compare a repeater to an amplifier, but the comparison is

inaccurate. An amplifier cannot discriminate between the intended signal and noise; it
amplifies equally everything fed into it. A repeater does not amplify the signal; it
regenerates the signal. When it receives a weakened or corrupted signal, it creates a
copy, bit for bit, at the original strength.

A repeater is a regenerator, not an amplifier.

The location of a repeater on a link is vital. A repeater must be placed so that a

signal reaches it before any noise changes the meaning of any of its bits. A little noise
can alter the precision of a bit‘s voltage without destroying its identity (see Fig ()). If
the corrupted bit travels much farther, however, accumulated noise can change its
meaning completely. At that point, the original voltage is not recoverable, and the error
needs to be corrected. A repeater placed on the line before the legibility of the signal

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 becomes lost can still read the signal well enough to determine the intended voltages
and replicate them in their original form.

Usage

Repeaters are often used in trans-continental and submarine

communications cables, because the attenuation (signal loss) over such distances
would be unacceptable without them. Repeaters are used in both copper-wire cables
carrying electrical signals, and in fiber optics carrying light.

Repeaters are used in radio communication services. Radio repeaters often

transmit and receive on different frequencies. A special subgroup of those repeaters is
those used in amateur radio.

Repeaters are also used extensively in broadcasting, where they are known as
translators, boosters or TV relay transmitters.
Disadvantages
However, repeaters have certain disadvantages. A repeater does not filter out
any data that passes through it. It cannot differentiate between valid signals and
noise, and amplifies both. This increases the network traffic. Therefore, if you use
repeaters to connect multiple LANs, there will be performance problems due to heavy
network traffic.

Bridges

A bridge is a device used to connect different LAN segments. These LAN segments
may use same or different data link layer protocols. A bridge operates at the data link
layer level. Bridges are also known as level- 2 devices.

Need for Bridges

Bridges are used for traffic isolation, that is, to isolate one part of a network from
another part and to reduce traffic between two segments.

Bridges can be used to extend a network over a wide area, and to interconnect 5 LANs.
For a large organization, it is possible that the network is dispersed over a wide
geographical area. Bridges use protocols, such as HDLC to send frames across point-
to-point links. Bridges use X.25 to send frames across public networks.

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 Functions of Bridges:
A bridge translates a data link layer frame of one protocol into another. It does not
operate on the network layer packet header; therefore, a bridge can forward packets of
different network layer protocols. In other words, a bridge is network layer protocol
independent. Bridges enable packet forwarding between both homogenous and
heterogeneous networks. For example, a bridge may forward TCP/IP packets as well as
SPX/IPX packets.

The functions of bridges include:

1. Forwarding the frames:

Based on the destination address, frames from one segment of LAN are forwarded to
other segments. Bridges perform this operation of forwarding appropriate frames. To
forward frames to correct LAN segments, bridges must know the LAN segment to
which the destination computer is connected.
2. Building address tables:
Bridges require address mapping to forward a packet to the correct LAN segment,
therefore, they need to build address tables. Address table maps the LAN segments
to the Ethernet addresses of the NICs.

3. Breaking collision domains:

Bridges can be used to divide large collision domains into small ones to reduce the
chances of data collisions. This helps increase the performance of networks.

Advantages and Disadvantages of Bridges:

Disadvantages:

 A bridge is more complex than a repeater. Bridges require both hardware and
software to function.

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 Advantages:

 They break up a given collision domain into multiple collision domains,

resulting in better performance.
 They can interconnect LANs of the same or different types.
 They can connect LANs to WANs.
 However, bridges introduce latency in networks. Latency is the time taken by a
bridge to receive the signal on one port, and then transmit it. It occurs because
of the inspection of the layer 2 headers.

Types of Bridges:

1. Homogeneous bridges
2. Heterogeneous bridges

Homogeneous Bridges:

Homogeneous Bridges connect two or more LAN segments based on the same
standard. For example, a bridge connecting two Ethernet or Token Ring LANs is a
homogeneous bridge.

Fig 5.3 shows a homogeneous bridge connecting two Token Ring – based LANs.

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 Fig (5.3) Homogeneous Bridges

Heterogeneous Bridges:

Fig (5.4) Heterogeneous Bridges

Fig (5.4) shows a heterogeneous bridge. This bridge connects an IEEE 802.3
Ethernet based LAN to an IEEE 802.5 Token Ring – based LAN.
Heterogeneous Bridges connect two or more LAN segments based on different
LAN standards. For example, a heterogeneous bridge may connect an Ethernet LAN to
a token Ring LAN.
Another way of classifying bridges is based on the algorithm that they use. They are:

 Transparent
 Source – routing
 Translational
 Source-route transparent.

Routers

A router is a purposely customized computer used to forward data among

computer networks beyond directly connected devices. (The directly connected
devices are said to be in LAN, where data are forwarded using Network switches.)

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 Fig (5.5) Routers

More technically, a router is a networking device whose software and

hardware [in combination] are customized to the tasks of routing and forwarding
information. A router differs from an ordinary computer in that it needs special
hardware, called interface cards, to connect to remote devices through either copper
cables or Optical fiber cable. These interface cards are in fact small computers that
are specialized to convert electric signals from one form to another. In the case of
optical fiber, the interface cards (also called ports) convert between optical signals
and electrical signals.
Routers connect two or more logical subnets, which do not share a common
network address. The subnets in the router do not necessarily map one-to-one to the
physical interfaces of the router. The term "layer 3 switching" is used often
interchangeably with the term "routing". The term switching is generally used to refer
to data forwarding between two network devices that share a common network
address. This is also called layer 2 switching or LAN switching.
Routers are physical devices that run the designated routing protocol and
perform the routing job for the network. A router can be an independent device
running only the specialized routing protocol or it can be a software process running
on a multipurpose computer. Generally, the latter approach is preferred.
Broadcast traffic and multicast are the special categories of traffic that are
given special attention during the routing operation. This is because of the generally
higher load over a network due to such traffic and their multiple destination behavior.

Broadcast and Multicast

Unicast:

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 Broadcast and multicast transport services are from one host to another host. It
means that a packet carries a single source address as well as single destination
address. Their type of transfer is also called unicast communication or transfer date.

Broadcast:

However, at times, a single packet is intended for more than one receiver.
This happens in two types of scenarios. The first is called a broadcast transfer in
which a single sender sends data to all the hosts available on the network. An
example of such communication is the broadcasting of messages by a server to all its
clients before it goes down.

The second possible scenario is when a sender wants to send data to some
specific destinations over the network. This is further complicated situation because
not all the hosts on the network destinations. Therefore, there must be some means to
identify the intended hosts and leave the others out.

Broadcasts can be of two types. The first is called a network broadcast and
includes all the hosts present on the local network or subnet. The second type of
broadcast, on the other hand, is called a universal broadcast and involves all the
hosts present on all the networks connected to the local network or subnet.

In such cases where there is more than one destination, two implementations
are possible. The first and simpler method is to send the packet to each receiver
individually. This involves one packet for each receiver, which puts a lot of traffic
over the network whenever any such multi destination communication is required.
However, it eliminates the need for any special addressing technique. The second
method, on the other hand, involves developing a special addressing technique that
can be used to identify a specific host or all the hosts on the network at time.

Multicast:

Clearly, the second approach makes more efficient use of network resources
because only a single copy of a datagram is routed over the network, which reaches
all the destinations. However, considerable support from the network layer is required

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 to implement such a network layer protocol that is aware of multicast and broadcast
communication.

A routing table is used for finding the IP address to which a packet should be
forwarded so that it may reach its destination. The routing table, as shown in figure 6.2,
has the following entries:

 Network destination: The IP address refers to the IP address of the

destination to which the packet has to reach.
 Netmask: This is the subnet mask for the network.
 Gateway: Gateway specifies the IP address to which a given packet should
be sent.

Types of routers

A demonstration of a router forwarding information to many clients.

Routers may provide connectivity

 inside enterprises,
 between enterprises and the Internet, and
 inside Internet Service Providers (ISPs).

The largest routers (for example the Cisco CRS-1 or Juniper T1600)
interconnect ISPs, are used inside ISPs, or may be used in very large enterprise
networks.

The smallest routers provide connectivity for small and home offices.

Routers for Internet connectivity and internal use

Routers intended for ISP and major enterprise connectivity almost invariably
exchange routing information using the Border Gateway Protocol (BGP).

RFC 4098 defines several types of BGP-speaking routers according to the

routers' functions:

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  Edge Router:
An ER is placed at the edge of an ISP network. The router speaks
external BGP (EBGP) to a BGP speaker in another provider or large enterprise
Autonomous System(AS). This type of routers is also called PE (Provider
Edge) routers.
 Subscriber Edge Router:
An SER is located at the edge of the subscriber's network, it speaks
EBGP to its provider's AS(s). It belongs to an end user (enterprise)
organization. This type of routers is also called CE (Customer Edge) routers.
 Inter-provider Border Router:
Interconnecting ISPs, this is a BGP speaking router that maintains BGP
sessions with other BGP speaking routers in other providers' ASes.

 Core router:

A Core router is one that resides within an AS as back bone to carry

traffic between edge routers.

 Within an ISP:
Internal to the provider's AS, such a router speaks internal BGP (IBGP)
to that provider's edge routers, other intra-provider core routers, or the
provider's inter-provider border routers.

Routers are also used for port forwarding for private servers.

Routers use headers and forwarding tables to determine the best path for
forwarding the packets, and they use protocols such as ICMP to communicate with
each other and configure the best route between any two hosts.

Very little filtering of data is done through routers.

Gateway

A gateway is a point of entry or exit at which a gate may be hung.

Gateway may also refer to:

Computer terminology

 Gateway (telecommunications), a computer or a network that allows or controls

access to another computer or network.

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  Gateway (web page), a webpage designed to attract visitors and search engines
to a particular website.
 Payment gateway, the software interface between a web-based shopping cart
and a merchant account.
 Gateway (computer program), a link between two computer programs allowing
them to share information and bypass certain protocols on a host computer.
 Residential gateway, a home networking device.

In telecommunications, the term gateway has the following meaning:

 In a communications network, a network node equipped for interfacing with

another network that uses different protocols.

 A gateway may contain devices such as protocol translators,

impedance matching devices, rate converters, fault isolators, or
signal translators as necessary to provide system interoperability. It
also requires the establishment of mutually acceptable administrative
procedures between both networks.
 A protocol translation/mapping gateway interconnects networks with
different network protocol technologies by performing the required
protocol conversions.

 Loosely, a computer configured to perform the tasks of a gateway. For a

specific case, see default gateway.

Routers exemplify special cases of gateways.

 Gateways, also called protocol converters, can operate at any layer of the OSI
model. The job of a gateway is much more complex than that of a router or
switch. Typically, a gateway must convert one protocol stack into another.

Gateways work on all seven layers of OSI architecture. The main job of a
gateway is to convert protocols among communications networks. A router by itself
transfers, accepts and relays packets only across networks using similar protocols.
A gateway on the other hand can accept a packet formatted for one protocol (e.g.
AppleTalk) and convert it to a packet formatted for another protocol (e.g.
TCP/IP) before forwarding it.

A gateway can be implemented in hardware, software or both, but they are

usually implemented by software installed within a router. A gateway must
understand the protocols used by each network linked into the router. Gateways
are slower than bridges, switches and (non-gateway) routers.

A gateway is a network point that acts as an entrance to another network.

On the Internet, a node or stopping point can be either a gateway node or a host (end-
point) node. Both the computers of Internet users and the computers that serve pages to

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 users are host nodes, while the nodes that connect the networks in between are
gateways. For example, the computers that control traffic between company networks
or the computers used by internet service providers (ISPs) to connect users to the
internet are gateway nodes.

In the network for an enterprise, a computer server acting as a gateway

node is often also acting as a proxy server and a firewall server. A gateway is often
associated with both a router, which knows where to direct a given packet of data
that arrives at the gateway, and a switch, which furnishes the actual path in and out
of the gateway for a given packet.

On an IP network, clients should automatically send IP packets with a

destination outside a given subnet mask to a network gateway. A subnet mask defines
the IP range of a network. For example, if a network has a base IP address of
192.168.0.0 and has a subnet mask of 255.255.255.0, then any data going to an IP
address outside of 192.168.0.X will be sent to that network's gateway. While
forwarding an IP packet to another network, the gateway might or might not perform
Network Address Translation.

A gateway is an essential feature of most routers, although other devices (such

as any PC or server) can function as a gateway.

Most computer operating systems use the terms described above. A computer
running Microsoft Windows however describes this standard networking feature as
Internet Connection Sharing; which will act as a gateway, offering a connection
between the Internet and an internal network. Such a system might also act as a DHCP
server. Dynamic Host Configuration Protocol (DHCP) is a protocol used by
networked devices (clients) to obtain various parameters necessary for the clients
to operate in an Internet Protocol (IP) network. By using this protocol, system
administration workload greatly decreases, and devices can be added to the network
with minimal or no manual configurations.

Internet-to-Orbit Gateway

An Internet to orbit gateway (I2O) is a machine that acts as a connector

between computers or devices connected to the Internet and computer systems
orbiting the earth, like satellites or even manned spacecrafts. Such connection is
made when the I2O establishes a stable link between the spacecraft and a computer or a
network of computers on the Internet, such link can be control signals, audio
frequency, or even visible spectrum signals.

Examples

 A very popular example is connecting a Local Area Network or Wireless LAN

to the Internet or other Wide Area Network. In this case the gateway connects a

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 LAN to the provider-specific network which in turn connects to the Internet. In
the case of a home, this gateway is called a residential gateway.

Routing algorithms

Routing:

To transfer packets from a sender to the destination, the network layer must
determine the path or the route that each packet must follow. The network layer must
determine the path for a packet, either throughout the transmission or at the start of
transfer. This job is done by the network layer routing protocol. The path information
is stored in a routing table.

Routing is a method of path selection for packets. Addresses are assigned to

the packets to facilitate successful data delivery. Addresses convey at least partial
information to the routers about the location of the host. This permits routers to
forward packets towards their destination.

Types of routing
1. Static routing
2. Dynamic routing
Static routing:
This is the simplest type of routing, in which the routing information is entered
once and it does not usually change. The network administrator enters the routing
information into the routing tables.

Dynamic routing:
In dynamic routing, routers build their routing tables dynamically. Routes are
added or deleted on the basis of the information exchanged among various routers.

For large and rapidly changing networks, dynamic routing is the most useful. It
reflects the state of a network more accurately than static routing.

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 Routing Algorithms:

Any routing protocol is based upon an algorithm that finds out the path for a
packet. The purpose of a routing algorithm is to find out a path from source to the
destination for a given set of routers.

In a given graph of routers, the problem of finding the route from a source to a
destination reduces to identifying a series of consecutive links in which:
 The source is connected to the first link in the path
 The destination is connected to the last link in the path.

Some of the important routing algorithms include:

 Dijkstra‘s algorithm
 Shortest path
 Flooding
 Distance vector

1. Dijkstra’s Algorithm

E.W.Dijkstra discovered an algorithm to find the shortest path between two

nodes in a graph. Many routing algorithms use Dijkstra‘s algorithm to find the shortest
path between two routers on the network.

Dijkstra‘s algorithm assumes that the cost is a positive number between any
pair of routers, the nodes in the graph. The network with the routers is considered as a
graph represented by the nodes on the network. The algorithm starts from the source
node and proceeds with finding out the routes for all the other nodes. The first routes
found are the one-hop routes, which are from the source node to the nodes directly
connected to this source. Then, the algorithm proceeds to the next node. All the direct
routes are calculated and the costs of all the nodes from the original node are calculated
again. The procedure is repeated until the routes for all the nodes are found. Fig 5.6
shows a graph with six nodes, the cost of each link being equal to one.

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 Fig (5.6)

In fig 5.6, A is the source node. The nodes directly connected to A are B and C,
and the cost of A-B and A-C is equal to 1. The nodes directly connected to B are A,C
and E. From node B, the cost of B-A, B-C and B-E are all 1. however, when you
calculate the cost of reaching C through B from A (A-C-B) it equals 2.

In a similar manner, each node is traversed and the cost of reaching all the
nodes from the source node is recalculated. Only the minimum cost of reaching a node
from the source node is taken into consideration. In this way, the costs of reaching all
the nodes on the network can be found out.
Table 5.1 lists all the source destination pairs with the source as node A and the
corresponding costs of reaching the destination node.

Source Node - Destination Node Cost

A-B 1

A-C 1

A-D 2

A-E 2

A-F 2

2. Shortest Path

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 The shortest path algorithm assumes that the router knows the complete
topology of the network. Therefore, it looks at computing the optimal path with the
least cost for the given source and the destination. It uses Dijkstra‘s algorithm for this
purpose. The shortest path algorithm guarantees convergence to the shortest path
available on the known network. The problem with the approach is that it requires prior
information about ht entire network topology, which becomes very difficult in real
scenarios, where frequent changes are also possible on the underlying network.

3. Flooding
Flooding also guarantees the shortest path delivery of packets to their
destination. As the name suggests, every router that receives a packet during transition
sends the packet to its entire connected links. This technique is call flooding. As the
packet passes through all the possible paths on the network, it is guaranteed to pass
through the optimal path also.
This approach does not require any prior information regarding network
topology. However, the main problem with this approach is that duplicate copies reach
the destination and, therefore, the destination host has to make some arrangements to
manage these duplicate packets. In addition, if no restriction is put on the outgoing
links at the routers, these duplicate packets travel on the network for a long time and
cause unnecessary load on the network. To tackle this problem, routers check the
packets before flooding them out. The checking process involves the following
considerations:

 The checking process must send packets out over the links excluding the incoming
link.
 If the packet has been flooded in the past, it should not be flooded again.
 The packet should be flooded only over the links that probably go towards the
destination, instead of all the links.
Flooding guarantees the quickest possible delivery of the packet. It is used in
critical applications such as military networks where parallel paths are created on the
network. In such networks, flooding ensures maximum connectivity even if some links
are broken.

4. Distance-Vector Algorithm

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 The distance-vector algorithm is an iterative, asynchronous, and disturbed
routing algorithm. Each router receives some information from one or more of its
directly attached neighbors, perform a calculation, and them distributes the results of
its calculation back to its neighbors. This process is iteratively continued until no more
information is exchanged between neighbors. The algorithm is asynchronous because it
does not require all the nodes to operate in lock step with each other.
The principal data structure in the algorithm is the distance table maintained on
each node. The distance table for a node is a table that has a row for each known
destination on the network and a column indicating the neighbor node having the least
cost for that destination.

The neighbor-to-neighbor communication takes place in such a way that each

node comes to know the cost of each of its neighbor‘s minimum cost path to each
destination. Similarly, whenever a node computers a new minimum cost to some
destination, it informs its neighbors of this new minimum cost. The associated numeric
value in the distance table gives the cost of the minimum cost path to the corresponding
destination.

Fig (5.7)
Routing Table
A host or a router has a routing table with an entry for each destination, or a
combination of destinations, to route IP packets. The routing table can be either static
or dynamic.
Static Routing Table

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 A static routing table contains information entered manually. The administrator
enters the route for each destination into the table. When a table is created, it cannot
update automatically when there is a change in the Internet. The table must be
manually altered by the administrator.

A static routing table can be used in a small internet that does not change very
often. or in an experimental internet for troubleshooting. It is poor strategy to use a
static routing table in a big Internet such as the Internet.

Dynamic Routing Table

A dynamic routing table is updated periodically by using one of the dynamic
routing protocols such as RIP, OSPF, or BGP. Whenever there is a change in the
Internet, such as a shutdown of a router or breaking of a link, the dynamic routing
protocols update all the tables in the routers (and eventually in the host) automatically.
The routers in a big internet such as the Internet need to be updated
dynamically for efficient delivery of the IP packets.

TCP/IP Network

The TCP/IP model, or Internet Protocol Suite, describes a set of general design
guidelines and implementations of specific networking protocols to enable
computers to communicate over a network. TCP/IP provides end-to-end
connectivity specifying how data should be formatted, addressed, transmitted,
routed and received at the destination. Protocols exist for a variety of different types
of communication services between computers.

TCP/IP is generally described as having four abstraction layers (RFC 1122). This
layer architecture is often compared with the seven-layer OSI Reference Model; using
terms such as Internet Reference Model in analogy is however incorrect as the Internet
Model is descriptive while the OSI Reference Model was intended to be prescriptive,
hence Reference Model.

TCP/IP Protocol Architecture

TCP/IP protocols map to a four-layer conceptual model known as the DARPA

model , named after the U.S. government agency that initially developed TCP/IP. The

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 four layers of the DARPA model are: Application, Transport, Internet, and Network
Interface.

Figure 5.8 shows the TCP/IP protocol architecture.

Fig (5. 8) TCP/IP Protocol Architecture

1. Network Interface Layer

The Network Interface layer (also called the Network Access layer) is
responsible for placing TCP/IP packets on the network medium and receiving TCP/IP
packets off the network medium. TCP/IP was designed to be independent of the
network access method, frame format, and medium. In this way, TCP/IP can be used to
connect differing network types. These include LAN technologies such as Ethernet and
Token Ring and WAN technologies such as X.25 and Frame Relay. Independence
from any specific network technology gives TCP/IP the ability to be adapted to new
technologies such as Asynchronous Transfer Mode (ATM).
The Network Interface layer encompasses the Data Link and Physical layers of
the OSI model. Note that the Internet layer does not take advantage of sequencing and
acknowledgment services that might be present in the Data-Link layer. An unreliable
Network Interface layer is assumed, and reliable communications through session
establishment and the sequencing and acknowledgment of packets is the responsibility
of the Transport layer.

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 2. Internet Layer
The Internet layer is responsible for addressing, packaging, and routing
functions. The core protocols of the Internet layer are IP, ARP, ICMP, and IGMP.

 The Internet Protocol (IP) is a routable protocol responsible for IP addressing,

routing, and the fragmentation and reassembly of packets.
 The Address Resolution Protocol (ARP) is responsible for the resolution of the
Internet layer address to the Network Interface layer address such as a hardware
address.
 The Internet Control Message Protocol (ICMP) is responsible for providing
diagnostic functions and reporting errors due to the unsuccessful delivery of IP
packets.
 The Internet Group Management Protocol (IGMP) is responsible for the
management of IP multicast groups.

The Internet layer is analogous to the Network layer of the OSI model.

3. Transport Layer
The Transport layer (also known as the Host-to-Host Transport layer) is
responsible for providing the Application layer with session and datagram
communication services. The core protocols of the Transport layer are Transmission
Control Protocol (TCP) and the User Datagram Protocol (UDP).
 TCP provides a one-to-one, connection-oriented, reliable communications service.
TCP is responsible for the establishment of a TCP connection, the sequencing and
acknowledgment of packets sent, and the recovery of packets lost during
transmission.
 UDP provides a one-to-one or one-to-many, connectionless, unreliable
communications service. UDP is used when the amount of data to be transferred is
small (such as the data that would fit into a single packet), when the overhead of
establishing a TCP connection is not desired or when the applications or upper
layer protocols provide reliable delivery.
The Transport layer encompasses the responsibilities of the OSI Transport layer
and some of the responsibilities of the OSI Session layer.
4. Application Layer
The Application layer provides applications the ability to access the services of
the other layers and defines the protocols that applications use to exchange data. There
are many Application layer protocols and new protocols are always being developed.

The most widely-known Application layer protocols are those used for the
exchange of user information:

 The Hypertext Transfer Protocol (HTTP) is used to transfer files that make up
the Web pages of the World Wide Web.
 The File Transfer Protocol (FTP) is used for interactive file transfer.

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  The Simple Mail Transfer Protocol (SMTP) is used for the transfer of mail
messages and attachments.
 Telnet, a terminal emulation protocol, is used for logging on remotely to network
hosts.

Additionally, the following Application layers protocols help facilitate the use
and management of TCP/IP networks:

 The Domain Name System (DNS) is used to resolve a host name to an IP address.
 The Routing Information Protocol (RIP) is a routing protocol that routers use to
exchange routing information on an IP internetwork.
 The Simple Network Management Protocol (SNMP) is used between a network
management console and network devices (routers, bridges, intelligent hubs) to
collect and exchange network management information.

Examples of Application layer interfaces for TCP/IP applications are Windows

Sockets and NetBIOS.

 Windows Sockets provides a standard application programming interface (API)

under Windows 2000.
 NetBIOS is an industry standard interface for accessing protocol services such as
sessions, datagrams, and name resolution.

Transport and Application Layers of TCP/IP

Transport Layer

The Transport Layer's responsibilities include end-to-end message transfer

capabilities independent of the underlying network, along with error control,
segmentation, flow control, congestion control, and application addressing (port
numbers). End to end message transmission or connecting applications at the transport
layer can be categorized as either connection-oriented, implemented in Transmission
Control Protocol (TCP), or connectionless, implemented in User Datagram Protocol
(UDP).

The Transport Layer can be thought of as a transport mechanism, e.g., a vehicle

with the responsibility to make sure that its contents (passengers/goods) reach their
destination safely and soundly, unless another protocol layer is responsible for safe
delivery.

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 The Transport Layer provides this service of connecting applications through
the use of service ports. Since IP provides only a best effort delivery, the Transport
Layer is the first layer of the TCP/IP stack to offer reliability. IP can run over a reliable
data link protocol such as the High-Level Data Link Control (HDLC). Protocols above
transport, such as RPC, also can provide reliability.
For example, the Transmission Control Protocol (TCP) is a connection-oriented
protocol that addresses numerous reliability issues to provide a reliable byte stream:
 data arrives in-order
 data has minimal error (i.e. correctness)
 duplicate data is discarded
 lost/discarded packets are resent
 includes traffic congestion control

The newer Stream Control Transmission Protocol (SCTP) is also a reliable,

connection-oriented transport mechanism. It is Message-stream-oriented — not byte-
stream-oriented like TCP — and provides multiple streams multiplexed over a single
connection. It also provides multi-homing support, in which a connection end can be
represented by multiple IP addresses (representing multiple physical interfaces), such
that if one fails, the connection is not interrupted.
User Datagram Protocol is a connectionless datagram protocol. Like IP, it is a
best effort, "unreliable" protocol. Reliability is addressed through error detection using
a weak checksum algorithm. UDP is typically used for applications such as streaming
media (audio, video, Voice over IP etc) where on-time arrival is more important than
reliability, or for simple query/response applications like DNS lookups, where the
overhead of setting up a reliable connection is disproportionately large. Real-time
Transport Protocol (RTP) is a datagram protocol that is designed for real-time data
such as streaming audio and video.
TCP and UDP are used to carry an assortment of higher-level applications. The
appropriate transport protocol is chosen based on the higher-layer protocol application.
For example, the File Transfer Protocol expects a reliable connection, but the Network
File System (NFS) assumes that the subordinate Remote Procedure Call protocol, not
transport, will guarantee reliable transfer. Other applications, such as VoIP, can tolerate
some loss of packets, but not the reordering or delay that could be caused by
retransmission.
The applications at any given network address are distinguished by their TCP
or UDP port. By convention certain well known ports are associated with specific
applications.

Application Layer
The Application Layer refers to the higher-level protocols used by most
applications for network communication. Examples of application layer protocols
include the File Transfer Protocol (FTP) and the Simple Mail Transfer Protocol
(SMTP)[10]. Data coded according to application layer protocols are then encapsulated
into one or (occasionally) more transport layer protocols (such as the Transmission

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 Control Protocol (TCP) or User Datagram Protocol (UDP)), which in turn use lower
layer protocols to effect actual data transfer.
Since the IP stack defines no layers between the application and transport
layers, the application layer must include any protocols that act like the OSI's
presentation and session layer protocols. This is usually done through libraries.
Application Layer protocols generally treat the transport layer (and lower)
protocols as "black boxes" which provide a stable network connection across which to
communicate, although the applications are usually aware of key qualities of the
transport layer connection such as the end point IP addresses and port numbers. As
noted above, layers are not necessarily clearly defined in the Internet protocol suite.
Application layer protocols are most often associated with client-server applications,
and the commoner servers have specific ports assigned to them by the IANA: HTTP
has port 80; Telnet has port 23; etc. Clients, on the other hand, tend to use ephemeral
ports, i.e. port numbers assigned at random from a range set aside for the purpose.
Transport and lower level layers are largely unconcerned with the specifics of
application layer protocols. Routers and switches do not typically "look inside" the
encapsulated traffic to see what kind of application protocol it represents, rather they
just provide a conduit for it. However, some firewall and bandwidth throttling
applications do try to determine what's inside, as with the Resource Reservation
Protocol (RSVP). It's also sometimes necessary for Network Address Translation
(NAT) facilities to take account of the needs of particular application layer protocols.
(NAT allows hosts on private networks to communicate with the outside world via a
single visible IP address using port forwarding, and is an almost ubiquitous feature of
modern domestic broadband routers).

World Wide Web

Architecture
The WWW today is a distributed client server service, in which a client using a
browser can access a service using a server. However, the service provided is
distributed over many locations called sites, as shown in Fig (5.9).

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 Fig (5.9)

One of the most popular Internet services is electronic mail (e-mail). The
designers of the Internet probably never imagined the popularity of this application
program. Its architecture consists of several components.

At the beginning of the Internet era, the messages sent by electronic mail were
short and consisted of text only; they let people exchange quick memos. Today,
electronic mail is much more complex. It allows a message to include text, audio, and
video. It also allows one message to be sent to one or more recipients.
The World Wide Web (WWW) is a repository of information linked together
from points all over the world. The WWW has a unique combination of flexibility,
portability, and user-friendly features that distinguish it from other services provided
by the Internet. The WWW project was initiated by CERN (European Laboratory for
Particle Physics) to create a system to handle distributed resources necessary for
scientific research. In this chapter we first discuss issues related to the Web.
Each site holds one or more documents, referred to as Web pages. Each Web
page can contain a link to other pages in the same site or at other sites. The pages can
be retrieved and viewed by using browsers. Let us go through the scenario shown in
Fig (9). The client needs to see some information that it knows belongs to site A. It
sends a request through its browser, a program that is designed to fetch Web
documents. The request. among other information, includes the address of the site and
the Web page, called the URL, which we will discuss shortly. The server at site A finds
the document and sends it to the client. When the user views the document, she finds
some references to other documents, including a Web page at site B. The reference has
the URL for the new site. The user is also interested in seeing this document. The client
sends another request to the new site, and the new page is retrieved.
(Client) Browser
A variety of vendors offer commercial browsers that interpret and display a
Web document, and all use nearly the same architecture.
Each browser usually consists of three parts:
i. a controller,

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 ii.client protocol, and
iii.interpreters
The controller receives input from the keyboard or the mouse and uses the
client programs to access the document. After the document has been accessed, the
controller uses one of the interpreters to display the document on the screen.
The client protocol can be one of the protocols such as FTP and HTTP.
The interpreter can be HTML, Java, or JavaScript, depending on the type of
document.
We discuss the use of these interpreters based on the document type later in the
chapter (see Figure (5.10)).

Fig (5.10) URL

Server
The Web page is stored at the server. Each time a client request arrives, the
corresponding document is sent to the client. To improve efficiency, servers normally
store requested files in a cache in memory; memory is faster to access than disk. A
server can also become more efficient through multithreading or multiprocessing. In
this case, a server can answer more than one request at a time.
Uniform Resource Locator
A client that wants to access a Web page needs the address. To facilitate the
access of documents distributed throughout the world, HTTP uses locators. The
uniform resource locator (URL) is a standard for specifying any kind of information on
the Internet. The URL defines four things: protocol, host computer, port, and path (see
Fig 5.11).

Fig (5.11)
Cookies
The World Wide Web was originally designed as a stateless entity. A client
sends a request; a server responds. Their relationship is over. The original design of

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 WWW, retrieving publicly available documents, exactly fits this purpose. Today the
Web has other functions; some are listed here.
1. Some websites need to allow access to registered clients only.
2.. Websites are being used as electronic stores that allow users to browse through the
store, select wanted items, put them in an electronic cart, and pay at the end with a
credit card.
3. Some websites are used as portals: the user selects the Web pages he wants to see.
4. Some websites are just advertising.

HTML
Hypertext Markup Language (HTML) is a language for creating Web pages.
Data for a Web page are formatted for interpretation by a browser.

Common Gate way Interface (CGI)

The Common Gateway Interface (CGI) is a technology that creates and handles
dynamic documents. CGI is a set of standards that defines how a dynamic document is
written, how data are input to the program, and how the output result is used.

HTTP
The Hypertext Transfer Protocol (H1’TP) is a protocol used mainly to
access data on the World Wide Web. HTTP functions as a combination of FTP and
SMTP. It is similar to FTP because it transfers files and uses the services of TCP.
However, it is much simpler than FTP because it uses only one TCP connection.
There is no separate control connection: only data are transferred between the client
and the server.
HTTP is like SMTP because the data transferred between the client and the
server look like SMTP messages. In addition, the format of the messages is
controlled by MIME-like headers. Unlike SMTP, the HTTP messages are not
destined to be read by humans; they are read and interpreted by the HTTP server
and HTTP client (browser). SMTP messages are stored and forwarded, but HTTP
messages are delivered immediately. The commands from the client to the server are
embedded in a request message. The contents of the requested file or other
information are embedded in a response message.
HTTP uses the services of TCP on well-known port 80. HTTP uses the
services of TCP on well-known port 80.
The Internet is a global system of interconnected computer networks that use
the standard Internet Protocol Suite (TCP/IP) to serve billions of users worldwide. It is
a network of networks that consists of millions of private, public, academic, business,
and government networks of local to global scope that are linked by a broad array of
electronic and optical networking technologies. The Internet carries a vast array of

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 information resources and services, most notably the inter-linked hypertext documents
of the World Wide Web (WWW) and the infrastructure to support electronic mail.
Most traditional communications media, such as telephone and television
services, are reshaped or redefined using the technologies of the Internet, giving rise to
services such as Voice over Internet Protocol (VoIP) and IPTV. Newspaper publishing
has been reshaped into Web sites, blogging, and web feeds. The Internet has enabled or
accelerated the creation of new forms of human interactions through instant messaging,
Internet forums, and social networking sites.
The origins of the Internet reach back to the 1960s when the United States
funded research projects of its military agencies to build robust, fault-tolerant and
distributed computer networks. This research and a period of civilian funding of a new
U.S. backbone by the National Science Foundation spawned worldwide participation in
the development of new networking technologies and led to the commercialization of
an international network in the mid 1990s, and resulted in the following popularization
of countless applications in virtually every aspect of modern human life. As of 2009,
an estimated quarter of Earth's population uses the services of the Internet.
The Internet has no centralized governance in either technological
implementation or policies for access and usage; each constituent network sets its own
standards. Only the overreaching definitions of the two principal name spaces in the
Internet, the Internet Protocol address space and the Domain Name System, are
directed by a maintainer organization, the Internet Corporation for Assigned Names
and Numbers (ICANN). The technical underpinning and standardization of the core
protocols (IPv4 and IPv6) is an activity of the Internet Engineering Task Force
(IETF), a non-profit organization of loosely affiliated international participants that
anyone may associate with by contributing technical expertise.
The first part of the address is called a protocol identifier and it indicates what
protocol to use, and the second part is called a resource name and it specifies the IP
address or the domain name where the resource is located. The protocol identifier and
the resource name are separated by a colon and two forward slashes.
For example, the two URLs below point to two different files at the domain
pcwebopedia.com. The first specifies an executable file that should be fetched using
the FTP protocol; the second specifies a Web page that should be fetched using the
HTTP protocol.

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