CHAPTER

OVERVIEW OF COMMUNICATIONS NETWORKS

1.1 WHAT IS COMMUNICATION?

Generally, communication is the techniques of exchanging thoughts, messages, or information, such as speech,
signals, writing, or behaviour. It is the field of study concerned with the transmission (including reception) of
information by various means, i.e., print, broadcasting, mail, telephone, fax or television, for sending and
receiving messages or information. In the electronic world, communication is the transfer of information from one
point to another.
1.2

WHAT ARE NETWORKS?

Historically, most communications systems were started with point-to-point links which directly connected
together the users wishing to communicate using a dedicated communications circuit. As the distance between
users increased beyond the length of the cable, the connection between the users was formed by a number of
sections which were connected end-to-end in series to form the circuit. The connection between the users (A and
D) in the figure below is represented by a series of links (AB, BC, and CD) each link connects two points known
as nodes. For a point-to-point circuit, (also known as a permanent circuit) the nodes are patch panels which
provide a simple connection between the two links (i.e. the two transmission circuits).

Fig 1.1 A connection among four nodes (A & D) formed from 3 links
As the number of connected users increased, it has become infeasible to provide a circuit which connects every
user to every other user, and some sharing of the transmission circuits (known as "switching") has become
necessary. Therefore, Networks are sets of nodes that are interconnected to permit the exchange of information.
1.3

WHAT ARE COMMUNICATIONS NETWORKS?

In Telecommunications Engineering, Communications Networks are the channels and nodes that interconnect all
transmission and switching systems including the last mile elements (Customer Assess Networks - CAN), as
well as interface devices, supporting hardware and software. Communication networks elaborate the
Fundamental Model of Communications. The model shown in diagram below describes point-to-point
communications. One modern example of this communications mode is the modem that connects a personal
computer with an information server via a telephone line. The key aspect of this model is that the channel is
dedicated: Only one communications link through the channel is allowed at a time.

1.4

FUNDAMENTAL MODEL OF A COMMUNICATION NETWORK

Regardless whether we have a wire-line or wireless channel, communication bandwidth is precious, and if it could
be shared without significant degradation in communications performance (measured by signal-to-noise ratio for
analogue signal transmission and by bit-error probability for digital transmission) so much the better.

Fig. 1.2 Fundamental Model of Communications

The prototypical communications networkbe it the Postal Service, Cellular Telephone, or the Internetconsists
of nodes interconnected by links. Messages formed by the source are transmitted within the network by dynamic
routing. Two routes are shown. The longer one would be used if the direct link were disabled or congested.
The first electrical communications network was the telegraph. Here the network consisted of telegraph operators
who transmitted the message efficiently using Morse code and routed the message so that it took the shortest
possible path to its destination while taking into account internal network failures.

Fig. 1.3 Model of Communications Transmission Network.

1.5

BASIC SIGNAL V PROCESSED SIGNAL

Frequently communications networks process basic signals in order to increase the capacity of the network. One
effective means of increasing the range or number of services that can be carried on any particular cable or
wireless system is to compress the basic signal, so that much of the redundant information is removed before
transmission. For instance, the information in the 30 frames that make up a second of video can be compressed
from about 27 million to 1 million bits without serious loss of quality.
Another way to enhance the capacity of a transmission system is to carry different types of signal or service at the
same time. This can be done by multiplexing techniques that for instance mix voice and data signals on the same
transmission path. A whole variety of signal compression and multiplexing technologies are now massively
expanding the capacity of communications networks.
1.6

STAND-ALONE NETWORKS V CONVERGED OFFERINGS

Traditionally, communications networks have been created to serve specific purposes, such as the broadcasting
network for television, the telephone network for voice, and the Internet for data. Increasingly both physical
networks and customer offerings are converging because digitalisation of networks makes this possible and
competitive marketplaces make it desirable.

At least three types of convergence are occurring: between voice and data through services like ISDN and
networks like the Internet; between telecommunications and broadcasting through telephone companies offering
video on demand and cable companies providing telephony service; and between fixed and mobile telephony as
customers are offered packages combining both services and phones have been developed that can be used in the
home or office or on the move.

CHAPTER

REVIEW OF ANALOGUE AND DIGITAL SYSTEMS

Overview: One way of understanding how telecommunications systems work and how they are changing is to
consider a number of basic paired concepts.
2.1
ANALOGUE V DIGITAL
One way of understanding how telecommunications systems work and how they are changing is to consider a
number of basic paired concepts, such as, Analogue and Digital Systems.
2.1.1

WHAT ARE ANALOGUE SYSTEMS?

Analogue Systems are electronic devices that use continuous wave-like form signals to function. This means,
Analogue signals have a continuous wave-like form. In Analogue Communications Systems, the further the
signal travels, the more likely it is to become degraded and therefore, over anything other than short distances,
these signals are boosted in strength at various points in the network and any subsequent distortions are amplified
at the same time.

Fig. 2.1 Analogue Signal Wave-form

2.1.2

DEMERITS OF ANALOGUE SYSTEMS

These include:
Poor Quality of Signal - due to High Levels of Noise, Distortion, Harmonics, Etc.), Low Capacity, Low
Speed; Low Short Distance Coverage; Etc.
2.1.3

WHAT ARE DIGITAL SYSTEMS?

By contrast, digital systems are electronic devices that utilize discrete signals (Pulses) to operate. They
essentially represent power on or off; or to use the binary language of computers Zeros (0s) or ones (1s).
1
0

Fig. 2.2 Digital Signal Wave-form

Digital signals can be recreated (regenerated) at their destination / terminating-ends in exactly the same form they
left their starting point despite any distortion, noise etc., along the route, since the only two possibilities for the
original signal are 0s and 1s, on or off.
2.1.4

WHY DIGITAL SYSTEMS?

All communications networks whether telecommunications or broadcasting, whether fixed or mobile are
increasingly becoming digital because the QUALITY of the signal is so much SUPERIOR, the CAPACITY of the
network is so much greater and the DISTANCE COVERED is so ENORMOUS / MASSIVE - due to signal
regeneration.
2.1.5

DIGITAL SYSTEM DESIGN

Digital System Design involves a significant amount of custom/conventional logic circuitry which also includes
pre-designed major components, such as processors, memory units and various types of input/output (I/O)
interfaces. Digital signals can be recreated at their destination in exactly the form they left their starting point
despite any distortion along the way, since the only two possibilities for the original signal are 0 and 1. All
communications networks whether telecommunications or broadcasting, whether fixed or mobile are
increasingly becoming digital because the quality of the signal is so much better and the capacity of the network is
so much greater.
The Need For Digital Signals
The tremendous development of computers and the various kinds of data exchanged between major of human
activity have led to the necessity to process data digitally and at high speed. This tendency has resulted in the
adoption of time division telephone switching and naturally to digitizing of the transmitted signals. This is now a
policy deliberately chosen by telecommunications administrations for technical and cost reasons, and to provide
their subscribers with new service at the same as the telephone service and over the same public network.
Digital multiplexes are developed, leading to the standardization of the orders of transmission data rates and
allowing easy interconnection of networks. All these techniques, based on pulse code modulation (P.C.M) are
contributing to the fact that telecommunications are rapidly reaching the age of telematics and videomatics. In
this general perspective, microwave digital links have on important role to play.
The beginning of modern digital transmission took place in Europe when CEPT and CCITT published the first
recommendations for pulse coded modulation (P.C.M) techniques and the associated multiplex hierarchy,
following the model existing for analog transmission. The information to be transmitted can be any of the
following:

Telephone signals, Television signals, Digital data, Telegraphic message, etc.

At the present time Telephone and Television signals represent the major part of the traffic dispatched by radio
links. These signals being fundamentally analog signals, the question arises: why using digital modulation. The
first reason is a technical reason: The advantage of digital transmission over analog transmission can be shown
reference to the signal and noise relationship with the distance.

Fig. 2.3 Signal Power vs Distance

Fig 2.4 Noise Level vs Distance

The significant difference between the two types of transmission lies in the resultant noise level. The cumulative
noise characteristic of analog transmission and the non cumulative noise characteristic of digital transmission.
The noise contributions of the analog transmission stages or repeaters are additive that the signal to noise ratio
become worse and worse with the distance and there is no known way to avoid this problem.
On the other hand, in digital transmission the information is contained in the serial combination of <<0>> and
<<1>> logic levels. Once the decision has been mode whether the received signal information is a <<0>> or
<<1>> logic level, the received signal can be regenerated without any noise contribution. Therefore in digital
transmission the noise is not additive with the number of repeaters and does not accumulate with increased
distance.

CHAPTER 3
TELEPHONY
3.1

OVERVIEW

What Is Telephony?
In the field of Telecommunications Technology, the term Telephony, embodies the general use of equipment
and devices to provide voice communication over distances, specifically by connecting telephones to each other.
Basically, Telephony is about Telephone Systems used for Telephone Conversation. Telephones originally were
connected directly together in pairs. Each user had separate telephones wired to the various places he/she might
wish to reach.
This became inconvenient when people wanted to talk to many other telephones, so the telephone exchange was
invented. Each telephone could then be connected to other local ones, thus inventing the local loop and the
telephone call. Soon, nearby exchanges were connected together by trunk lines, and eventually far long distance
calls were made.
3.1.1

Digital Telephony

The Public Switched Telephone Network (PSTN) has gradually evolved towards digital telephony which has
improved the capacity and quality of the network. End-to-end analogue telephone networks were first modified in
the early 1960s by upgrading transmission networks with T1 carrier systems. Later technologies such as SONET
and fiber optic transmission methods further advanced digital transmission. Although analogue carrier systems
existed, digital transmission made it possible to significantly increase the number of channels multiplexed on a
single transmission medium.
3.1.2

IP Telephony

IP Telephony is a modern form of telephony which uses the TCP/IP protocol popularized by the internet to
transmit digitized voice data. Contrast this with the operation of Plain Old Telephone Service (POTS). Internet
Protocol (IP) telephony (also known as Internet telephony) is a service based on Voice over IP (VoIP), a disruptive
technology that is rapidly gaining ground against traditional telephone network technologies. IP telephony uses a
broadband Internet connection and IP Phones to transmit conversations as data packets. Internet Protocol (IP)
Telephony (also known as internet telephony) is a service based on Voice over IP (VoIP), a disruptive technology
that is rapidly gaining ground against traditional telephone network technologies. IP telephony uses a broadband
internet connection to transmit conversations as data packets.
In addition to replacing POTS, IP telephony is also competing with mobile phone networks by offering free or
lower cost connections via WiFi hotspots. VoIP is also used on private wireless networks which may or may not
have a connection to the outside telephone network.
3.1.3

Telephone Local Loop

Telephone Local Loop is the portion of the telephone system that connects our home or office to the nearest
Exchange / Central office (CO) of our local Telco. A local loop is the transmission path between a central office
and a subscriber's premises. The wiring used in the local loop is usually unshielded twisted-pair (UTP) cabling.

Fig. 3.1

A Typical Local Loop

The transmission method is analogue transmission, and the maximum distance from the Exch. to the subscribers
customer premises is about 5 kilometers. A local loop may be provisioned to support data communications
applications, or combined voice and data such as digital subscriber line (DSL).
Local loop connections can be used to carry a range of services, including: Analogue voice and signalling used in
traditional POTS - Integrated Services Digital Network (ISDN) - Variants of Digital Subscriber Line (XDSL).
The diagrams and descriptions that follow illustrate how other parts of the phone system connect to that local
loop.
3.1.4

Basic Phone Loop and Its Environment

Each subscriber is connected via a 2-wire (2W) loop to a switch office which provides inter-subscriber loop
switching and signal processing (analogue and/or digital). The SLIC is the primary interface between the 4 wire
(4W) (ground referenced) low voltage switch environment and the 2W (floating) high voltage loop
environment. The loop consists of a wire A (the Tip wire), the telephone set or its equivalent, and wire B (the ring
wire). A DC voltage is applied across the Tip and Ring wires at the line card which is housed in the telephone
office. The battery is usually a nominal -48V, and is often called the quiet or talking battery. When the telephone
is off-hook, a DC path is established around the loop. DC loop current will flow around the loop from tip feed to
ring feed. This is called Battery Feed.

Fig. 3.2 Typical circuit route for a medium-distance call

A local loop may be provisioned to support data communications applications, or combined voice and data such
as Digital Subscriber Line (DSL). Local loop connections can be used to carry a range of services, including:

Analogue voice and signalling used in traditional POTS

Integrated Services Digital Network (ISDN)

Variants of Digital Subscriber Line (XDSL)

However, modern implementations may include a digital loop carrier system segment or fiber optic transmission
system known as Fibre-In-The-Loop OR Fibre-To-The Home (FTTH).
3.1.5

Data Network Local Loop

The Local Loop is often called "the last mile", and it refers to the last section of analogue phone line that goes
from the telephone company's central office (CO) to your house. Typical local loop protocols are: Voice lines;
Modem connections - 56 kbps; ISDN - 2 x 64 kbps digital lines; ADSL - up to 8 Mbps; Cable Modems - up to 30
Mbps

Fig. 3.3 Simple Data Network Local Loop

The local loops are sill analog. Consequently, when a computer wishes to send digital data over a dial-up line, the
data must first be converted to analog form by a modem for transmission over a local loop, then converted to
digital form for transmission over the long-haul trunks, then back to analog over the local loop at the receiving
end, and finally back to digital by another modem for storage in the destination computer (Fig. 3.4).

Fig.3.4 Local Loop for Data Communication Systems

The use of both analogue and digital transmission for a computer to computer call. Conversion is done by.
the modems and Codecs. For leased lines it is possible to go digital from start to finish, but these lines are still
expensive. For advanced future services, such as video on demand, the 3-kHz will not do. Two possibilities of
what to do are:

3.1.6

Running a fibre from end office into everyone's house called FTTH (Fibre To The Home). This
solution fits in well with the current system but it is too expensive.

Running an optical fibre from each end office into each neighbourhood (the curb) that it serves
(FTTC: Fibre-To-The Curb).

Fibre in the Local Loop

For advanced future services, such as video on demand, the 3-kHz channel currently used will not do. Two
possibilities of what to do are discussed:
1. Running a fibre from end office into everyone's house called FTTH (Fibre To The Home). This solution
fits in well with the current system but it is too expensive.
2. Running an optical fibre from each end office into each neighbourhood (the curb) that it serves (FTTC Fibre To The Curb). The fibre is terminated in a junction box that all the local loops enter.
3. Since the local loops are now much shorter (around 100 m), they can be run at higher speeds, around 1
Mbps fig. a). An alternative design uses existing cable TV infrastructure (Fig. b).

Fig. 3.6 Fibre to the curb. (a) Using the telephone network. (b) Using the cable TV network.
3.1.7

Local Loop Line Characteristics

Telephone lines are not perfect devices due to their analogue nature. The quality of the telephone line determines
the rate that modulated data can be transferred. Good noise free lines allow faster transfer rates such as 14.4 kbps,
poor quality lines require the data transfer rate to be stepped down to 9.6k bps or less. Phone lines have several
measurable characteristics that determine the quality of the line:
Attenuation, Distortion, Propagation Delay & Envelop Delay
a) Attenuation: Attenuation Distortion is the change in amplitude of the transmitted signal over the Voice
Band. It is the frequency response curve of the Voice Band.

10

b) Distortion: Alteration / Change in the original waveform/shape of Electrical Signals.

c) Propagation Delay: Signals transmitted down a phone line will take a finite time to reach the end of the
line. The delay from the time the signal was transmitted to the time it was received is called Propagation
Delay.
d) Envelop Delay: The Propagation Time Delay of the envelop of an Amplitude Modulated signal as it
passes through a filter. It is sometimes called Time Delay or Group Delay./
3.1.8

Local Loop Line Impairments

Line Impairments are faults in the line due to improper line terminations or equipment out of specifications.
These cannot be conditioned out but can be measured to determine the amount of the impairment. There are 6
categories of line impairments: Crosstalk; Echo or signal return; Frequency shift; Non-linear distortion; Jitter:
amplitude and phase; Transients: impulse noise, gain hits dropouts and phase hits.
3.1.9

Telephone Line Conditioning

Any process that prevents undesirable electrical signals from damaging networks or telecommunication
equipment and guards against data loss due to electrical noise, sags, and surges, etc., is known as Line
Conditioning. Any electronic device or circuit used in this process is called Line Conditioner.
3.2

PUBLIC SWITCHED TELEPHONE NETWORK

The Public Switched Telephone Network (PSTN) is a circuit-switched network that is used primarily for voice
communications worldwide, with more than one billion subscribers, in much the same way that the Internet is the
network of the world's public IP-based packet-switched networks. Originally a network of fixed-line analogue
telephone systems, the PSTN is now almost entirely digital and also includes mobile as well as fixed telephones.
For more than a hundred years, the PSTN was the only bearer network available for telephony.
3.2.1

Early PSTN

The first telephones had no network but were in private use, wired together in pairs. Users who wanted to talk to
different people had as many telephones as necessary for the purpose.
A user who wished to speak, whistled into the transmitter until the other party heard. Soon, however, a bell was
added for signalling, and then a switch hook, and telephones took advantage of the exchange principle already
employed in telegraph networks.
Each telephone was wired to a local telephone exchange, and the exchanges were wired together with trunks.
Networks were connected together in a hierarchical manner until they spanned cities, countries, continents and
oceans. This was the beginning of the PSTN, though the term was unknown for many decades. There are a
number of large private telephone networks which are not linked to the PSTN, usually for military purposes.
There are also private networks run by large companies which are linked to the PSTN only through limited
gateways, like a large Private Branch Exchange (PBX).

11

PSTN Block Diagram - General Voice Network Hierarchy

Class 1:
regional centers

Class 1:
regional centers

Class 2:
sectional centers

Fig.3.8
PSTN
Network
Hierarchy

Class 2:
sectional centers

Class 3:
primary centers

Class 3:
primary centers

Class 4:
toll centers

Class 4:
toll centers

Class 5:
local central office

Class 5:
local central office
Tandem office

Local
loops

Local
loops

Residential
customer

Business
customer

Residential
customer

Business
customer

Local Carrier's Domain of Influence, Intra-LATA

GOLDMAN & RAWLES:
FIG. 02-04

ADC3e

The term PSTN is used to refer to the public communication system that provides local, extended local and long
distance telephone services.
This system is made available to the public through a group of common communications carriers (Telephone
companies) who have agreed to exchange calls and connections on behalf of their subscribers as mandated by
international law and the laws of the country in which they do business and provide telephone service.
3.3

ECHO, HYBRIDS & ECHO SUPPRESSION

What is Echo? In audio signal processing and acoustics, an echo is a reflection of sound, arriving at the listener
some time after the original sound. Typical examples are the echo produced by the bottom of a well, by a
building, or by the walls of an enclosed room. A true echo is a single reflection of the sound source. The time
delay is the extra distance divided by the speed of sound. If so many reflections arrive at a listener that they are
unable to distinguish between them, the proper term is reverberation. In telephony, "Echo" is the reflected copy
of the voice heard some time later after one had spoken. The echo is a delayed version of the original signal. On a
telephone, if the delay is fairly significant (more than a few hundred milliseconds), it is considered annoying.

12

If the delay is very small, the phenomenon is called "sidetone" and while not objectionable to humans, can
interfere with the communication between data modems. In a phone conversation, echo is the sound of our
own voices being played back to us after a delay. Sometimes it is overlaid with the other partys voice.
Strong and delayed echo signals can be very annoying, and in some extreme cases, make conversation
impossible.
Echo Sources: PSTN, Mobile, and VoIP communications systems can get echo from a number of sources, so
network-based echo cancellers are critical for good quality of service. In todays competitive market, the
absence of an efficient echo cancellation method can prove detrimental on the carriers ability to retain
subscribers. There is the need to explore the sources of echo in telecommunications networks, the impact
of echo on service quality, and the methods used to keep echo under control. Standards used to benchmark
echo cancellers are introduced, as are performance requirements for echo cancellers in todays
PSTN/Mobile and VoIP-converged networks.
Types of Echo: Although it is difficult for a listener to differentiate them, there are two types of echo that occur
in the typical communications network: hybrid echo and acoustic echo. Hybrid echo (also known as
electrical echo) is caused by an impedance mismatch on the 4-wire to 2-wire conversion in wireline
networks. It is the primary network-induced echo in todays networks. Acoustic echo is created as a result
of insufficient acoustic isolation between the earpiece and the microphone in small handsets, or when
acoustic waves are reflected against a wall or enclosure, typically when using a hands-free unit.
Hybrid Echo: Hybrid echo is generated by the public switched telephone network (PSTN) through the reflection
of electrical energy by a device called a hybrid (hence the term Hybrid Echo). Most telephone local loops are
two-wire circuits while transmission facilities are four-wire circuits. Each hybrid produces echoes in both
directions, though the far end echo is usually a greater problem for voiceband.
Acoustic Echo: Acoustic echo arises when sound from a loudspeakerfor example, the earpiece of a telephone
handsetis picked up by the microphone in the same roomfor example, the microphone in the very
same handset. In other words, Acoustic Echo occurs when some of the sound energy from the speaker part
of the telephone set gets picked up and transmitted back by the microphone. The problem exists in any
communications scenario where there is a speaker and a microphone. Examples of acoustic echo are found
in everyday surroundings.

Echo Cancelling: Since the invention of the telephone, various techniques and technologies have been employed
to cancel echo. Todays echo cancellation technology uses digital signal processing (DSP) and echo
cancellation algorithms. Solving the echo problem in the Unacceptable and Limiting Case areas of the
Echo Objection Rate graph is the key to providing service quality.
Echo Suppression: The method used to reduce the echo heard on long telephone circuits, particularly circuits that
traverse satellite links is referred to as Echo Suppression. The telecommunications device or system used to
execute echo suppression is termed as an Echo Suppressor or "Acoustic Echo Suppressor" AES.
Echo suppressors were first developed in the 1950s in response to the first use of satellites for
telecommunications, but they have since been largely supplanted by better performing cancellers.

13

3.4

SWITCHING TECHNIQUES

What is meant by Switching?

Switching is the Process of Making, Breaking, Turning-On or Turning-Off Contacts, Circuits, Interconnections,
Networks, Terminals, Nodes, etc., in order to establish or disengage electronic or electrical setups, contacts or
connections.
Why Switching? The basic purpose of switching is to Activate (Enable) or Deactivate (Disable) Circuits,
Networks, Nodes, etc., by the use of manual or electronic switch or switches.
What Is A Switch? In Telecommunications Networks, a Switch Is An Electronic Device That Channels (Directs)
Incoming Data From Any Of Multiple Input Ports Or Nodes To The Specific Output Port Or Node That Will Take
The Information Or Data Toward Its Intended Destination. A telephone switch is the brains of telephone
exchange. It is a device for routing calls from one telephone / network to another.
Electronic Switching Systems: The Electronic Switching Systems were made possible by the invention of the
transistor. They apply the basic concepts of an electronic data processor, operating under the direction of a storedprogram control, and high-speed switching networks.
Digital Switches: Digital switches work by connecting two or more digital circuits together, according to a
dialed telephone number or typing the Email Address of a host Computer. Calls are setup between switches using
the Signalling System 7 protocol.
3.4.1

Requirements for Switching

Think about how things would be if we could only use our telephone connected permanently in order to talk to
just one other person! There would be millions of cables buried underground and trillions of copper or
aluminium wires passed overhead in towns and cities. We would not be very productive. So there are
requirements for switching systems to route our calls around the world. The purpose of a telecommunications
switching system is to provide the means, i.e., medium, to pass information from any terminal device to any other
terminal device selected by the originator. Three components are necessary for such systems:
a)

b)
c)

Terminals which are either input or output transducers. They convert the information into an electrical
signal at the transmitting end and convert electrical signal back into a usable form at the receiving
end.
A further function of a terminal is to generate and transmit control signals to indicate the required
destination of the information signal.
Transmission links to convey the information and control signals between the terminals and switching
centres.
Switching centres or exchanges to receive the control signals and to forward
or connect the information signals.

The third point in the above components is the main reason why switching techniques are required in ICT.
3.4.2

Switching Hierarchy

14

As the number of separate switching centres increases the number of different trunk routes between them
increases. Above about ten centres the number of trunk routes becomes very large and routes tend to contain too
few circuits to make the network economic. The process of centralizing switching centres can occur at several
levels leading to what is called a hierarchical network.

LS = Local Switching Centre

TS = Trunk Switching Centre

Hierarchical Network of Switching Centres

The Hierarchy of Switching Systems in its Most Basic Form Consists of Five Classes of Offices.
Fig. 3.9 Typical Switching
Hierarchy (USA)

Note that the only office / exchange that has people as its subscribers is the Class-5 office. The other exchanges
in this hierarchy have lower-level exchanges as their subscribers. Those lines connecting switching offices to
switching offices, rather than to subscribers, are called trunks. Trunk Circuits are the media or resources that
interconnect two or more switches or exchanges for the purpose of establishing communications networks,
basically, over long distances.
Fig. 3.910 Conventional
Representation of
Hierarchical Switching Network

3.5

CENTRAL OFFICE SWITCHING SYSTEM

A simple way of structuring a switched network is to arrange that each terminal has a direct transmission link to
every other terminal as shown below. Each terminal needs a switch to connect it to the required link and a switch
to make connection to a link in order to receive an incoming call. This lacks efficient and economic utilization of
resources.

15

Fig. 3.11 Meshed Telephone

Network Topology

A more efficient and economic way of structuring switching networks is the ATM - Centralized Switching design.
Asynchronous Transfer Mode (ATM) is the technology of choice for the Broadband Integrated Services Digital
Network (B-ISDN). The ATM transports a wide variety of services in a seamless manner. In this mode, user
information is transferred in a connection oriented fashion between communicating entities using fixed-size
packets, known as ATM cells. The ATM cell is fifty-three bytes long, consisting of a five byte header and a fortyeight byte information field, sometimes referred to as payload.
An ATM switch contains a set of input ports and output ports, through which it is interconnected to users, other
switches, and other network elements. Meaning that, the ATM Switch is central to all switches in a given
network. It might also have other interfaces to exchange control and management information with special
purpose networks.
Centralized Switching ATM

Fig. 3.12
Centralised Telephone Network Topology

Switching Techniques is the mechanism or method by which switching circuit, nodes and links are
interconnected. Three switching techniques have been proposed for building networks: Circuit witching; Packet
Switching; Message Switching. Each allows sharing communication facilities among multiple users (end
systems), and each uses equipment located at the nodes (intermediate systems) to replace the patch-panels used in
a point-to-point connection. Packet switching is most often used for data communication. Most networks consist
of many links (see the figure below) which allow more than one path through the network between nodes.
A data communications network must be able to select an appropriate path for each required connection. Any of
the three approaches (circuit switching, message switching, and packet switching) could yield minimum delay in a
particular situation, though situations where message switching yields minimum delay are rare. The relative
performance of circuit switching and packet switching depends strongly on the speed and "cost" of establishing a
connection.
TRUNK CIRCUITS

16

Trunk Circuits are the media or resources that interconnect two or more switches or exchanges for the purpose of
establishing long distance communications networks.
3.6

CIRCUIT SWITCHING

In telecommunications, a circuit switching network is one that establishes a fixed bandwidth circuit (or channel)
between nodes and terminals before the users may communicate, as if the nodes were physically connected with
an electrical circuit. The bit delay is constant during the connection, as opposed to packet switching, where packet
queues may cause varying delay.
In Circuit Switching a channel (circuit) is dedicated for the duration of a connection, even if no data is being
transferred. Once the circuit is established, the network is effectively transparent to the users, resulting in
negligible delays. It was developed to handle voice traffic but is now also used for data traffic. There is a
common misunderstanding that circuit switching is used only for connecting voice circuits (Analog or digital).The
concept of circuit switching can be extended to other forms of digital data. Dedicated path still remains between
two communicating parties and rest of the procedure remains same as voice circuits. But this time around the data
is transferred non-stop NOT in the form of packets and without any overhead bits.
Although possible, circuit switching is rarely used for transferring digital data (Except voice circuit) and this
scheme is not employed in networks where digital data needs to be transferred.
Each circuit cannot be used by other callers until the circuit is released and a new connection is set up. Even if no
actual communication is taking place in a dedicated circuit that channel remains unavailable to other users.
Channels that are available for new calls to be set up are said to be idle. This method of switching involves the
physical interconnection of two devices. A good example of circuit switching involves the Public phone network.
Circuit switching is the most familiar technique used to build a communications network. It is used for ordinary
telephone calls. It allows communications equipment and circuits, to be shared among users. Each user has sole
access to a circuit (functionally equivalent to a pair of copper wires) during network use. Refer to fig. 1.1 above.
3.6.1

Circuit-Switching Stages

a)
Circuit Establishment: During this stage the station requests connection from node which determines
best route and sends message to next link to establish the path. Each subsequent node continues the establishment
of a path. Once nodes have established connection, test message is sent to determine if receiver is ready/able to
accept message
b)

Data Transfer: Messages are sent from station to station during this phase.

c)

Circuit Disconnect: When data transfer is complete, one station initiates termination signals which must
be propagated to all nodes used in transit in order to free up resources (links).
Circuit Switching Applications

3.6.2

PSTN; Private Branch Exchanges (PBX) Private Wide Area Networks (often used to interconnect PBXs in a
single organization) Data Switches
3.6.3 Examples of circuit switched networks
PSTN; ISDN B-channel; Circuit Switched Data (CSD) and High-Speed Circuit-Switched Data (HSCSD) service
in cellular systems such as GSM; X.21 (Used in the German DATEX-L and Scandinavian DATEX circuit
switched data network), etc.

17

A
B

Fig. 3.13 Circuit

Switching
Data is transmitted along the dedicated path as rapid as possible.
3.7
3.7.1

PACKET SWITCHING
What is packet switching?

Packet switching is a communications method in which packets (discrete blocks of data) are routed between
nodes over data links shared with other traffic. In each network node, packets are queued or buffered, resulting in
variable delay. This contrasts with the other principal paradigm, circuit switching, which sets up a limited number
of constant bit rate and constant delay connections between nodes for their exclusive use for the duration of the
communication.
Packet switching refers to protocols in which messages are broken up into small packets before they are sent.
Each packet is transmitted individually across the net. The packets may even follow different routes to the
destination, depends on the type of packet switching. Thus, each packet has header information which enables to
route the packet to its destination. At the destination the packets are reassembled into the original message. To
prevent unpredictably long delays and ensure that the network has a reliably fast transit time, a maximum length
is allowed for each packet. It is for this reason that a message submitted to the transport layer may first have to be
divided by the transport protocol entity into a number of smaller packet units before transmission. In turn, they
will be reassembled into a single message at the destination.

Use of Packets
Fig. 3.14
Packet Switching

18

3.7.2

Circuit Switching vs. Packet Switching

In Circuit Switching networks, when establishing a call a set of resources is allocated for this call. These resources
are dedicated for this call, and cannot be used by any of the other calls. Circuit Switching is ideal when data must
be transmitted quickly, must arrive in sequencing order and at a constant arrival rate. There for when transmitting
real time data, such as audio and video, Circuit Switching networks will be used. Packet switching main
difference from Circuit Switching is that, the communication lines are not dedicated to passing messages from the
source to the destination. In Packet Switching, different messages can use the same network resources within the
same time period. Since network resources are not dedicated to a certain session the protocol avoid from waste of
resources when no data is transmitted in the session. Packet Switching is more efficient and robust for data that is
burst in its nature, and can withstand delays in transmission, such as e-mail messages, and Web pages.
3.7.3

Kinds of Packet Switching

There are two basic types of Packet Switching.

1. Virtual Circuit Packet Switching Networks
An initial setup phase is used to set up a route between the intermediate nodes for all the packets passed during
the session between the two end nodes. In each intermediate node, an entry is registered in a table to indicate the
route for the connection that has been set up. The packets passed through this route, have short headers,
containing only a Virtual Circuit Identifier (VCI). Each intermediate node passes the packets according to the
information that was stored in its table, in the setup phase and according to the packets header content. In this
way, packets arrive at the destination in the correct sequence. This approach is slower than Circuit Switching,
since different virtual circuits may compete over the same resources. As in Circuit Switching, if an intermediate
node fails, all virtual circuits that pass through it are lost. The most common forms of Virtual Circuit networks
are ATM and Frame Relay, which are commonly used for Public Data Networks (PDN).

Fig. 3.145
Virtual Circuit
Switching

19

2 Datagram Packet Switching Networks

This approach uses a different, more dynamic scheme, to determine the route through the network links. Each
packet is treated as an independent entity, and its header contains full information about the destination of the
packet. The intermediate nodes examine the header of the packet, and decide the next hop of this packet. In the
decision two factors are taken into account:

The shortest way to pass the packet to its destination - protocols such as RIP/OSPF is used to determine
the shortest path to the destination.

Finding a free node to pass the packet to - in this way, bottle necks are eliminated, since packets can reach
the destination in alternate routes. Thus, in this method, the packets don't follow a pre-established route,
and the intermediate nodes (the routers) don't have pre-defined knowledge of the routes that the packets
should be passed through.

Packets can follow different routes to the destination. Due to the nature of this method, the packets can reach the
destination in a different order than they were sent, thus they must be sorted at the destination to form the original
message. This approach is time consuming since every router has to decide where to send each packet. The main
implementation of Datagram Switching network is the Internet which uses the IP network protocol.
3.7.4

Routing in Packet Switched Network

Complex, crucial aspect of packet switched networks

Characteristics required: Correctness, Simplicity, Robustness, Stability, Fairness, Optimality & Efficiency.
Performance Criteria Used for Selection of Route:
Minimum hop, Least cost, Delay &Throughput
Example of Packet Switched Network
Communicating Nodes: Node-1 To Node-6
What Is Of Interest? Shortest Path (1-3-6)
Least Cost Path (1-4-5-6)
3.7.5

Fig.3.16 Packet Switched Routes

Routing Strategies

20

These include: Fixed Routing, Flooding Routing, Random Routing, Adaptive Routing.
Once a route is determined for a packet, it is entirely possible that the route may change for the next packet, thus
leading to a case where packets from the same source headed to the same destination could be routed differently,
hence, there is considerable delay in arrival time of packets.
3.7.6

Packet Switched Network

Packet switching is used to optimize the utilization of the channel capacity available in digital
telecommunications networks such as computer networks, to minimize the transmission latency (i.e. the time it
takes for data to pass across the network), and to increase robustness of communication. The most well-known
use of packet switching is the Internet and Local Area Networks (LAN). The Internet uses the Internet protocol
suite over a variety of data link layer protocols. For example, Ethernet and frame relay are very common. Newer
mobile phone technologies (e.g., GPRS, I-mode) also use packet switching. X.25 is a notable use of packet
switching in that, despite being based on packet switching methods, it provided virtual circuits to the user. These
virtual circuits carry variable-length packets. In 1978, X.25 was used to provide the first international and
commercial packet switching network, the International Packet Switched Service (IPSS). Asynchronous Transfer
Mode (ATM) also is a virtual circuit technology, which uses fixed-length cell relay connection oriented packet
switching.
Datagram packet switching is also called connectionless networking because no permanent connections are
established. Technologies such as Multiprotocol Label Switching (MPLS) and the Resource Reservation Protocol
(RSVP) create virtual circuits on top of datagram networks. Virtual circuits are especially useful in building
robust failover mechanisms and allocating bandwidth for delay-sensitive applications.
3.7.7

Advantages of Packet Switching

Line efficiency
Single node to node link can be shared by many packets over time
Packets queued and transmitted as fast as possible
Data rate conversion
Each station connects to the local node at its own speed
Nodes buffer data if required to equalize rates
Packets are accepted even when network is busy
Delivery may slow down
Priorities can be used
Packet switching optimizes the use of bandwidth by enabling many devices to route packets through the same
network channels. At any given time, a switch can route packets to several different destination devices, adjusting
the routes as required to achieve the best efficiency.

3.8

X.25 NETWORK

3.8.1

What is X.25?

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

21

telephone companies. 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.
3.8.2 What is an X.25 Network?
An X.25 network provides a means by which one X.25 DTE (a Terminal or Host of some kind) can exchange data
with one or more other X.25 Host, on the other side of the network.

Fig.3.17 X.25 Network

Data is carried within individual packets - X.25 is often referred to as a Packet Switching Protocol. This makes it
similar to a TCP/IP network - the difference is that IP networks employ a Connectionless protocol: each packet is
routed according to the information within that packet (typically by using the Destination Address). By contrast,
X.25 is a Connection-Oriented protocol: the routing information used by the network is carried only in the packets
used to establish the connection; thereafter addressing information is not required. This does, however, mean that
the X.25 network switching nodes need be aware of each connection, unlike IP routers.
3.9

ROUTING IN COMMUNICATION NETWORKS

3.9.1

What is Routing?

In Telecommunications, Routing is the process of selecting paths in a network along which to send / receive
network traffic. Routing is performed for many kinds of networks, including the telephone network (PSTN),
electronic data networks data communications (such as the Internet), and transportation (transport) networks.
In Circuit Switching, Routing is a process in which a dedicated circuit is established for the duration of the
transmission of each message. The dominant circuit switched network is the public switched telephone network
(PSTN), which is the worldwide collection of interconnected public telephone networks that was designed
primarily for voice traffic. In Packet Switching, Routing is the process of moving packets across a network from
one host to a another. It is usually performed by dedicated devices called routers. In packet switching networks,
routing directs (forwarding), the transit of logically addressed packets from their source toward their ultimate
destination through intermediate nodes; typically hardware devices called routers, bridges, gateways, firewalls, or
switches.
Routing Principles: The goal of routing in a communications network is to direct user traffic from a source to
the correct destination in accordance with the networks service requirements. The service requirements
for a given network are often expressed as a set of objectives. Objectives include maximizing network
performance (e.g., delay and throughput) and minimizing costs (e.g., equipment and facilities). The
underlying technology of the network imposes constraints on the network objectives. Constraints arise
from the limitations of the switching technology, the volume of user and network traffic, and the services
requested by the network. It is the multi-objective, multi-constraint nature of routing that makes it such a
complex problem in communications networks.

22

Routing in Circuit Switched Network: In Circuit witched networks, many connections will need different paths
through more than one switching node during peak hours, failures, breakdowns, network congestions, etc.
3.9.2

Routing Types

Alternate Routing: Different routing scenarios are used and possible routes between end offices predefined.
Originating switches select appropriate routes. Because Routes are listed in order of preference, different sets of
routes may be used at different times.
Adaptive Routing: Used by almost all packet switching networks. Here, Routing decisions change as conditions
(Failures, Congestions, etc) on the network change. Since routing decisions in this case are more complex,
information about the network required.
Dynamic Routing: A Dynamic Routing system selects routes based on current state information for the
network. The state information can be predicted or measured but the route will change depending on the available
state information at the time of the traffic request. The advent of smart digital switches in the network allowed for
the evolution of traffic routing from fixed hierarchical to dynamic non-hierarchical, in order to eliminate problems
that are presently well identified.
Fixed Routing: Single permanent route for each source destination pair is established. Routes are determined
using a least cost algorithm, and determined routes are fixed at least until changes in network topologies occur. A
typical Fixed Route Topology and a Routing Table is as shown below.
Flooding Routing: When applying link-state algorithms, each node uses as its fundamental data, a map of the
network in the form of a graph. To produce this, each node floods the entire network with information about what
other nodes it can connect to, and each node then independently assembles this information into a map. Using this
map, each router then independently determines the least-cost path from itself to every other node using a
standard shortest paths algorithm such as Dijkstra's algorithm. The result is a tree rooted at the current node such
that the path through the tree from the root to any other node is the least-cost path to that node. This tree then
serves to construct the routing table, which specifies the best next hop to get from the current node to any other
node. This process is known as Flooding Routing.

Fig. 3.18
Routing
Principles

23

3.9.3

Routing Strategies

These include: Fixed Routing, Flooding Routing, Random Routing &Adaptive Routing. Once a route is
determined for a packet, it is entirely possible that the route may change for the next packet, thus leading to a case
where packets from the same source headed to the same destination could be routed differently.
3.10

NETWORK CONTROL SIGNALING

3.10.1 What is Network Control?

Network Control is the transmission of signals or messages that perform call control, equipment configuration, or
information management functions. Network control can be centralized or distributed. The control of public
telecommunications networks is a centralized system as call processing is coordinated through a controlled
common channel signalling (CCS) network. E.g., The Internet uses distributed control as the switching
information dynamically changes in packet switching centres (routers) throughout the Internet network.
3.10.2 Control Signaling
a)

What is Signalling

Signalling is the Generation, Transmission and Reception of Information needed to Direct and Control the Setups
and Disconnections of Calls. It is the method used for the switching offices and the CPE to control the switching
of calls over the transmission facilities in the PSTN. Signaling is the generation, transmission, and reception of
information needed to direct and control the setup and disconnect of a call. There are many kinds of signals
which need to be transmitted between offices for an interoffice call, some of which include:

Addressing Outpulsing of dialed digits; Information signals Dial tone, audible ring, busy back tone,
reorder, etc. Supervision Off-hook, on-hook, seizures.

Control Signals are used in Switching to manage the establishment, maintenance, and termination of connections.
They include signaling from subscriber to network, and signals within the network. This process is known as
Control Signaling.
b)

Control Signals are of Two Types:

1. In-channel Signalling uses the same channel for both traffic and control signals. Each trunk line
will have its independent control signalling.
2. Common Channel Signalling (CCS) uses a channel specifically for carrying control signals for a
large number of traffic channels (D Channels in ISDN carry control signals).

Control Signals can be visuals (CLI) or audible (Dial Tone - DT).

c)

Signalling and Control

24

Referring to the internal communications functions that must take place within a network in order to ensure that it
operates properly, signalling and control communications are in contrast to the communications of user data, or
payload. The various elements of a network must have the capability to signal (i.e., alert and inform) each other,
indicating their status and condition. Typical status indications include available (dial tone), unavailable (busy),
and alerting (ringing signal). The endpoints (i.e., terminal devices) or end offices also must pass identification
information and instructions, such as the originating address, the target address, and the pre-selected carrier.
Switches and routers within the carrier network must pass information such as route preference and route
availability. Signalling and control systems and networks also handle billing matters, perhaps querying centralized
databases in the process. Network management information often is passed over signalling and control links.
Such information is used for remote monitoring, diagnostics, fault isolation, and network control. There are two
basic types of signalling and control: in-band and out-of-band.
d)

In-Band versus Out-Of-Band Signalling

i)

In-band Signalling

In the PSTN, in-band signalling is the exchange of signalling (call control) information within the same channel
that the telephone call itself is using. An example is DTMF signalling, which is used on most telephone lines to
exchanges.
ii)

Out-of-band Signalling

Out-of-band signalling is telecommunication signalling (exchange of information in order to control a telephone

call) that is done on a channel that is dedicated for the purpose and separate from the channels used for the
telephone call. Out-of-band signalling is used in Signalling System #7 (SS7), the standard for signalling among
exchanges that has control most of the world's phone calls for more than twenty years now.
e)

Channel-Associated versus Common-Channel Signalling

f)

Channel-Associated signalling employs a signalling channel which is dedicated to a specific bearer

channel.
Common-Channel signalling is so-called, because it employs a signalling channel which conveys
signalling information relating to multiple bearer channels. These bearer channels therefore have their
signalling channel in common.
The Common Channel Signalling (CCS)

CCS is a signalling method that uses a separate dedicated channel to send and receive signalling information for a
group of Trunks or Facilities by means of labelled messages. As mentioned earlier there are two methods of interoffice signaling, inband and out-of-band signaling. Inband signaling was used in the call example as the signaling
was sent on the same transmission path that was used for the call. When sending the signals, out-of-band
signaling uses a dedicated data link transmission path that is separate from the transmission path used for the call.
Out-of-band signaling is also referred to as common channel signaling (CCS). CCS is a form of data
communications specialized for the transfer of information between the nodes in a telecommunications network.

25

3.10.3

The Signalling System #7 (SS7)

There are two essential components to all telephone calls. The first, and most obvious, is the actual contentour
voices, faxes, modem data, etc. The second is the information that instructs telephone exchanges to establish
connections and route the content to an appropriate destination. Telephony signaling is concerned with the
creation of standards for the latter to achieve the former. These standards are known as protocols. SS7 or
Signaling System Number 7 is simply another set of protocols that describe a means of communication between
telephone switches in public telephone networks. They have been created and controlled by various bodies around
the world, which leads to some specific local variations, but the principal organization with responsibility for their
administration is the ITU-T.
Signalling System Number 7 (SS#7 or C7) is the protocol used by the telephone companies for interoffice
signalling. In the past, in-band signalling techniques were used on interoffice trunks. This method of signalling
used the same physical path for both the call-control signalling and the actual connected call. This method of
signalling is inefficient and is rapidly being replaced by out-of-band or common-channel signaling techniques. To
understand SS7 we must first understand something of the basic inefficiency of previous signaling methods
utilized in the Public Switched Telephone Network (PSTN). Until relatively recently, all telephone connections
were managed by a variety of techniques centered on in band signaling. A network utilizing common-channel
signalling is actually two networks in one:
1. First there is the circuit-switched "user" network which actually carries the user voice and data traffic. It
provides a physical path between the source and destination.
2. The second is the signalling network which carries the call control traffic. It is a packet-switched network
using a common channel switching protocol.
The original common channel interoffice signalling protocols were based on Signalling System Number 6 (SS#6).
Today SS#7 is being used in new installations worldwide. SS#7 is the defined interoffice signalling protocol for
ISDN. It is also in common use today outside of the ISDN environment. The primary function of SS#7 is to
provide call control, remote network management, and maintenance capabilities for the inter- office telephone
network. SS#7 performs these functions by exchanging control messages between SS#7 telephone exchanges
(signalling points or SPs) and SS#7 signalling transfer points (STPs).
The switching offices (SPs) handle the SS#7 control network as well as the user circuit-switched network.
Basically, the SS#7 control network tells the switching office which paths to establish over the circuit-switched
network. The STPs route SS#7 control packets across the signalling network. A switching office may or may not
be an STP.

Fig. 3.19
SS7 Architecture

26

a)

SS7 Protocol Layers:

The SS7 network is an interconnected set of network elements that is used to exchange messages in support of
telecommunications functions. The SS7 protocol is designed to both facilitate these functions and to maintain the
network over which they are provided. Like most modern protocols, the SS7 protocol is layered.
b)
Physical Layer (MTP-1): This defines the physical and electrical characteristics of the signaling links of
the SS7 network. Signaling links utilize DS0 channels and carry raw signaling data at a rate of 56 kbps or 64
kbps (56 kbps is the more common implementation).
c)
Message Transfer PartLevel 2 (MTP-2): The level 2 portion of the message transfer part (MTP Level
2) provides link-layer functionality. It ensures that the two end points of a signaling link can reliably exchange
signaling messages. It incorporates such capabilities as error checking, flow control, and sequence checking.
d)
Message Transfer PartLevel 3 (MTP-3): The level 3 portion of the message transfer part (MTP Level
3) extends the functionality provided by MTP level 2 to provide network layer functionality. It ensures that
messages can be delivered between signaling points across the SS7 network regardless of whether they are
directly connected. It includes such capabilities as node addressing, routing, alternate routing, and congestion
control.
e)
Signaling Connection Control Part (SCCP): The signaling connection control part (SCCP) provides
two major functions that are lacking in the MTP. The first of these is the capability to address applications within
a signaling point. The MTP can only receive and deliver messages from a node as a whole; it does not deal with
software applications within a node. While MTP network-management messages and basic call-setup messages
are addressed to a node as a whole, other messages are used by separate applications (referred to as subsystems)
within a node. Examples of subsystems are 800 call processing, calling-card processing, advanced intelligent
network (AIN), and custom local-area signaling services (CLASS) services (e.g., repeat dialing and call return).
The SCCP allows these subsystems to be addressed explicitly.
f)
ISDN User Part (ISUP): ISUP user part defines the messages and protocol used in the establishment and
tear down of voice and data calls over the public switched network (PSN), and to manage the trunk network on
which they rely. Despite its name, ISUP is used for both ISDN and nonISDN calls. In the North American
version of SS7, ISUP messages rely exclusively on MTP to transport messages between concerned nodes.
g)
Transaction Capabilities Application Part (TCAP): TCAP defines the messages and protocol used to
communicate between applications (deployed as subsystems) in nodes. It is used for database services such as
calling card, 800, and AIN as well as switch-to-switch services including repeat dialing and call return. Because
TCAP messages must be delivered to individual applications within the nodes they address, they use the SCCP for
transport.

h)
Operations, Maintenance, and Administration Part (OMAP): OMAP defines messages and protocol
designed to assist administrators of the SS7 network. To date, the most fully developed and deployed of these
capabilities are procedures for validating network routing tables and for diagnosing link troubles. OMAP includes
messages that use both the MTP and SCCP for routing.

27

MODULE 4

28

TRANSMISSION MEDIA FOR COMMUNICATION NETWORKS

4.1

OVERVIEW

Communication involves the transfer of information in the form of text, graphic images, sound or moving
pictures from one place to another. The media through which this transfer takes place is called the Transmission
Media. Transmission media refers to the physical means by which information is transferred. These include:
Equipment and systemsMetal or Electrical Wires, Coaxial Cables, Radio or Microwave Links, Satellite Links,
and infra-red optical systems employed in the transmission of electromagnetic signals. The transmission medium
used in a given network determines the effectiveness of such communication. It is important to understand the
types of media available, their method of construction and their advantages and disadvantages in different
situations.
4.1.1

What is Transmission?

Transmission is the means and the use of electronic devices to establish communication channels and trunks for
the purpose of sending (receiving) information from one place to another or to multi-points.
4.1.2

What are Transmission Media?

Transmission media are the highways and arteries that provide paths for communications devices. There is a
general tendency to say that one transmission medium is better than another. In fact, each transmission medium
has its place in the design of any telecommunications system. Each has characteristics which will make it the
ideal medium to use based on a particular set of circumstances. It is important to recognize the advantages of each
and develop a system accordingly. Factors to consider when choosing transmission media include: cost, ease of
installation and maintenance, availability, and most important, efficiency of transmission.
4.2

TYPES OF TRANSMISSION MEDIUM

There are two basic categories of Transmission Media. These are:

a)

Guided Transmission Media b) Unguided Transmission Media

4.2.1

GUIDED TRANSMISSION MEDIA

Guided transmission media use a cabling system that guides the data signals along a specific path. The data
signals are bound by the cabling system. Guided media is also known as bound media. "Cabling" is meant in a
generic sense, and is not meant to be interpreted as copper wire cabling only. Unguided transmission media
consist of a means for the data signals to travel but nothing to guide them along a specific path. The data signals
are not bound to a cabling media and are therefore often called unbound media. There are five basic types of
Guided Transmission Media: Open Wire Cable; Twisted Pair Cable; Coaxial Cable; Submarine Cable; Optical
Fibre.
Many engineers will argue that one transmission medium is the best, or better than some of the others. The reader
should keep in mind that each medium has advantages and disadvantages.
Which medium is best depends upon the purpose of the communications system and the desired end results. In
fact, most systems are a hybrid. That is, two or more media are combined to effect the most efficient

29

communication network infrastructure. There are many traffic signal systems that combine a twisted copper pair
infrastructure with wireless links to serve part of the system.
a)
The Copper Wire
There are four primary types of cables containing copper wire used for communication:
Open; Untwisted; Twisted Pairs; Coaxial cables
i)
Open Wire
Open wire is traditionally used
to describe the electrical wire
strung along power poles.
There is a single wire strung
between poles.
Fig. 4.1 Open Wires
ii)

Untwisted Copper Pair

Two insulated wires molded into a straight, flat, parallel pair cable can be used for short distance or low bit rate
communication. This is a cheap method but it is subject to cross talk and spurious noise pick up. The wires are
easy to terminate.
iii) Twisted Copper Pair

Fig. 4.2 Untwisted Pair

The wires in twisted pair cabling are twisted together in pairs. Each pair consists of a wire used for the positive
data signal and a wire used for the negative data signal. Any noise that appears on 1 wire of the pair will also
occur on the other wire. Because the wires are opposite polarities, they are 180 degrees out of phase. When the
noise appears on both wires, it cancels or nulls itself out at the receiving end. Twisted pair cables are most
effectively used in systems that use a balanced line method of transmission: polar line coding (Manchester
Encoding) as opposed to unipolar line coding (TTL logic).
Fig. 4.3
Twisted
Copper
Pair
Two insulated wires are twisted around each other to form a twisted pair. This system leads to better electrical
performance and significantly higher bit rates than untwisted pairs. These pairs are often then, in turn, twisted
with other pairs to form a cable that is capable of high-speed communication. The twisting of wires around each
other helps to reduce the noise by cancelling unwanted induced electrical signals and each pair usually carries a
single communication circuit. Twisted pairs have become the most common form of transmission media cable
used today having applications in telephone and computer networks. A Cable consisting of four twisted pairs that
have been, in turn, twisted and insulated is shown above (on the right). Twisted pair is categorized by the number
of twists per meter. A greater number of twists provide more protection against crosstalk, and other forms of
interference and results in a better quality of transmission. For data transmission, better quality equates to fewer
transmission errors.

30

iv) Unshielded Twisted Pair (UTP)

The degree of reduction in noise interference is determined specifically by the number of turns per foot.
Increasing the number of turns per foot reduces the noise interference. To further improve noise rejection, a foil or
wire braid "shield" is woven around the twisted pairs. This shield can be woven around individual pairs or around
a multi-pair conductor (several pairs).
Fig. 4.4 Unshielded Twisted Copper Pair

v)

Shielded Twisted Pair (STP)

Cables with shields are called shielded twisted pair and are commonly abbreviated STP. Cables without a shield
are called unshielded twisted pair or UTP. Twisting the wires together, results in reducing the characteristic
impedance for the cable. Typical impedance for UTP is 100 ohm for Ethernet 10BaseT cable. UTP or unshielded
twisted pair cable is used on Ethernet 10BaseT and can also be used with Token Ring. It uses the RJ line of
connectors (RJ45, RJ11, etc.) STP is used with the traditional Token Ring cabling or ICS - IBM Cabling System.
It requires a custom connector. IBM STP has a characteristic impedance of 150 ohms.
Fig. 4.5 Shielded Twisted Pair

b)

Coaxial Cable

The second transmission medium to be introduced was coaxial cable (often called coax), which began being
deployed in telephony networks around the mid-1920s. The components of coax are as shown below. In the
center of a coaxial cable is a copper wire that acts as the conductor, where the information travels. The copper
wire in coax is thicker than that in twisted-pair, and it is also unaffected by surrounding wires that contribute to
EMI, so it can provide a higher transmission rate than twisted-pair. The center conductor is surrounded by plastic
insulation, which helps filter out extraneous interference. The insulation is covered by the return path, which is
usually braided-copper shielding or aluminum foiltype covering. Outer jackets form a protective covering for
coax; the number and type of outer jackets depend on the intended use of the cable (e.g., whether the cable is
meant to be strung in the air or underground, whether rodent protection is required).
c)
Why the Name Co-Axial?
Coaxial cable is called "coaxial" because it includes one physical channel (the copper core) that carries the signal
surrounded (after a layer of insulation) by another concentric physical channel (a metallic foil or braid), and an
outer cover or sheath, all running along the same axis. The outer channel serves as a shield (or ground). Many of
these cables or pairs of coaxial tubes can be placed in a single conduit and, with repeaters, can carry information
for a great distance. In fact, this type of cable was used for high bandwidth and video service by the telephone
companies prior to the introduction of fibre in the 1980's.
d)

Optical Fibre

i)
What is Optical Fibre?
Optical Fibre or Fibre optics refers to the medium and the technology associated with the transmission of
information as light impulses along a strand of glass.

31

A fibre optic strand carries much more information than conventional copper wire and is far less subject to
electromagnetic interference (EMI). Almost all telephone long-distance (cross country) lines are now fibre optic.
Optical fibre consists of a glass or plastic core surrounded by a cladding with a higher refractive index. Data is
carried as pulses of light from a laser or high-power light emitting diode (LED). The light pulses are contained
within the core as a result of internal reflection. Optical fibre is non-electrical, and therefore completely immune
from electrical radiation and interference problems. Optical Fibre has the highest bit rate of all media. Each fibre
optic strand can support thousands of speech channels and multiple television channels simultaneously. High
bandwidths give enormous transmission capacity for national and intercontinental links and greater distances can
be tolerated between repeaters. The small physical size, non-corrosive construction and immunity from electrical
interference make optical fibre a highly attractive form of transmission media. Transmission over fibre optic
strands requires repeating (or regeneration) at varying intervals. The spacing between these intervals is greater
(potentially more than 100 km, or 50 miles) than copper based systems.
ii)
Optical Fibre Construction
Optical fiber consists of thin glass fibers that can carry information at frequencies in the visible light spectrum and
beyond. The typical optical fiber consists of a very narrow strand of glass called the core. Around the core is a
concentric layer of glass called the cladding. A typical core diameter is 62.5 microns (1 micron = 10 -6 meters).
Typically cladding has a diameter of 125 microns. Coating the cladding is a protective coating consisting of
plastic, it is called the Jacket. A typical core diameter is 62.5 microns (1 micron = 10 -6 meters).
An important characteristic of fiber optics is refraction.
Refraction is the characteristic of a material to either
pass or reflect light. When light passes through a medium,
it "bends" as it passes from one medium to the other.
Fig 4.6 Fibre Optic Construction

iii)
Types Of Fiber-Optic Cable
Fiber-optic cable is available in many sizes. It can have as few as a couple of pairs of fiber or it can have bundles
that contain upward of 400 or 500 fiber pairs. Figure 2.8 shows the basic components of fiber-optic cable. Each of
the fibers is protected with cladding, which ensures that the light energy remains within the fiber rather than
bouncing out into the exterior. The cladding is surrounded by plastic shielding, which, among others things,
ensures that you cant bend the fiber to the point at which it would break; the plastic shielding therefore limits
how much stress you can put on a given fiber. That plastic shielding is then further reinforced with Kevlar
reinforcing materialmaterial that is five times stronger than steelto prevent other intrusions. Outer jackets
cover the Kevlar reinforcing material, and the number and type of outer jackets depend on the environment where
the cable is meant to be deployed (e.g., buried underground, used in the ocean, strung through the air).

Fig. 4.7 Fibre Optic Cable

32

Fig. 4.8 Basic Fiber Optic Strand Construction

Transmission over fiber optic strands requires repeating (or regeneration) at varying intervals. The spacing
between these intervals is greater (potentially more than 100 km, or 50 miles) than copper based systems.
By comparison, a high speed electrical signal such as a E1 / T1 signal carried over twisted-pair must be
repeated every 1.8 kilometers or 6000 feet. Inside plant cable is constructed to be flexible and lightweight.
The cable may be coated to meet fire protection codes. Outside plant cable is constructed to withstand
immersion in water, will resist exposure to ultraviolet rays, and is protected from rodents and birds.
Fibre cables are produced in two basic forms:

Loose Tube Buffered Cable & Tight Buffered Cable

iv) Fibre Optic Cable Illustration

Fiber strands and cables are manufactured with a standard colour coding. This permits effective management of
cables because of the normally high strand counts contained within a cable. There are 24 colour
combinations used. A loose buffer tube cable with 576 strands would have 24 tubes coloured. Within each
buffer tube would be 24 fibre strands using the same colour scheme. Therefore, strand number 47 would be
in an orange buffer tube and have a rose with a black tracer coloured protective coating.

Fig.4.9
Fibre Optic
Colour Coding

v)

Two Basic Fibre Structures

Fiber optic cables are available in a wide variety of physical constructions. Fiber cables can be anything from
simple simplex or duplex (zipcord) cables used for jumpers to 144-fiber cable for intercity transmission.

33

Fig. 4.10
Fiber-optic Cable

vi)

Light Sources

In the realm of light sources, there are also two categories: light-emitting diodes (LEDs) and laser diodes. The
cheaper, lower-performer category is LEDs. LEDs are relatively inexpensive, they have a long life, and they are
rather tolerant of extreme temperatures. However, they couple only about 3% of light into the fiber, so their data
rates are low, currently about 500Mbps. Laser diodes are capable of much higher transmission speeds than LEDs.
A laser diode is a pure light source that provides coherent energy with little distortion. Therefore, laser diodes are
commonly used for long-haul and high-speed transmission. Laser diodes offer better performance than LEDs, and
they are more expensive, although the cost of these components has been dropping about 40% per year.
vii)

Electrical Signal v Optical Signal

Twisted copper pairs and coaxial cables transmit signals in electrical form as a flow of electrons. By contrast,
optical fibre transmits signals in the form of pulses of light generated by miniature lasers and received by tiny
diodes. Both forms of transmission travel at the fastest speed known to physics, the speed of light which is
approximately 300,000 km per second. Currently all switching is done electrically, so that light signals are
converted into electrical signals for the switching process and then afterwards reconverted back into light signals.
However, in future we can expect the development of optical switches. Indeed developments may be so profound
that some have suggested that, if the last century was that of the electron (electricity), then this century will be
that of the photon (photonics).
viii)

How Fiber-Optic Transmission Works

As shown below, in fiber-optic transmission, the digital bit-stream enters the light source, in this case the laser
diode. If a one bit is present, the light source pulses light in that time slot, but if there is a zero bit, there is no light
pulse (or vice versa, depending on how it is set up). The absence or presence of light therefore represents the
discrete 1s and 0s. Light energy, like other forms of energy, attenuates as it moves over a distance, so it has to run
though amplification or repeating process. As mentioned earlier, until about 1994, electronic repeaters were used
with fiber, so the optical signal would have to stop; be converted into electrical energy; be resynchronized,
retimed, and regenerated; and then be converted back into optical energy to be passed to the next point in the
network. This was a major problem because it limited the data rate to 2.5Gbps. But some developments
introduced in the early 1990s dramatically changed long-distance communications over fiber.

Fig. 4.11
Fiber-optic
Transmission

34

ix)

Applications of Fiber Optics

Fiber has a number of key applications. It is used in both public and private network backbones, so the vast
majority of the backbones of the PSTNs worldwide have been upgraded to fiber. The backbones of Internet
providers are fiber. Cable TV systems and power utilities have reengineered and upgraded their backbones to fiber
as well. Surprisingly, electric power utilities are the second largest network operator after the Telcos. They have
vast infrastructures for generating and transmitting electricity; these infrastructures rely on fiber-optic
communications systems to direct and control power distribution. After they have put in fiber, electric companies
have often found themselves with excess capacity and in a position to resell dark fiber to interested parties
including several Telcos! When we lease dark fiber, we basically leasing a pair of fibers without the active
electronics and photonics included, so we are responsible for acquiring that equipment and adding it to the
network.
Another application of fiber is in the local loop. There are numerous arrangements of fiber in the local loop,
including HFC (i.e., fiber to a neighborhood node and then coax on to the subscribers); fiber to the curb with a
twisted-pair solution to the home; fiber to the home that terminates on its own individual optical unit; and passive
optical networking, which greatly reduces the cost of bringing fiber to the home.
x)

Advantages of Optical Fibre:

ix)

Noise immunity: RFI and EMI immune (RFI - Radio Frequency Interference, EMI -Electromagnetic
Interference) - Security: cannot tap into cable - Large Capacity due to BW (bandwidth) - No corrosion Longer distances than copper wire - Smaller and lighter than copper wire - Faster transmission rate
Disadvantages of optical fibre:

Physical vibration will show up as signal noise - Limited physical arc of cable. Bend it too much and it
will break - Difficult to splice.

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

f)

Submarine Communications Cable

A submarine communications cable is a cable laid beneath the sea to carry telecommunication signals between
countries. The first submarine communications cables carried telegraphy traffic. Subsequent generations of
submarine cables carried first telephony traffic, then data communications traffic. All modern cables use
fibre Optic technology to carry digital payloads, which are then used to carry telephone traffic as well as
Internet and private data traffic. As of 2003, submarine cables link all the world's continents except
Antarctica.
i) Outline of Submarine Cable Connection Technology
The maximum depth the trans-Pacific submarine communication cable passes through is about 8,000 m.
Submarine cables laid along the seabed are directly subject to the ambient water pressure. As the maximum
length of manufactured submarine cables is 30 - 40 kilometers, connectors are necessary for them to cover long
distances. Technical conditions necessary for submarine cable connecting are:

35

No deformation caused by pressure, even at depths of 8,000 m.

No decrease in insulation performance as power is supplied to relays
Long-term stability to match the designed life of 25 years.
An example of a submarine cable connection is illustrated below. The figure shows the basic geometry and
components of a typical connection (known as JB - Joint Box). Detailed technicalities are not shown.

Fig. 4.12
Fibre Cable
Joint Box (JB)

A cross-section of a submarine communications cable. 1. Polyethylene. 2. "Mylar" tape. 3. Stranded

steel wires. 4. Aluminum water barrier. 5. Polycarbonate. 6. Copper or aluminum tube. 7. Petroleum
jelly. 8. Optical fibers.

Fig. 4.13
Fibre Optic Sub-marine
Cable (Cross-section)

ii) Submarine Fiber Optic Cable (Undersea Fiber Optic Cable)

Submarine cables are used in fresh or salt water. To protect them from damage by fishing trawlers and
boat anchors they have elaborately designed structures and armors. Long distance submarine cables are
especially complex designed.
Fig. 4.14 Submarine Cable

Submarine cables are laid using special cable-laying ships, such as the modern Ren Descartes, operated
by France Tlcom Marine.

36

Fig. 4.15
French Cable-laying Ship

Fig. 4.16 Maps of submarine cables throughout the world

37

4.4

UNGUIDED TRANSMISSION MEDIA

Unguided transmission media is data signals that flow through the air. They are not guided or bound to a channel
to follow. They are classified by the type of wave propagation.
RF Propagation
There are three types of RF (radio frequency) propagation:

a)

Ground Wave - Ionospheric - Line of Sight (LOS)

Ground Wave

Ground wave propagation follows the curvature of the Earth.

Ground waves have carrier frequencies up to 2 MHz. AM
radio is an example of ground wave propagation.
b)

Fig. 4.17 Ground Waves

Ionospheric Wave

Ionospheric propagation bounces off of the Earth's

ionospheric layer in the upper atmosphere. It is
sometimes called double hop propagation. It
operates in the frequency range of 30 - 85 MHz.
Because it depends on the Earth's ionosphere, it
changes with the weather and time of day. The signal
bounces off of the ionosphere and back to earth.
c)

Fig. 4.10 Ionospheric Propagation

Line of Sight (LOS) Propagation

Line of sight propagation transmits exactly in the

line of sight.

Fig. 4.18 Line of Sight Propagation

The receive station must be in the view of the transmit station. It is sometimes called
space waves or tropospheric propagation. It is limited by the curvature of the Earth for ground-based stations
(100 km, from horizon to horizon). Reflected waves can cause problems. Examples of line of sight propagation
are: FM radio, microwave and satellite. The frequency spectrum operates from 0 Hz (DC) to gamma rays (10 19
Hz). Radio frequencies are in the range of 300 kHz to 10 GHz. We are seeing an emerging technology called
wireless LANs. Some use radio frequencies to connect the workstations together, some use infrared technology.
d)

Microwave Transmission Medium

Microwave was used during World War II in military applications, and when it was successful in that
environment, it was introduced into commercial communications. Microwave was deployed in the PSTN as a
replacement for coaxial cable in the late 1940s. As mentioned earlier, twisted-pair and coax both face limitations
because of the frequency spectrum and the manner in which they are deployed. But microwave promises to have a
much brighter future than twisted-pair or coax.
Many locations cannot be cost-effectively cabled by using wires (e.g., the Sahara, the Amazon, places where
buildings are on mountaintops, villages separated by valleys), and this is where microwave can shine.

38

In addition, the microwave spectrum is the workhorse of the wireless world: The vast majority of wireless
broadband solutions operate in the microwave spectrum.
i)

General Characteristics of Microwave Transmission

Microwave transmission is line of sight transmission.

The transmit station must be in visible contact with
the receive station.
Fig. 4.19 Microwave Transmission

This sets a limit on the distance between stations depending on the local geography. Typically the line of sight
due to the Earth's curvature is only 50 km to the horizon. Repeater stations must be placed so the data signal can
hop, skip and jump across the country. Microwaves operate at high operating frequencies of 3 to 10 GHz. This
allows them to carry large quantities of data due to their large bandwidth. Microwave is defined as falling in the 1
GHz to 100 GHz frequency band. But systems today do not operate across this full range of frequencies. In fact,
current microwave systems largely operate up to the 50 GHz range.
At the 60 GHz level, we encounter the oxygen layer, where the microwave is absorbed by the surrounding
oxygen, and the higher frequencies are severely affected by fog. However, we are now producing systems called
Virtual Fibre that operate in the 70 GHz to 95 GHz range at very short distances.
e)

Satellite Transmission Medium

i)

What are Satellites?

Satellites are transponders (units that receive on one frequency and retransmit on another) that are set in
geostationary orbits directly over the equator. These geostationary orbits are 36,000 km from the Earth's surface.
At this point, the gravitational pull of the Earth and the centrifugal force of Earth's rotation are balanced and
cancel each other out. Centrifugal force is the rotational force placed on the satellite that wants to fling it out into
space.

Fig. 4.20 Satellite Orbiting the Equator

The uplink is the transmitter of data to the satellite. The downlink is the receiver of data. Uplinks and downlinks
are also called Earth stations because they are located on the Earth.

39

Fig. 4.21 Satellite Footprints,

Up & Down Links

The footprint is the "shadow" that the satellite can transmit to, the shadow being the area that can receive the
satellite's transmitted signal. Another thing thats very important about and unique to satellites is the broadcast
property. After we send data uplink to a satellite, it comes back downlink over the entire footprint. So a satellite
can achieve point-to-multipoint communications very cost-effectively. This has a dramatic impact on the media
business.

Fig. 4.22
Satellite System
Connectivities

An interesting design parameter associated with satellites is that as the number of locations increases, the
economic benefit of using satellites increases. With leased lines, the more locations and the greater the distances
between them, the more expensive the network.
ii)

Satellite Power Requirements

The frequency spectrum in which most satellites operate is the microwave frequency spectrum. Therefore,
microwave and satellite signals are really the same thing. The difference is that with satellite, the repeaters for
augmenting the signals are placed on platforms that reside in high orbit rather than on terrestrial towers. Of
course, this means that the power levels associated with satellite communications are greater than those of
terrestrial microwave networks.
The actual power required depends on the orbit the satellite operates in (geosynchronous-orbit satellites require
the most power, and low-earth-orbit satellites require the least), as well as the size of the dish on the ground. If the
satellite has a lot of power, we dont need as big a dish on the ground. This is particularly important for satellite
TV, where the dish size should be 2 feet (0.6 m) or smaller.

40

iii)

Very Small Aperture Terminals (VSATs)

VSATs Business enterprises use VSAT networks as a means of private networking, essentially setting up point-topoint links or connections between two locales. A VSAT station is so compact that it can be put outside a window
in an office environment. VSATs are commonly deployed to reduce the costs associated with leased lines, and
depending on how large the network is, they can reduce those costs by as much as 50%. Most users of VSATs are
enterprises that have 100 or more nodes or locations (e.g., banks that have many branches, gas stations that have
many locations, convenience stores). VSATs elegantly and economically help these types of enterprises do things
such as transport information relevant to a sale made from a remote location to a central point, remotely process
information, and process reservations and transactions. Another use for VSATs is business video.
Before about 1994, the only way to do point-to-multipoint video was by satellite. No terrestrial conference
bridges would allow it. So, if you wanted to have a CEO provide a state-of-the-company address to all employees
at different locations, the easiest way to get that footprint was by satellite.

Fig. 4.23 VSAT Systems

4.5

TRANSMISSION IN THE PSTN

4.5.1

Transmission Parameters

In mathematics, statistics, engineering, etc., parameters (auxiliary measures) are quantities that define certain
characteristics and status of functions or systems. In telecommunications, and for that matter, in transmission,
parameters are the properties or measured variables of transmission systems.
Some Transmission Parameters Include:

4.5.2

FTX - Transmitting Frequency, FRX Receiving Frequency, PT Transmitting Power, GT Tx Gain

BW Tx Bandwidth (Capacity), Attenuation, Fading, Noise Factor (F), Distortion, Echo, Standing
Wave Ratio, Antenna Diameter, Quality of Service (QoS), etc.
Analogue Transmission

Analogue Transmission is a method of conveying voice, data, image, signal or video information using a
continuous signal (carrier) which varies in amplitude, phase, or some other property in proportion to that of a
variable input signal. Analogue signals have a continuous wave-like form. The further the signal travels, the
more likely it is to become degraded and therefore, over anything other than short distances, these signals are
boosted in strength at various points in the network and any subsequent distortions are amplified at the same time.

41

4.5.3

Digital Transmission

This is a transmission system in which:

(a) All circuits carry digital signals.
(b) The signals are combined into one or more serial bit streams that include all framing and supervisory signals.
Note: A-D / D-A conversion, if required, is accomplished externally to the system.
Digital signals are discrete in form and essentially represent power on or off or to use the binary language of
computers zeros (0s) & ones (1s). Digital signals can be REGENERATED OR recreated at their destination in
exactly the form they left their starting point despite any distortion along the way, since the only two possibilities
for the original signal are 0s and 1s. All communications networks whether telecommunications or broadcasting,
whether fixed or mobile are increasingly becoming digital because the quality of the signal is so much better
and the capacity of the network is so much greater.
Analogue signals can be combined (i.e., multiplexed) by combining them with a carrier frequency. When there is
more than one channel, this is called frequency division multiplexing (FDM). FDM was used extensively in the
past but now has generally been replaced with the digital equivalent: time division multiplexing (TDM). The most
popular TDM system is known as Tier 1 (T1). In a T1 system, an analog voice channel is sampled 8,000 times per
second, and each sample is encoded into a 7-bit byte. Twenty-four such channels are mixed on these two copper
pairs and transmitted at a bit rate of 1.544 megabits per second. T1 remains an important method of transmitting
voice and data in the PSTN.

42

CHAPTER 5

ANALOGUE NETWORKS, MODEMS AND MULTIPLEXORS, PDH NETWORKS

(E1 - E4, T1 - T4, SONET/SDH (STS-N, STM-N, OC-N, PACKET FORMATS/PAYLOAD)

NOTE: This Chapter Is Given Out As Assignment 1

CHAPTER 6
INTEGRATED SERVICES DIGITAL NETWORKS

6.1

INTEGRATED SERVICES DIGITAL NETWORK (ISDN) OVERVIEW

a)

ISDN Defined

ISDN, which stands for Integrated Services Digital Network, is a system of digital phone connections which has
been available for some time now. This system allows voice and data to be transmitted simultaneously across the
world using end-to-end digital connectivity. With ISDN, voice and data are carried by bearer channels (B
channels) occupying a bandwidth of 64 kb/s (bits per second). Some switches limit B channels to a capacity of 56
kb/s. A data channel (D channel) handles signaling at 16 kb/s or 64 kb/s, depending on the service type.
Note that, in ISDN terminology, "k" means 1000 (103), not 1024 (210) as in many computer applications (the
designator "K" is sometimes used to represent this value); therefore, a 64 kb/s channel carries data at a rate of
64000 b/s.
b)

What Exactly is the ISDN?

ISDN is basically the telephone network turned all-digital end to end, using existing switches and wiring (for the
most part) upgraded so that the basic "call" is a 64 kbps end-to-end channel. The ISDN is an evolutionary circuit
switched network based on digital telephony. It uses a common set of interface standards and allows users to send
and receive information over the network. ISDN offers end-to-end (caller to receiver) digital connectivity
between Terminal Equipment (TEs), via Network Terminators (NTs) and digital exchanges, both private and
public. The ISDN is a set of international standards for access to advanced, all-digital public telecommunications
networks.

43

The key elements of this definition are:

Integrated Services: Voice Video Image Data - Mixed media at a number of standard data rates.

Digital: Digital terminal equipment - Digital local loops - Digital trunks - Digital switching - Digital
signalling
Network: Worldwide, interoperating communications fabric under distributed control using common
standards.

ISDN standards have been defined by the ITU-T, a branch of the United Nations' International
Telecommunications Union (ITU), in the series I and Q recommendations.
c)

Narrowband ISDN (N-ISDN)

Anticipating user demand for end-to-end digital services the world's telephone companies agreed in 1984 under
the auspices of CCITT to build a new, fully digital, circuit-switched telephone system by the early part of the 21st
century. This system was called ISDN (Integrated Services Digital Network) and its primary goal was to
integrate the voice and non-voice services.
6.2

ISDN BACKGROUND

Before ISDN was introduced, dedicated networks were required to provide services of different nature, e.g. POTS
(Plain Old Telephone Service) analogue service, packet service, telex service, data service, etc. The PSTN
provides analogue telephone services to customers; the PSTN provides packet services to customers. Different
networks were required because of the very different transmission characteristics. Dedicated and isolated network
requirements lead to a number of drawbacks: high costs, low efficiency, and inconvenience. ISDN, based on the
telephony network, was conceived of to provide multiple voice and non-voice services over a single network and
a digital user network interface over regular phone lines, instead of dedicated and isolated user-network
interfaces.
Using ISDN, users not only can do telephony, but can access additional benefits such as telecommuting, Internet
access, and video conferencing. These services were not possible in large deployment with regular services
provided by the phone companies. ISDN is an integrated solution for providing basic telephony and data services,
whilst offering more telephony services such as supplementary services. Its proven technology continues to be
deployed and hence must be tested and maintained.
6.2.1

Development of the ISDN

ISDN is often referred to as the successor of the existing public networks like the telephone network, data- and
text networks (Telegraphy). A continuation of this development leads to an overall digital network. ISDN is such
a network, providing moreover services available - in the past - only in other networks and/or with other
interfaces, e.g., Teletex, Telefax. ISDN is built on top of IDN. It allows communication, not only with older
equipment, for instance an analogue telephone connected analogue to the local exchange, but also with other
networks, e.g., PSPDN (Public Switched Packet Data Network). Thus ISDN can be seen as an evolution rather
than a revolution.

6.2.2

ISDN is Integrated Services

44

In the past; video, audio, voice and data services needed different types of communication channels. One of the
main advantages of ISDN is the ability to integrate these features over the same network and cable plant. Not only
is this possible using ISDN technology, but the quality of the transmission is better also. In the past, four networks
were needed and video was distributed on coaxial lines, audio over balanced lines, voice used copper cable pairs
and data services required coaxial or twisted pair cables. Using one network allows reductions in installation
costs, as well as easier installation. Other features available include demand networking, automatic bandwidth and
on the fly connectivity. Advances in the services available are due to ISDN being digital.
6.2.3

ISDN is Digital

Data applications, in particular, seemed to have problems with the old analogue services. This is due to the fact
that computers are digital devices and the transmission of data needs to be modified from binary to analogue
tones, then changed back to binary when it is received. This process requires a modem, which handles the
MOdulation and DEModulation (MODEM) of the data. Whilst the data is in transit it is susceptible to outside
influences like noise, spikes and echoes. Bandwidth is also limited, with the speed of modems being close to the
maximum possible.
6.24

ISDN is a Network

Networks require high speed connectivity if they are going to be useful. ISDN is an excellent vehicle for
connecting LANs, because it scales in increments of 64 kbps. Computers are digital devices, as is ISDN, meaning
that no translation of information is required, improving quality and speed. Due to the characteristics of ISDN it
can be used, with the same level of performance, across a room or halfway across the continent. This has
unlimited benefits.
6.2.5

How ISDN Works

ISDN carries voice and data on bearer (B) channels which transmit at 64 Kbps each. (H channels, which are the
functional equivalent to B channels, are available and provide faster bit rates.) A data (D) channel, sometimes
referred to as a delta channel, operates at 16 or 64 Kbps and provides signaling to construct and tear down a
connection, request network services, and route data over the B channels. The D channel can also be used to
transmit user packet or frame data at times when bandwidth on the D channel is not required for signaling and
control. Utilizing the D channel in this way provides the most efficient use of ISDN.
ISDN has three different services: 1) Basic Rate Interface (BRI), 2) Primary Rate Interface (PRI), and 3)
Broadband (B-ISDN). BRI is the most common service and was intended to be the most widely available for
residential customers. BRI services provide two B channels and one D channel (2B+D).
PRI services are implemented differently in North America and Japan than in Europe where they are the most
common services. European PRI services deliver 30 B channels and one D channel (30B+D). North American and
Japanese PRI services consist of 23 B channels and one D channel (23B+D). B-ISDN is still under development
but will support up to 622 Mbps transmission rates over a fiber optic network.
One major advantage of the ISDN architecture is its dynamic bandwidth allocation feature - also known as
bandwidth-on-demand, inverse multiplexing, and channel aggregation. Dynamic bandwidth allocation is the
process of combining any or all of the B channels into a single broadband conduit. For PRI service, the combining
of multiple B channels is often programmed into the ISDN switch serving the location.
6.2.6

Access to ISDN

45

Two types of ISDN have been specified: Narrowband-ISDN (N-ISDN) and Broadband-ISDN (B-ISDN). The
main difference between NISDN and B-ISDN is the transmission capacity and the used transfer mode. N-ISDN
can serve with a capacity of up to 2Mbps, while for B-ISDN exist specifications for 150Mbps and 600Mbps.
Because the definition of B-ISDN and its services are still in process, the main point of effort will be layed on NISDN. The term ISDN will be used for N-ISDN. If B-ISDN is referred, it will be explicitly named.
6.2.7

Benefits of ISDN

ISDN affords many benefits to service providers and customers. The increasing popularity of ISDN allows pricing
that continues to fall and compete with standard analog service. Some of the many benefits are:

6.3

Simultaneous audio, video, and data services over a single pair of copper wires reduces infrastructure and
maintenance costs for service and subscribers.
ISDN BRI service can use data compression which boosts the 128 Kbps transmission rate to between 256
Kbps and 632 Kbps, depending upon the compression ratio used.

Digital transmissions produce clearer and quieter voice telephone service and more reliable and accurate
connectivity than analog technology.

Remote computer users benefit from high performance ISDN connections at home or on the road.

ISDNs dynamic bandwidth allocation feature accommodates the bandwidth-intensive applications.

Up to eight different devices can be operated simultaneously over a single ISDN line.

LAN protocols such as IP and IPX are better supported by ISDN connections across WANs due to faster
connect times (between 1 and 4 seconds) than analog service (between 10 and 40 seconds).

ISDN is compatible with other WAN services like X.25, Frame Relay, Switched Multi-megabit Data
Services (SMDS) and higher speed services like Asynchronous Transfer Mode (ATM).
ISDN DEVICES AND NETWORK POINTS

In the context of ISDN standards,

Standard Devices refers not to
actual hardware, but to standard
collections of functions that can
usually be performed by individual
hardware units.

Figure 6.1 Sample ISDN Configuration Illustrates Relationships Between Devices and Reference Points

NOTE :
ET - Exchange terminal
LT - Line Terminal

TA - Terminal Adaptor
TE1 - ISDN Terminal

46

NT1 - Network Termination 1

NT2 - Network Termination 2

TE2 - Non-ISDN Terminal

S,T,U,V - Reference points

The NT1, LT and ET will be provided by Telecom as an inherent part of the ISDN service. The TA may also be
provided with some service offerings. The interface is functionally organised into the first three layers of the ISO
Open Systems Interconnection 7-layer model, consisting of the physical layer (Layer 1), the data link layer (Layer
2) and the network layer (Layer 3). Layer 1 for the Primary rate access is the focus of this Specification.
ISDN devices 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 (TE1). Non-ISDN terminals, such as DTE, that predate the ISDN
standards are referred to as terminal equipment type 2 (TE2). TE1s connect to the ISDN network through a fourwire, twisted-pair digital link. TE2s connect 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-232-C), V.24, and
V.35.
Beyond the TE1 and TE2 devices, the next connection point in the ISDN network is the network termination type
1 (NT1) 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. ISDN specifies a number of reference points that
define logical interfaces between functional groups, such as TAs and NT1s. ISDN reference points include the
following:

RThe reference point between non-ISDN equipment and a TA.

SThe reference point between user terminals and the NT2.

TThe reference point between NT1 and NT2 devices.

UThe 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 NT1 function is not provided by the carrier network.
The Figure below illustrates a sample ISDN configuration and shows three devices attached to an ISDN switch at
the central office. Two of these devices are ISDN-compatible, 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 NT1/2 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.
6.3.1

Sample of ISDN Configuration

Sample of ISDN Configuration Illustrating Relationships between Devices and Reference Points.

47

Fig. 6.2 ISDN

Configuration

The ISDN Standard Devices are:

a)

Terminal Equipment (TE); Terminal Adapter (TA) ; Network Termination 1 (NT1); Network Termination
2 (NT2); Exchange Termination (ET)
Terminal Equipment (TE)

A TE is any piece of communicating equipment that complies with the ISDN standards. Examples include: digital
telephones, ISDN data terminals, Group IV Fax machines, and ISDN-equipped computers. In most cases, a TE
should be able to provide a full Basic Rate Access (2B+D), although some TEs may use only 1B+D or even only
a D channel.
b)

Terminal Adapter (TA)

A TA is a special interface-conversion device that allows communicating devices that don't conform to ISDN
standards to communicate over the ISDN. The most common TAs provide Basic Rate Access and have one RJtype modular jack for voice and one RS-232 or V.35 connector for data (with each port able to connect to either of
the available B channels). Some TAs have a separate data connector for the D channel.
c)

Network Termination (NT1 and NT2)

The NT devices, NT1 and NT2, form the physical and logical boundary between the customer's premises and the
carrier's network. NT1 performs the logical interface functions of switching and local-device control (local
signalling). NT2 performs the physical interface conversion between the dissimilar customer and network sides of
the interface. In most cases, a single device, such as a PBX or digital multiplexer, performs both physical and
logical interface functions. In ISDN terms, such a device is called NT12 ("NT-one-two") or simply NT.
d)

Exchange Termination (ET)

The ET forms the physical and logical boundary between the digital local loop and the carrier's switching office.
It performs the same functions at the end office that the NT performs at the customer's premises. In addition, the
ET:
1. Separates the B channels, placing them on the proper interoffice trunks to their ultimate destinations

48

2. Terminates the signalling path of the customer's D channel, converting any necessary end-to-end
signalling from the ISDN D-channel signalling protocol to the carrier's switch-to- switch trunk signalling
protocol
e)

Reference Points (Connection Interfaces)

User terminals can be attached in different ways to the ISDN network, depending on the terminal capabilities and
configuration needs of the user station. The CCITT (International Telegraph and Telephone Consultative
Committee) Recommendations, contain a specification of the subscriber station and its interface to the network.
R, S, T, U are reference points (or interfaces) which determine mechanical and electrical characteristics, and
specifications of operating procedures. TE is the terminal equipment. It can be a telephone, telefax, computer etc.
NT is the network termination, and can be divided into the two units NT1 and NT2. NT1 translates the signals at
ref. point T into Signals for the network. NT2 allows the connection of multiple TE to ISDN. Up to 8 TE's can be
connected with Basic Rate Access to the so called S-bus. If only this passive bus (S-bus) is attached to NT2, the
NT2 doesn't need to provide any function and can be called "zero NT2". In other configurations NT2 can act as a
private branch exchange, concentrating the traffic of several TE's including the passive bus. NT2 could also
handle the internal traffic between TE's. Older Terminal equipments which do not meet the specification of the S
reference point can be attached through a Terminal Adaptor. The responsibility of the network provider can end at
the ref. points S, T or U, depending on the national regulations.
6.3.2

Advantages of ISDN

a)

Speed

The modem was a big breakthrough in computer communications. It allowed computers to communicate by
converting their digital information into analogue signal to travel through the public phone network (the PSTN).
There is an upper limit to the amount of information that an analogue telephone line can hold. Currently, it is
about 56 kb/s bi-directionally. Commonly available modems have a maximum speed of 56 kb/s, but are limited by
the quality of the analogue connection and routinely go about 45-50 kb/s. Some phone lines do not support 56
kb/s connections at all.
b)

Multiple Devices

Previously, it was necessary to have a separate phone line for each device we wished to use simultaneously. For
example, one line each was required for a telephone, fax, computer, bridge/router, and live video conference
system. Transferring a file to someone while talking on the phone or seeing their live picture on a video screen
would require several potentially expensive phone lines. SDN allows multiple devices to share a single line. It is
possible to combine many different digital data sources and have the information routed to the proper destination.
Since the line is digital, it is easier to keep the noise and interference out while combining these signals. ISDN
technically refers to a specific set of digital services provided through a single, standard interface.
6.4

ISDN STANDARDS

The OSI (Open Standard Interconnection) concept was developed for computer-to-computer communications.
Although ISDN was developed based on telephony network, its implementation requires the support from data
terminal communications to make non-voice service possible. The OSI model was adopted to develop a suite of
ISDN related standards. The standards also ensure interoperability and compatibility between equipment in a
multi-vendor environment.

49

Fig. 6.3 Mapping of ISDN Model

unto OS|I Reference Model

The Layer 1 characteristics of the user-network interface at S- and T-reference points (for the basic rate interface)
are defined in ITU-T I.430. Layer 1 characteristics of the user-network interface at the primary rate interface, are
defined in ITU-T I.431. The other two upper layers, Layer 2 and Layer 3, are defined to enable signalling to be
accomplished independently of the type of user-network interface involved. The characteristics of Layer 2 and
Layer 3 are specified in ITU-T Q.921 and Q.931 respectively.
6.4.1 ISDN Protocols
The ISDN protocols are signalling protocols that govern the exchange of data on the D channel. The two ISDN
signalling protocols make up a layered protocol stack, with the Link Access Protocol for the D Channel (LAPD,
also known as Q.921) providing Layer 2 data-link services and the Q.931 protocol providing higher-layer
services. LAPD is a simple, bit-oriented data-link protocol similar in structure and operation to HDLC and
SDLC. The Q.931 signalling protocol is one of the most complex and feature-rich communication protocols ever
designed.

6.5

ISDN INTERFACES AND FUNCTIONS

The ISDN Interface: The ISDN bit pipe supports multiple channels interleaved by time division multiplexing.
Several channel types have been standardized;

A - 4 kHz analog telephone channel

B - 64 kbps digital PCM channel for voice or data
C - 8 or 16 kbps digital channel for out-of-band signaling
D - 16 kbps digital channel for out-of-band signaling
E - 64 kbps digital channel for internal ISDN signaling
H - 384, 1536, or 1920 kbps digital channel.

It is not allowed to make arbitrary combination of channels on the digital pipe. Three combinations have been
standardized so far:

Basic rate: 2B + 1D. It should be viewed as a replacement for POTS (Plain Old Telephone Service). Each
of the 64 kbps B channels can handle a single PCM voice channel with 8 bits samples made 8000 times
per second. D channel is for signalling (i.e., to inform the local ISDN exchange of the address of the
destination). The separate channel for signalling results in a significantly faster setup time.

50

Primary rate: 23B + 1D (US and Japan) or 30B + 1D (Europe). It is intended for use at the T reference
point for businesses with a PBX.
Hybrid: 1A + 1C

Fig. 6.4 ISDN Bit Rates

(a) Basic rate digital pipe. (b) Primary rate digital pipe.

Because ISDN is so focused on 64 kbps channels, it is referred to as N-ISDN (Narrowband ISDN), in contrast to
broadband ISDN (ATM); B-ISDN.
6.5.1 PRI vs. BRI
In the ISDN, there are two levels of service: the Basic Rate Interface (BRI), intended for the home and small
enterprise, and the Primary Rate Interface (PRI), for larger users. Both rates include a number of B-channels and a
D-channel. Each B-channel carries data, voice, and other services. The D-channel carries control and signaling
information. The Basic Rate Interface consists of two 64 kbit/s B-channels and one 16 kbit/s D-channel. Thus, a
Basic Rate Interface user can have up to 128 kbit/s service. The Primary Rate Interface consists of 23 B-channels
and one 64 kbit/s D-channel using a T1 line (North American standard) or 30 B-channels and one D-channel using
an E1 line (Europe/Rest of World). Thus, a Primary Rate Interface user on a T1 line can have up to 1.544 Mbit/s
service or up to 2.048 Mbit/s service on an E1 line. PRI uses the Q.931 protocol over the D-channel.
6.5.2

Channel Types : BRI and PRI

There are two types of ISDN services, Basic Rate Interface(BRI), operating at 192 kbps, and Primary Rate
Interface (PRI), operating at 1.544 Mbps in North America or 2.048 Mbps in Europe. Both services support two
common kinds of channel types, the "B" (Bearer) and the "D" (Data).
1. BRI consist of two B channels at 64 kbps and one 16 kbps D channel, which is called 2B+D. The BRI has
144 kbps information carrying capacity.
2. PRI is designed to accommodate applications requiring more than two simultaneous connection and/or
data rate in excess of 64 kbps. The two distinct configurations are 23B+D structure at 1.544 Mbps and
30B+D running at 1.920 Mbps. In some cases more than one PRI connects between equipment.

Fig.6.5
ISDNUNI
Structure

51

6.5.3 Functions of ISDN Interfaces (Standard Reference Points)

The ISDN standards specify four distinct interfaces in the customer's connection to the network: R, S, T, and U.
From the standards viewpoint, these are not "real" physical interfaces, but simply STANDARD REFERENCE
POINTS where physical interfaces may be necessary. However, in common practice, the names of reference
points are used to refer to physical interfaces.
The R Interface: The interface at reference point R is the physical and logical interface between a non-ISDN
terminal device and a terminal adapter (TA). The R interface is not really part of the ISDN; it can conform
to any of the common telephone or data interface standards.
The S Interface: The interface at reference point S is the physical and logical interface between a TE (or TA)
and an NT. It uses four wires and employs a bipolar transmission technique known as Alternate Mark
Inversion (AMI). A special feature of the S interface is the "Short Passive Bus" configuration, which
allows up to eight ISDN devices (TE or TA) to contend for packet access to the D channel in a prioritized,
round-robin fashion. Only one device at a time can use a given B channel.
The T Interface: The interface at reference point T is the physical and logical interface between NT1 and NT2,
whenever the two NTs are implemented as separate pieces of hardware. The specification for the T
interface is identical to the specification for the S interface. In most implementations, NT1 and NT2 exist
in the same physical device, so there is no real T interface.
The U Interface: The interface at reference point U is the physical and logical interface between NT (or NT2)
and the ISDN carrier's local transmission loop. It is also the legal demarcation between the carrier's loop
and the customer's premises. The U interface is implemented with two wires and uses a special
quaternary signal format (i.e., four possible electrical states, with one pulse encoding a predefined
combination of 2 bits) called 2B1Q. Quaternary encoding allows the U interface to carry data with a
logical bit rate of 192 Kbps over a signal with a physical pulse rate of only 96 Kbps. The slower pulse rate
is better suited to the less-predictable environment of the outside-plant loop carrier system.
6.5.4

Interface Applications

The Primary Rate Interface channels are typically used by medium to large enterprises with digital PBXs to
provide them digital access to the PSTN and to the Digital Switched Network. The 23 (or 30) B-channels can be
used flexibly and reassigned when necessary to meet special needs such as videoconferences. The Primary Rate
user is hooked up directly to the telephone company central office.
6.5.5

ISDN Applications

ISDN in Business: For business users and even residential subscribers, videoconferencing is the biggest
communication advancement that ISDN has to offer. With the simultaneous high speed transfer of voice and
video, ISDN can provide real time video communication on a PC that once was only capable on sophisticated
systems costing upwards of $100,000. A shared electronic chalk board is another tool available through ISDN.
Ideas and illustrations can be distributed in real time to remote locations so people in other cities or other
countries can participate in meetings. Telecommuting is becoming a rule more than an exception; more and more
people are working from home. ISDN provides the facilities for users to tap into central network resources from
the privacy of their own homes and do so with the functionality of a network node. Node connections are possible
with Serial Line Interface Protocol (SLIP) and Point-to-Point Protocol (PPP).

52

ISDN in Education: Students will also reap the benefits of videoconferencing by relating with other students
worldwide. Using the video capabilities of ISDN allows students to see the surroundings of other countries or
speak with pen-pals. The value of videoconferencing in educational settings is unlimited. Computers have
become important learning tools for students. Children are introduced to computers and networking at an early
age, and ISDN allows the high speed connections to vast amounts of information and resources.
Security and Digital Networking: Security issues are a prime concern for digital transmissions. Although data
encryption schemes can alleviate the problem, other security issues prevail. E.g., in the National Security Agency
has approached the FCC with legislation that would make possible electronic surveillance of all digital
transmissions. The underlying reason stems from the possible use of the telecommunications system by criminal
organizations. Without a means to monitor encrypted network traffic, there could be no detection or intervention
of illegal activities taking place over the public network.
6.6

ISDN USER-NETWORK INTERFACE CONFIGURATIONS

6.6.1

USER-NETWORK INTERFACE

The ISDN is defined in the I-series Recommendations of the Standardization Bureau of ITU-T. It is a plan for
organising digital technology to provide advanced services to sophisticated digital terminals over an end-to-end
digital network. ISDN services are offered by the network to a user via an interface that provides either Basic
access, consisting of one 16 kbit/s D-channel and up to two 64 kbit/s B-channels, or Primary rate access,
consisting of one 64 kbit/s D-channel and up to 30 64 kbit/s B-channels. This Specification describes the Layer 2
requirements for the access protocol for the Basic Rate Access user-network interface between Telecom's ISDN
and a single unit of user's equipment, e.g., a terminal or small business system. This protocol applies at the T
reference point or the coincident S/T reference point and to both Basic (2B + D) and Primary rate (30B + D)
interfaces. See fig 6.1 for examples of interface configurations.
6.6.2

ISDN ACCESS INTERFACES

ITU-T I.412 defines different interface structures for ISDN user-network physical interfaces at the S- and T ISDN
reference points. These include:
Basic Rate Interface structure
Primary Rate B-channel interface structure
Primary Rate H-channel interface structure
Primary Rate Interface structures for mixtures of B and H0-channels
6.6.3

Basic Rate Interface Structure

A typical configuration for ISDN Basic Rate Access is shown below, illustrating the U- and S/T-interfaces. It is
composed of two B-channels and one 16 kbit/s Dchannel, i.e., 2B+D. The two B-channels may be used
independently. Typical BRI Circuit is shown below.
EXCHANGE

NT 1
CUSTOMER
PREMISES

LT

U
Fig. 6.6

BRI Interface

NT2

R/S/T INTERFACE 4 WIRES

TE
TE

CUSTOMER
PREMISES

53

A typical configuration for ISDN Basic Rate Access in reference to functional groups is shown above. A reference
point is often referred to as an interface. The various interfaces are:
U: Full-duplex 2-wire interface, using echo-cancellation technique between the NT1 and the LT for basic rate
ISDN. In most countries, a compression transmission line code called 2B1Q is used at this interface.
T: 4-wire interface between a NT1 and NT2
S: 4-wire interface connects an NT (or NT2) to a TE or TA
R: Non-ISDN interface between a non-ISDN compatible terminal and a TE2
6.6.4

Primary Rate B-Channel Interface Structure

A typical configuration for ISDN Primary Rate Access is shown below. This illustrates the use of E1 primary rate
connecting a PBX to the central office. E1 interface has a transmission rate of 2.048 Mbit/s and uses HDB3
Coding, PCM-31 framing. It is composed of thirty B-channels and one 64 kbit/s D channel, i.e., 30B+D.
All the 30 B-channels are always present at the user network interface, but the number of B-channels supported
by the network may be fewer. For multiple interfaces, the D-channel in one structure may carry signalling
information for B channels in another primary rate structure without an activated D-channel. The time slot for the
non activated D-channel may or may not be used to provide one additional B-channel over this structure.
S/T INTERFACE 4.WIRES
EXCHANGE
LT

PBX

TE

CUSTOMER
PREMISES

TE

E1 LINK
Fig. 6.7 Typical PRI Circuit
Primary Rate H-Channel Interface Structure: For primary rate at 2048 kbit/s, the 1920 kbit/s H12- channel
structure is defined. It is composed of one 1920 kbit/s H12-channel and one 64 kbit/s D-channel. In a multiple
interface arrangements, a single D channel may carry signaling information for channels
in another interface.
Primary Rate Interface Structures for Mixtures of B- and H0-Channels: This structure consists of one 64
kbit/s D-channel. In multiple interface arrangements, a single D channel may carry signaling information for
channels in another interface. Any mixture of B- and H0-channels
User-Network Interface Reference Points: LT: Line Termination; a device at the exchange office terminates an
ISDN circuit. NT1: Network Termination 1; a device at the customer premises (terminating an ISDN circuit)
performs physical layer functions such as signal conversion synchronization; converts 2-wire U-Interface to 4wire S/T Interface.
Physical Interfaces: An understanding of the format of interfaces and channel type is critical to any analysis of
ISDN because they provide the framework through which the protocols and applications flow. ISDN
defines a full network architecture as shown Fig. 1. This architecture separates access functions from
actual network functions.

54

6.7

ISDN NETWORK ARCHITECTURE

ISDN provides complete digital capabilities. Figure 1 shows the basic ISDN architecture, revealing the user
network interface and network capabilities, as well as the signaling system in the network. An ISDN user can
access the following services using an ISDN Terminal Equipment (TE):
Packet-switched data Circuit-switched data Circuit-switched voice User-to-user signaling
6.7.1

ISDN System Architecture

The key idea behind ISDN is that of the digital bit pipe between the customer and the carrier through which bits
flow in both directions. Whether the bits originate from a digital telephone, a digital terminal, a digital facsimile
machine, or some other device is irrelevant. The digital bit pipe can support multiple independent channels by
time division multiplexing of the bit stream. Two principal standards for the bit pipe have been developed:

6.7.2

A low bandwidth standard for home use, and

A higher bandwidth standard for business use that supports multiple channels identical to the home use
channels.
ISDN Configurations

Multiple Line Services

ISDN services can be supplied in three different configurations from the ISDN-ready digital switch to a business
or residence. The alternatives are:
1. Through a direct BRI connection from an ISDN switch. One or more BRI connections are made from the
central switching office to a business or home. These connections can be made directly to ISDN equipment, or
they can be connected through a PBX or key system. Using a PBX allows devices to communicate with one
another without having to make a connection outside the premises.
2. Through ISDN Centrex service. One or more BRI connections are made to ISDN Centrex service which offers
the advantage of having the ISDN switch function as the switching system. Therefore an individual or company
does not have to own a PBX or key system. Centrex service is provided at a low cost and provides virtual
unlimited growth.
3. Through a PRI connection. 23 B channels and one D channel is connected to a business through a PBX. The
PBX then provides the switching necessary within the organization. For heavy data traffic, an ISDN router,
multiplexer, or controller may be used instead of a PBX to reduce the chance of a bottleneck through the switch.
6.8

ISDN PROTOCOL ARCHITECTURE

Protocols and Frame Structure

Protocols in ISDN are based on the ISDN-PRM (ISDN-Protocol Reference Model) which is constructed
following the principles of the ISO/OSI-RM (International Standardization Organization/Open Systems

55

Interconnection Reference Model). The ISDN-PRM consists of two different planes: the user and the control
plane. These are shown below.

Fig. 6.8 ISDN Protocol

Reference Model

The coordination of these planes is made through a management function. Both the user and control plane could
incorporate a 7 layer protocol stack as OSI, but only the first 3 are defined until now. For the D-channel
definitions for layer 1-3 were made. For the circuit switched service on the B-channel, only the physical layer has
been specified. In this case the user is free to decide which protocol stack to use for the higher levels. For the
packet mode the layers 1-3, based on X.25 have been proposed.
6.8.1

ISDN Connections

ISDN Connections at the customer premises require Access to Network Exchanges, via recommended Interface
Structures and Protocols. The ITU-T Rec. I.412 defines different interface structures for ISDN user-network
physical interfaces at the S- and T ISDN reference points. These are:
Basic interface structure Primary rate B-channel interface structure Primary rate H-channel
interface structure Primary rate interface structures for mixtures of B and H0-channels.
All the above connections / interfaces are described in section 3.4 above.

6.9

BASIC ISDN ADDRESSING

6.9.1

ISDN Address Field

The ISDN Address Field is composed of Terminal Endpoint Identifier (TEI) and Service Access Point Identifier
(SAPI). The Address Field is broken out of the Q.921 Frame Format shown below The TEI identifies the user
device. A TEI may be assigned automatically or in a fixed manner, by the switch. Fixed TEIs are used in PRI or in
BRI point-to-point configurations. Automatic TEIs are generally used with multi-point BRI terminals. Here are
the values:
- 0 - 63: Non-automatic TEI assignment
- 64 - 126: Automatic TEI assignment

Fig. 6.9
ISDN
Address
Field

56

6.9.2

Protocol Q.921 Frame Format and Address Field

a)

Flag Field

The opening and closing flags are used for frame synchronization. The flags of a frame serve as a unique marker
to delimit its beginning and end and consist of the symmetric bit pattern 0 1 1 1 1 1 1 0. The closing flag may also
used as the beginning flag of the next frame. This unique bit pattern should not be existed in between the opening
and closing flag field. To avoid their occurrence in the remaining fields of the frame bit stuffing technique is used.
The data link layer entity at the sending side examines the content of the frame between opening and closing flags
during transmission and insert a 0 bit after every five consecutive 1 bit. The data link layer entity at the receiver
side inverts this process by discarding any 0 bit that directly follows five 1 bits in a sequence.
b)

Address Field

The opening flag of a frame is followed in octet 2 and 3 by an address field. In order to have each device support
multiple logical data links, the data link address is divided into the Service Access Point Identifier (SAPI) and the
Terminal End point Identifier (TEI). These two fields are together called Data Link Control Identifier (DLCI).
This provides a form of multiplexing. The six bits allocated to the SAPI allow the specification of up to 64
distinct service access point. Table 1. is representing the meaning according to the different SAPI values. The
SAPI identifies the network entity for which the information in the LAPD frame is intended. A LAPD entity may
have more than a single Layer 3 entity above it and therefore will have more than one SAPI in use. The TEI is
associated with the user side of the user-to network interface. The TEI identifies the logical terminal or final
destination for the Layer 3 information. One or more TEIs can be used for point to point data transfer. The TEI for
broadcast connection occurs when a message is transmitted with the TEI set to 127.

Table 6.10 LAPD: SAPI

6.10

ISDN LAYERS

ISDN Layers are based on or derived from the following listed International Networking 7 OSI Model.
These are: Physical Layer; Data Layer; Network Layer; Transport Layer Section LAYER; Presentation Layer;
Application Layer.

Fig.6.11
ISDN Layers

57

6.10.1 Physical Layer (ITU-T I.430, I.431)

The Physical Layer is responsible:
Encoding of digital data for transmission across the interface Full-duplex transmission of B-channel data
Full-duplex transmission of D-channel data Multiplexing of channels to form basic or primary access
transmission structure Activation and deactivation of the physical circuit Power feeding from network
termination to the Terminal Faulty terminal isolation D-channel contention access; this is needed when there is
a multi-point configuration for basic rate access.
6.10.2 Layer 2

DATA LINK LAYER - LAP-D (ITU-T I.441, Q.921)

Data Link Layer Protocol: LAPD

The LAPD (Link Access Protocol - Channel D) is a layer 2 protocol which is defined in CCITT Q.920/921.
LAPD works in the Asynchronous Balanced Mode (ABM). This mode is totally balanced (i.e., no master/slave
relationship). Each station may initialize, supervise, recover from errors, and send frames at any time. The
protocol treats the DTE and DCE as equals.

Fig. 6.12
LAPD
Protocol

The LAPD Flag and Control fields are identical to those of HDLC. The LAPD Address field can be either 1 or 2
bytes long. If the extended address bit of the first byte is set, the address is 1 byte; if it is not set, the address is 2
bytes. The first Address-field byte contains the service access point identifier (SAPI), which identifies the portal
at which LAPD services are provided to Layer 3. The C/R bit indicates whether the frame contains a command or
a response. The Terminal Endpoint Identifier (TEI) field identifies either a single terminal or multiple terminals. A
TEI of all ones indicates a broadcast. The format of a standard LAPD frame is as follows:

Flag

Address Field

Control Field

Information

FCS

Flag

Fig. 6.13 LAPD Frame

Flag: In order to ensure that the bit pattern of the frame delimiter flag does not appear in the data field of the
frame (and therefore cause frame misalignment), a technique known as Bit Stuffing is used by both the transmitter
and the receiver.

58

Address Field: The first two bytes of the frame after the header flag is known as the address field. The format of
the address field is as follows:
8

SAPI

C/R

EA1

TEIEA2
LAPD address field
EA1
Fig. 6.13 Address Field Format

First Address Extension bit which

is always set to 0.
C/R - Command/Response bit. Frames from the user
with this bit set to 0 are command frames, as are
frames from the network with this bit set to 1.
Other values indicate a response frame.
EA2 Second Address Extension bit which is always set
to 1.
TEI Terminal Endpoint Identifier.

Control Field: The field following the Address Field is called the Control Field and serves to identify the type of
the frame. In addition, it includes sequence numbers, control features and error tracking according to the frame
type.
FCS: The Frame Check Sequence (FCS) enables a high level of physical error control by allowing the integrity
of the transmitted frame data to be checked. The sequence is first calculated by the transmitter using an algorithm
based on the values of all the bits in the frame. The receiver then performs the same calculation on the received
frame and compares its value to the CRC.
6.10.3 Basic Function of LAPD Protocol
ISDN standards are constructed using the Open System Interconnection seven-layer reference model. Layer 2
(data link) protocol for the D channel(called Link Access Procedure-D) is used to convey messages over common
D channel. The LAPD and higher layer protocols handle the hands haking(commands and responses), signalling,
and control for all of the voice and data calls that are setup through the ISDN D channel. The LAPD protocol
evolved from LAPB of the CCITT x.25 protocol in order to serve multiple terminals on a single subscriber loop.
LAPD allows multiplexing of layer 2 connections on the same physical connection. Eight users can have
simultaneous D channel sessions. Each user is assigned a Logical Channel Number(LCN) and bandwidth is
divided accordingly. The objective of LAPD is to provide a secure, error-free connection between two end-points
so as to reliably transport Layer 3 messages. LAPD protocol provides framing, sequence control, error detection,
and recovery of multiple logical data links on the same D channel.
6.10.4 Network Layer
Layer 3 The Network Layer (ITU-T I.450, I.451, Q.931):

59

It defines the D-channel call control signalling (Ref. to the Basic Call Control Procedure). Specifies the
procedures for establishing connections on the B-channels that share the same interface to ISDN as the D-channel
Provides user-to-user control signaling over the D channel
Packet switching signaling is also available using X.25 Layer 3 protocol.
This is the same for using B channel packet switching service. Layer 3 provides higher layer information for
supporting various ISDN functions.
6.11 ISDN SERVICES
There are two types of services associated with ISDN:

a.

BRI

&

b.

PRI

6.11.1 ISDN BRI Service

The ISDN Basic Rate Interface (BRI) service offers two B channels and one D channel (2B+D). BRI B-channel
service operates at 64 kbps and is meant to carry user data; BRI D-channel service operates at 16 kbps and is
meant to carry control and signaling information, although it can support user data transmission under certain
circumstances. The D channel signaling protocol comprises Layers 1 through 3 of the OSI reference model. BRI
also provides for framing control and other overhead, bringing its total bit rate to 192 kbps.
The BRI physical layer specification is International Telecommunication Union-Telecommunications Standards
Section (ITU-T) (formerly the Consultative Committee for International Telegraph and Telephone [CCITT])
I.430. ISDN services are the telecommunication services to which the user has access either at an ISDN interface
or a terminal connected to the ISDN. The diagram below summarises the services offered:
Two different groups of services are supported by ISDN. These are: the bearer services and the Teleservices.

Fig. 6.14
ISDN Services

With ISDNs bearer services it is possible to transfer data between two subscribers. The network is acting in this
case as a bit pipe. The bearer service can be divided into packet switching (PS), with both connection orientated
and connectionless modes, and circuit switched connections.
6.11.2

ISDN Teleservices

a)

Telephony:

60

This service enables subscribers to make phone calls using the ISDN. It is a considerable improvement to the
(partially) analogue telephone-system, as it offers a better signal-noise ratio and the attenuation is unaffected by
distance. In the future it would also be possible to provide a greater voice bandwidth, e.g. 7kHz (now 3.1kHz)
and stereo sound, an important service attribute for audio conferencing.
Teletex: Texttransmission over ISDN is faster than the conventional teletex service over public data networks. It
is possible to transmit a page (legal size or OSI A4) in less than 1 second. The connection with
teletexterminals in other networks is supported. This includes circuit-switched as well as packetswitched networks.
Telefax (Telefax 4): ISDN supports the pixelorientated transmission of documents. CCITT has made
recommendations for telefax4 and group 4 facsimile machines with a resolution of 300dpi and
optionally 400dpi or 1200dpi. One page of legal size (or OSI A4) paper can be transmitted in 15
seconds when using the 400dpi resolution. "dpi" or 'dots per inch' refers to the output size of an image
when printed. An image at 300 dpi means that in every linear inch there are 300 dots.
Mixed Service: The mixed service allows a combination of the teletex and telefax service for the transmission of
one document. Thus it is possible to send a letter where the text is sent character coded and a picture
and/or signature is sent pixel coded. This can be seen as the first step in the direction of a multimedia
document transmission service in ISDN. A multimedia document designed to be transmitted over the
BRA interface, could consist of text, fax, still images with high resolution, graphics and voice. Video
information is provided with BISDN.
Still Image Transfer: This service offers the possibility of transmitting TV freeze frames over ISDN. A still
image sequence can be compressed and send with an update rate between 1 and 10 seconds, depending
on the contents of the image.
Videophony: Videophony is used to transmit moving pictures from person to person or person to group. Because
of the relatively low transmission rate (64kbps or 2x64kbps) the quality of the video frames are
inferior to TV frames (480x240 Pixel) and must be transmitted at a very high compression.
6.11.3

Distribution Services Allow Data Alarm Services:

It is possible to make emergency calls over ISDN, even if the power supply on the user side breaks down and the
B-channels can not be used any more. An emergency call is made over the D-channel which then is feeded by the
provider.
Messaging: This service provides a mailbox function for the user. Text and voice mail can be deposited in the
mailbox, if the user is not available or both of his B-channels are busy. The recipient is sent a message
from his mailbox over the d-Channel. He receives the notification even if his B-channels are used at
that time.
Videotex: Videotex is a retrieval service which enables the subscriber to view text and graphics based images.
Other Services: Some other services are: Telewriting for placing short messages (written with an electronic pen
on note pad) in a mailbox. Teleaction is used for controlling installations like heating, gas, water.
Transmission in an unidirectional way - but this service is much more interesting with the higher
transmission rates of BISDN (video and TV transmission).

61

6.11.4

ISDN Basic and Supplementary Services

ISDN Basic Services

The Basic ISDN Services are Telephony and Data:

Telephony: This service enables subscribers to make phone calls using the ISDN. It is a considerable
improvement to the (partially) analogue telephone-system, as it offers a better signal-noise ratio and the
attenuation is unaffected by distance. In the future it would also be possible to provide a greater voice bandwidth,
e.g. 7 kHz (now 3.1kHz) and stereo sound, an important service attribute for audio conferencing.
Data: This type of service enables customers to send and receive information via networked computers.
Supplementary Services
What are ISDN Supplementary Services? These are services that add value to existing ISDN services (Value
Added Services). They include: Calling line Identification; Call Waiting & Forwarding; Sub-addressing; Advice
of Charge; Multiple Subscriber Numbering; Direct Dial In; Etc.

6.12

ISDN SIGNALING SYSTEM

There are three different types of ISDN signaling. These are: a) User network Signaling,
b) Intra-network
Signaling, and c) User-to-user Signaling. All three employ common-channel signaling technique. User-network
signaling is used to control signaling between the user terminal equipment and the network. Intra-network
signaling is used to control signaling between ISDN switches. User-to-user signaling is used between the end
users and can be transparently transferred through the network. Instead of the phone company sending a ring
voltage signal to ring the bell in your phone ("In-Band signal"), it sends a digital packet on a separate channel
("Out-of-Band signal"). The Out-of-Band signal does not disturb established connections, no bandwidth is taken
from the data channels, and call setup time is very fast. For example, a V.90 or V.92 modem typically takes 30-60
seconds to establish a connection; an ISDN call setup usually takes less than 2 seconds. The signaling also
indicates who is calling, what type of call it is (data/voice), and what number was dialed. Available ISDN phone
equipment is then capable of making intelligent decisions on how to direct the call.
Layer 3 information is used for intra as well as Inter-exchange calls using SS#7 based on the Common Channel
Signaling System (CCS) which uses a dedicated network for Call Setup and Tear down in addition to Network
Management & Maintenance. The SS#7 is required for ISDN because it provides the basis for global
transparency of ISDN services.
Low Bandwidth v High Bandwidth: Bandwidth is a measure of the information-carrying capacity of a
particular form of transmission and it is usually denoted in bits per second (bit/s) where a bit is a binary digit (that
is, a zero / nought or a one in computer terms). Services with low information content such as basic voice
telephony only require low bandwidth and are called narrowband; services with more information content
such as video require mid bandwidth and are called midband; and services with a great deal of information
content such as high definition television require high bandwidth and are called broadband. Even a
narrowband service like lower-rate data might need midband transmission if the intention is to download large
volumes of data at high speed. Therefore the trend is for all communications networks to provide more and more
bandwidth and the challenge to network operators is how to provide this in a cost-effective way.

62

6.13

BROADBAND ISDN

6.13.1 Overview
What is Broadband ISDN (B-ISDN)?
According to ITU-T, B-ISDN is best described as a Telecommunications Service requiring transmission channels
capable of supporting rates greater than the primary rate.
Behind this statement lies the plan for a network and services that will have far more impact on the world we
know today, than ISDN ever would. The Narrowband ISDN (N-ISGN) had been designed to operate over
the current communications infrastructure, which is heavily dependent on the copper cable. B-ISDN
however, relies mainly on the evolution of fibre optics. B-ISDN allows its users to communicate over high
speed, high quality digital channels. The media it supports include: Fax, Voice Telephone, Video Telephone,
Audio, High Definition TV (HDTV), computer networking, etc.
B-ISDN Background: In the 1980s the telecommunications industry expected that digital services would
follow much the same pattern as voice services did on the PSTN and conceived a grandiose / elaborative
vision of end-to-end circuit switched services, known as the B-ISDN. This was designed in the 1990s as a
logical extension of the end-to-end circuit switched data service, ISDN. The technology for B-ISDN was
going to be Asynchronous Transfer Mode (ATM), which was intended to carry both synchronous voice
and asynchronous data services on the same transport. The B-ISDN vision which redesigned from the NISDN has been overtaken by the disruptive technology of the Internet. The ATM technology survives as a
low-level layer in most DSL technologies, and as a payload type in some wireless technologies such as
WiMAX. For this purpose, the B-ISDN is much faster than the N-ISDN.
B-ISDN is very strongly related to the Asynchronous Transfer Mode (ATM). This is because ATM provides a
consistent data encapsulation scheme that is used throughout the network, starting with the TE1 or TA equipment,
and covering every piece of telecommunications equipment in use. ATM is so important to B-ISDN that many
people believe it's the same thing.
Table 6.2 Required Data Rates & Number of DS0 Channels
Data Rates
Upstream
64 Kb/s

Downstream
64 Kb/s

No. of DS0 Channels

Upstream
Downstream
1
1

64 Kb/s

64 Kb/s & up

1-many

For file
transfers

NTSC Video

64 Kb/s

4-6 Mb/s

60-100

MPEG-2
compression

HDTV
Video

64 Kb/s

20-30 Mb/s

300-500

MPEG-2
compression

Functions
Voice
Data
Communication
s

Comments
Fixed rate

63

Multimedia

64 Kb/s 1.5 Mb/s

64 Kb/s

1-24

MPEG-4

For many data communication applications, the data stream is bursty and the use of dedicated channels is
inefficient. For low data rate applications, channel switching could be used with only a few dedicated DS0
channels. Applications involving the transfer of large amounts of data, e.g., large file transfers, require moderately
high data rates (compared to 150 Mb/s) for short periods of time.
These applications could be accommodated using ATM or channel switching. With channel switching, a
relatively large number of DS0 channels could be assigned to a particular connection for a short period of time
and released when the transfer is completed. The B-ISDN switching elements would be capable of switching
quickly, and the time required to change the number of assigned channels would be limited by propagation delays
rather than by switching times. The B-ISDN network could be designed to assign or release in a fraction of a
second.
Subscriber Connections:

Implementing B-ISDN requires deployment of fiber into the subscriber loop

multiplexing nodes would be placed between the CO and the subscribers, with each multiplexing node connected
to the CO by a single fiber that carries signals in both directions. To achieve the full STM-1 data rate, fiber must
be extended all the way to the subscriber premises. There are various ways for connecting subscribers to the
multiplexing node. One option would be to run a separate multimode fiber from the multiplexing node to each
subscriber. Multimode fiber is relatively inexpensive compared to single mode fiber and can support a data rate
of 155 Mb/s in both directions for distances up to a few kilometres. Another option would be to wavelength
multiplex signals for multiple subscribers on one single mode fibre.
6.13.2 B-ISDN Standards.
The ITU-T Recommendation I.363 covers the B-ISDN ATM Adaptation Layer (AAL) Specification. ATM has
been selected as the B-ISDN standard for transferring data. ATM is a form of fast packet switching where data
blocks are divided into small fixed size packets, called cells, that are transferred at a variable rate in available time
slots. Each ATM cell contains 53 bytes, which consist of a 5-byte header and a payload of 48 bytes. The ATM
cell was designed to be very small to accommodate voice traffic with minimum delays. For voice encoded at 64
Kb/s, it takes 6 ms to fill an ATM cell. This compares favourably with transmitting voice using IP, where the
packet sizes and the delays can be much larger. Protocol layer requirements: I.361 (11/95) B-ISDN ATM layer
specification and I.363 (03/93) B-ISDN ATM Adaptation Layer specification.
6.13.3

ATM Signaling for B-ISDN

Signalling is the process by which ATM users and the network exchange the control of information, request the
use of network resources, or negotiate for the use of circuit parameters. The Q.2931 is ITU version of signalling
protocol. Q.2931 specifies the procedures for the establishment, maintenance and clearing of network connections
at the B-ISDN user network interface. The procedures are defined in terms of messages exchanged.
6.13.4

Broadband Services

64

There are various types of services on telecommunications networks. Here, we examine examine several types of
multimedia services and subscriber network architectures from the service nodes to the terminals on the B-ISDN
network.
6.13.5

Interactive Services

Broadband Video Telephony Services: Video telephony is the transfer of voice, moving pictures and scanned
images and documents between two points. Areas utilising such technology are sales, consulting, teaching
and legal services.
The problem with gaining widespread use of video telephony is the prohibitive costs of terminal equiptment. In
the future as demand and competition increases the cost of such equipment will fall, and the service will
become more widespread. Other services that are expected to be implemented using video telephony
include video conferencing and video surveillance.
High Speed Unrestricted Data and Information Transmission Services: These types of service include LANs
and WANs, as well as internet and other computer networking. Other Applications include document
transfers, facsimile and multimedia documents, text, graphics, voice, and audio visual information.
Messaging Services: Messaging services is the transmission of information on a user to user basis, but not
requiring the availability of both users at once. Due to this area consisting mainly of text transfers it
doesn't take much of the available resources. The application of this is primarily email, paging services
and many more.
6.13.6 Broadband Architecture
The B-ISDN control architectures and protocols have been defined by standardization bodies to provide switched
services over the emerging ATM transport networks. The first releases of such architectures provide almost the
same services as those provided by the N-ISDN but with increased bandwidth. In order to fully exploit the
transport capabilities offered by ATM a versatile control architecture that could support rapid integration of new
services into the existing control infrastructure, is desired; the method of providing such an evolution is leading to
an integration of IN concepts into the B-ISDN architecture.
6.13.7

B-ISDN Architecture and Operation

The B-ISDN architecture, shown below is similar to the N-ISDN architecture.

Fig. 6.15
B-ISDN
Architecture

65

Implementation of this B-ISDN architecture requires the following:

1) Upgrading of subscriber interfaces;
2) Conversion of N-ISDN switches to B-ISDN switches;
3) Increasing the capacity of the transmission lines interconnecting the switches,
4) Replacing IP/LAPD packet switches with ATM switches.
B-ISDN and N-ISDN switches operate in a similar manner. Both are based on switching of time slots within a
125 s frame, T-1 or SONET frames for N-ISDN and STM frames for B-ISDN. The STM-1 frame corresponds
to the third order SONET frame and the n th order STM frame corresponds to a SONET frame of order 3n.
Consequently, switching operations and switch architectures are similar for N-ISDN and B-ISDN switches.
Similarly, fibre optic transmission lines interconnecting the switches can be similar. Of course, the B-ISDN lines
must have a significantly higher capacity.
6.13.8 B-ISDN Protocol Reference Model
The architecture for the B-ISDN protocol is as shown below. The B-ISDN protocol uses a three plane approach.
These three separate planes are referenced as:
a) User Plane

b) Control Plane

c) Management Plane
Management plane

Control plane

Fig. 6.16 B-ISDN Protocol

Reference Model

Higher Layers

User plane
Higher Layers

ATM Adaptation Layer

ATM Layer

Virtual Channel Functions

Plane management

Virtual Path Functions

Physical Layer (PMD)

Layer management

The user plane is responsible for user information transfer including flow control and error control. The control
plane manages the call-control and connection-control functions, while the management plane includes plane
management, and layer management.
The protocol for B-ISDN also adopts a layered approach, made up of four layers:
d) Physical Layer

b) ATM Layer

c) ATM Adaptation Layer (AAL) d) Higher Layers.

The higher layers provide services for video, SMDS, Frame Relay, Access & Network Signaling. ATM is often
referred to as fast packet switching. For this reason, B-ISDN is a packet based network. B-ISDN handles both
packet and circuit-mode applications, and this is executed by the AAL. The AAL also handles non-ATM protocol,
such as Link Access Protocol-D.
The ATM layer provides the packet transfer capabilities, while the physical layer provides the base network
functions- physical connections of network devices. ATM is often refereed to as fast packet switching network,
thus, the B-ISDN is a packet based network. The ITU-T Recommendations and Standards state that B-ISDN
handles both packet and circuit-mode applications. This is executed by the AAL which also handles non-ATM

66

protocols, such as Link Access Protocol-D (LAPD). The ATM layer provides the packet transfer capabilities,
while the physical layer provides the base network functions.

Table 6.3 B-ISDN Protocol Model and Architecture

Layer Name

Function
Higher Layer Function

Higher Layers
A
A
Convergence
L
Sub-layer (CS)
SAR Sub-layer

Common Part (CP)

Segmentation and Re-assembly
Generic Flow Control
Cell Header Generation/Extraction
Cell VCI/VPI Translation
Cell Multiplexing/
De-multiplexing

ATM

P
H
Y

Service Specific (SS)

Transmission
Convergence (TC)

Cell Rate Decoupling

Cell Delineation
Transmission Frame Adaption

PMD

Bit Timing, Phy. Medium

Physical Medium Dependent (PMD) Sub-layer provides actual clocking of bit transmission over the physical
medium
6.13.9 B-ISDN Physical Layer.
The B-ISDN Physical Layer is responsible for basic physical layer activities such as information synchronization,
bit timing, transmission frame generation and recovery, transmission frame adaptation, cell delineation, HEC
sequence generation and cell header verification, as well as cell rate decoupling.
The Fig below represents the possible configurations users can adopt when connecting to a B-ISDN network. NT
represents the network terminating equipment. NT1 is the interface between the users phone system and the BISDN network provider. NT2 will exist when the users have internal phone systems, i.e. this is the companies
private telephone exchange. The TA is a device used to connect non ISDN equipment to a B-ISDN network. The
TE is the terminal equipment: type 1 is ISDN hardware, type 2 is not.

67

Fig. 6.17 User Access

Configuration at the
Physical Layer level

The S transmission line is used to multiplex several TE devices. There are currently three options for
transmission:
These options need to be taken into consideration in the physical layer. These functions are handled by two
sublayers the physical medium sublayer and the transmission convergence sublayer. The physical medium
sublayer handles the medium dependent processes, while the transmission convergence sublayer handles the
remaining functions.
6.13.10 Applications of B-ISDN
Most of the applications for ISDN have reached the extent of their development, and now the focus has shifted to
services that can be provided across broadband ISDN cables. The ITU-T defines the services and associated
standards of ISDN communications, and have recommended the two service areas for application with BISDN,
Interactive Services, and Distribution Services.
Table 6.4 B-ISDN Service Categories

Type of Service

SERVICE CATEGORY

EXAMPLE SERVICE

Conventional Services

TV Services

Messaging Services

Video Mail

Retrieval Services

Videotex

Without User Presentation Control

TV Broadcast

With User Presentation Control

Videography

Internal Services

Distribution Services

a)
i)

Interactive Services
Broadband video telephony services

Video telephony is the transfer of voice, moving pictures and scanned images and documents between two points.
Areas utilising such technology are sales, consulting, teaching and legal services. The problem with gaining
widespread use of video telephony is the prohibitive costs of terminal equiptment. In the future as demand and
competition increases the cost of such equiptment will fall, and the service will become more widespread. Other
services that are expected to be implemented using video telephony include video conferencing and video
surveillance.
ii)

High speed unrestricted data and information transmission services

68

This type of service will include LAN's(Local Area Networks), and WAN's (Wide Area Netowrks), as well as
internet and other computer networking. Other Applications include document transfers, facsimile and multimedia
documents including text, graphics, voice, and audio visual information.
Messaging Services: Messaging services is the transmission of information on a user to user basis, but not
requiring the availability of both users at once. Due to this area consisting mainly of text transfers it
doesn't take much of the available resources. The application of this is primarily email, but could be
expanded in the future to include paging services and more.

Retrieval Services: Retrieval services as the name suggests involves retrieval of information stored at remote
sites. This information would be available at public sites, and supplied to the user on a demand basis.
Items transferred in this way could include medical information, share market information and
transmission of audio, and video files. Unfortunately the amount of bandwidth required for transferring
such resources restricts the number of transmissions that can occur at any one time.
Distribution Services: Distribution services are divided into distribution with presentation control, and
distribution without presentation control. Distribution services provide a continuous flow of information
from a central source to any number of users. All of the users have access to the information but not control
over it. This type of service includes TV program distribution, and document distribution.
TV program distribution (without presentation control): TV program distribution is the most common
application within distribution services. With the capacities of broadband ISDN, higher quality, higher
resolution, interference free television can be provided. This quality should be equal to that provided in
cinemas, but will require transfer rates around 1 Gbps. ITU-T have suggested that data compression be
used to reduce the bit-rate requirements and enhance co-existence with multiple broadband ISDN
services.
Distribution With Presentation Control: This type of distribution is centrally located but information will be
transmitted in cycles. The user has individual access to the cyclical distributed information, but unlike TV
program distribution the user has control over the start and order of presentation. Applications of such
systems would be in education training and other services where it is essential for users to be able to
control the time of access.

69

CHAPTER 7

7.1

FRAME RELAY NETWORK

OVERVIEW - WHAT IS FRAME RELAY SERVICE?

Frame Relay is a high-performance WAN protocol that operates at the physical and data link layers of the OSI
reference model. Frame Relay originally was designed for use across Integrated Services Digital Network (ISDN)
interfaces. Today, it is used over a variety of other network interfaces as well. Frame Relay is a
telecommunications digital connection-oriented data service that sends packets of data, called frames, over the
network. This frame of data is transmitted through the network and checked for errors. Frame Relay is designed
for cost-efficient data transmission for intermittent traffic between local area networks (LAN -to- LAN) and
between end-points in a wide area network (WAN). It's a data network service bundled with leased line access for
transmitting data between remote networks.
Frame relay puts data in a variable-size unit called a frame and leaves any necessary error correction
(retransmission of data) up to the end-points, which speeds up overall data transmission. For most services, the
network provides a permanent virtual circuit (PVC), which means that the customer sees a continuous, dedicated
connection without having to pay for a full-time leased line, while the service provider figures out the route each
frame travels to its destination and can charge based on usage.
7.2

FRAME RELAY BACKGROUND

Frame Relay is a protocol standard for LAN internetworking which provides a fast and efficient method of
transmitting information from a user device to LAN bridges and routers. The Frame Relay protocol uses a frame
structured similar to that of LAPD, except that the frame header is replaced by a 2-byte Frame Relay header field.
The Frame Relay header contains the user-specified DLCI field, which is the destination address of the frame. It
also contains congestion and status signals which the network sends to the user. It is a communications
networking protocol that defines how frames are routed through a fast-packet network based on the address field
in the frame.
Why was Frame Relay Developed?
From the beginning, frame relay was embraced enthusiastically by users because it was developed in response to
a clear market need, namely the need for high speed and high performance transmission. Frame relay technology

70

also made cost-effective use of widespread digital facilities and inexpensive processing power found in end user
devices. Developed by and for data communications users, frame relay was simply the right technology at the
right time. Let's explore the network trends that contributed to the development of frame relay.
Close to the end of the 1980's, several trends combined to create a demand for and enable higher speed
transmission across the wide area network. These trends included:
The change from primarily text to graphics interaction
The increase in "bursty" traffic applications
Intelligent end-user devices (PCs, workstations, X-Windows terminals) with increased computing power
The proliferation of LANs and client/server computing
Widespread digital networks
Frame Relay origins
Frame relay began as a stripped-down version of the X.25 protocol, releasing itself from the error-correcting
burden most commonly associated with X.25. When frame relay detects an error, it simply drops the offending
packet. Frame relay uses the concept of shared-access and relies on a technique referred to as "best-effort",
whereby error-correction practically does not exist and practically no guarantee of reliable data delivery occurs.
Frame relay provides an industry-standard encapsulation utilizing the strengths of high-speed, packet-switched
technology able to service multiple virtual circuits and protocols between connected devices.
Frame Relay Technology Basics
Frame Relay provides a packet-switching data communications capability that is used across the interface
between user devices (for example, routers, bridges, host machines) and network equipment (for example,
switching nodes). User devices are often referred to as Data Terminal Equipment (DTE), while network
equipment that interfaces to DTE is often referred to as Data Circuit-terminating Equipment (DCE). The network
providing the Frame Relay interface can be either a carrier-provided public network or a network of privately
owned equipment serving a single enterprise.
As an interface to a network, Frame Relay is the same type of protocol as X.25. However, Frame Relay differs
significantly from X.25 in its functionality and format. In particular, Frame Relay is a more streamlined protocol,
facilitating higher performance and greater efficiency.
7.3
7.3.1

ADVANTAGES AND DISADVANTAGES OF FRAME RELAY

Advantages

The main advantage of Frame Relay over point-to-point leased lines is cost. Frame Relay can provide
performance similar to that of a leased line, but with significantly less cost over long distances. The reason is
because the customer only has to make a dedicated point-to-point connection to the provider's nearest frame
switch. From there the data travels over the provider's shared network. The price of leased lines generally
increases based on distance. So, this short-haul point-to-point connection is significantly less expensive than
making a dedicated point-to-point connection over a long distance. Frame Relay offers an attractive alternative to
both dedicated lines and X.25 networks for connecting LANs to bridges and routers.
7.3.2

Disadvantages

The two main disadvantages of Frame Relay are slow downs due to network congestion and difficulty ensuring
Quality of Service (QoS). Because all of a provider's Frame Relay customers use a common network there can be

71

times when data transmission exceeds network capacity. The difficulty ensuring QoS is due to the fact that Frame
Relay uses variable-length packets. It is easier to guarantee QoS when using a fixed-length packet.
7.4

FRAME RELAY PROTOCOL ARCHITECTURE

The protocol stack of frame relay is a very simple one. Only one and a half layers of the OSI Model are used. The
protocol layers used are the Physical Layer and a subset of the Data Link Layer, called LAPF core (LAPF = Link
Access Procedures / Processes to Frame Mode Bearer Services) which is defined in Q.922. LAPF is based on and
is an extension of LAPD, which is used in ISDN.

Comparing Protocol Stack of OSI, ISDN and Frame Relay

Fig. 7.1 Frame Relay OSI Model

The Physical Layer is no different from any network Physical Layer with definitions of how bits are transmitted.
The Data Link Layer on the other hand provides some of the same functions as defined in OSI model such as
framing, addressing and bit error detection. The difference is that there is no sequencing or no acknowledgements.
In case of errors the frame is simply discarded. Since frame relay has no error correction it assumes that the
network infrastructure is relatively error-free (e.g. fiberglass). Multiplexing is also performed at the Data Link
Layer.
7.5

HOW DOES FRAME RELAY WORK?

Frame Relay is a digital packet network service that provides all the features and benefits of a dedicated network
service without the expense of multiple dedicated circuits. This data is carried in the form of packets and given an
ID on a per packet basis. It is then sent across the network in a very efficient way. In a Frame Relay network,
circuits are connected to a packet switch within the network that ensures that packets are routed to the correct
location.
Frame Relay places data in variable-size units called "frames" and leaves any necessary error-correction (such as
re-transmission of data) up to the end-points. This speeds up overall data transfer. For most services, the network
supplies a Permanent Virtual Circuit (PVC), which implies that the customer benefits from a continuous,
dedicated connection without having to pay for a permanent leased line, while the Frame Relay Service Provider
sorts out the path each frame voyages to its target and can charge based on usage. The basic flow of data in a
frame relay network can best be described in a few key steps:

Information is sent over the WAN using a data link connection identifier (DLCI), which specifies the
frame's destination
The network discards the frames if there is a problem due to line errors or congestion
Error correction and lost frames are retransmitted by the end user hardware devices at higher-level
protocols than handled by the frame cloud

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7.6

Good performances are achieved by low error rates

FRAME RELAY DEVICES

Devices attached to a Frame Relay WAN fall into the following two general categories:

Data terminal equipment (DTE)

Data circuit-terminating equipment (DCE)

DTEs generally are considered to be terminating equipment for a specific network and typically are located on the
premises of a customer. In fact, they may be owned by the customer. Examples of DTE devices are terminals,
personal computers, routers, and bridges.
DCEs are carrier-owned internetworking devices. The purpose of DCE equipment is to provide clocking and
switching services in a network, which are the devices that actually transmit data through the WAN. In most
cases, these are packet switches.
The Figure shows the relationship between the two categories of devices.

Fig. 7.2 DTE v DCE

The connection between a DTE device and a DCE device consists of both a physical layer component and a link
layer component. The physical component defines the mechanical, electrical, functional, and procedural
specifications for the connection between the devices.
One of the most commonly used physical layer interface specifications is the recommended standard (RS)-232
specification. The link layer component defines the protocol that establishes the connection between the DTE
device, such as a router, and the DCE device, such as a switch. This chapter examines a commonly utilized
protocol specification used in WAN networking: the Frame Relay protocol.

7.7

FRAME RELAY CONNECTIONS - VIRTUAL CIRCUITS

Frame Relay provides connection-oriented data link layer communication. This means that a defined
communication exists between each pair of devices and that these connections are associated with a connection
identifier. This service is implemented by using a Frame Relay virtual circuit, which is a logical connection
created between two data terminal equipment (DTE) devices across a Frame Relay packet-switched network
(PSN). Virtual circuits provide a bidirectional communication path from one DTE device to another and are
uniquely identified by a Data-Link Connection Identifier (DLCI). A number of virtual circuits can be multiplexed
into a single physical circuit for transmission across the network. This capability often can reduce the equipment
and network complexity required to connect multiple DTE devices. A virtual circuit can pass through any

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number of intermediate DCE devices (switches) located within the Frame Relay PSN. Frame Relay virtual
circuits fall into two categories: switched virtual circuits (SVCs) and permanent virtual circuits (PVCs).
7.7.1

Switched Virtual Circuits

Switched virtual circuits (SVCs) are temporary connections used in situations requiring only sporadic data transfer
between DTE devices across the Frame Relay network. A communication session across an SVC consists of the
following four operational states:

Call setupThe virtual circuit between two Frame Relay DTE devices is established.

Data transferData is transmitted between the DTE devices over the virtual circuit.

IdleThe connection between DTE devices is still active, but no data is transferred. If an SVC remains in an
idle state for a defined period of time, the call can be terminated.

Call terminationThe virtual circuit between DTE devices is terminated.

After the virtual circuit is terminated, the DTE devices must establish a new SVC if there is additional data to be
exchanged.
It is expected that SVCs will be established, maintained, and terminated using the same signaling protocols used
in ISDN. Few manufacturers of Frame Relay DCE equipment support switched virtual circuit connections.
Therefore, their actual deployment is minimal in today's Frame Relay networks. Previously not widely supported
by Frame Relay equipment, SVCs are now the norm. Companies have found that SVCs save money in the end
because the circuit is not open all the time.
7.7.2

Permanent Virtual Circuits

Permanent virtual circuits (PVCs) are permanently established connections that are used for frequent and
consistent data transfers between DTE devices across the Frame Relay network. Communication across a PVC
does not require the call setup and termination states that are used with SVCs. PVCs always operate in one of the
following two operational states:

Data transferData is transmitted between the DTE devices over the virtual circuit.

IdleThe connection between DTE devices is active, but no data is transferred. Unlike SVCs, PVCs will not
be terminated under any circumstances when in an idle state. DTE devices can begin transferring data whenever
they are ready because the circuit is permanently established.
7.7.3

Data-Link Connection Identifier

Frame Relay virtual circuits are identified by Data-Link Connection Identifiers (DLCIs). DLCI values typically
are assigned by the Frame Relay service provider (for example, the telephone company). Frame Relay DLCIs
have local significance, which means that their values are unique in the LAN, but not necessarily in the Frame
Relay WAN.
7.7.4

Data Link Connection Identifier Number

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DLCI Number is a channel number which is attached to data frames to tell a Frame Relay network how to route
the data. In Frame Relay, multiple logical channels are multiplexed over a single physical channel. The DLCI says
which of these logical channels a particular data frame belongs to. It is the number of a private or switched
virtual circuit in a frame relay network. Located in the frame header, the DLCI field identifies which logical
circuit the data travels over, and each DLCI has a Committed Information Rate (CIR) associated with it. The
DLCI number is local to the FRAD and frame relay switch it connects to, and it is generally changed by the
switch within the network, because the receiving switch uses a different DLCI for the same connection.

7.7.5

Frame Relay Access Device (FRAD)

FRAD (Frame Relay Access Device or Frame Relay Assembler/Disassembler) is a device that turns data packets
into frame relay frames that can be sent over Frame Relay network and turns the received Frame Relay frames
into data packets. Its assembly/disassembly functionality is similar to a PAD or Packet Assembler/Disassembler,
which is used for accessing X.25 networks.
7.7.6 Frame Relay Network Implementation
A common private Frame Relay network implementation is to equip a T1/E1 multiplexer with both Frame Relay
and non-Frame Relay interfaces. Frame Relay traffic is forwarded out the Frame Relay interface and onto the data
network. Non-Frame Relay traffic is forwarded to the appropriate application or service, such as a private branch
exchange (PBX) for telephone service or to a video-teleconferencing application.
A typical Frame Relay network consists of a number of DTE devices, such as routers, connected to remote ports
on multiplexer equipment via traditional point-to-point services such as T1, fractional T1, or 56-Kb circuits. The
majority of Frame Relay networks deployed today are provisioned by service providers that intend to offer
transmission services to customers. This is often referred to as a public Frame Relay service. Frame Relay is
implemented in both public carrier-provided networks and in private enterprise networks. The following section
examines the two methodologies for deploying Frame Relay.
7.8

FRAME RELAY NETWORK

A frame relay network consists of endpoints (e.g., PCs, servers, host computers), frame relay access equipment
(e.g., bridges, routers, hosts, frame relay access devices) and network devices (e.g., switches, network routers,
T1/E1 multiplexers). Accessing the network using a standard frame relay interface, the frame relay access
equipment is responsible for delivering frames to the network in the prescribed format. The job of the network
device is to switch or route the frame through the network to the proper destination user device.

Fig. 7.3
Frame Relay Network

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A frame relay network will often be depicted as a network cloud, because the frame relay network is not a single
physical connection between one endpoint and the other. Instead, a logical path is defined within the network.
This logical path is called a virtual circuit. Bandwidth is allocated to the path until actual data needs to be
transmitted. Then, the bandwidth within the network is allocated on a packet-by-packet basis. This logical path is
called a virtual circuit. Frame Relay Network offers high throughput, reliability and flexibility that is perfect for a
variety of today's business applications. The simple packet switching technology is well suited for powerful PCs,
workstations and servers using protocols such as TCP/IP protocols.
A frame relay network is often depicted as a cloud or frame cloud because of the nature of a frame relay
configuration and the many layers between the connections. Frame relay consists of logical paths within the cloud
called virtual circuits. It appears to the user as a dedicated point-to-point circuit, but in fact is a logical path. A
frame relay network is based on virtual circuits which may be meshed or point-to-point, and these may be
permanent or switched connections.
7.9

FRAME RELAY APPLICATIONS

Frame relay was developed in the 1990s to provide solutions for several trends in the network, including;
The change from mostly text to graphic interaction
Increase in "bursty" traffic
End-user devices become more sophisticated (PCs, Workstations, Routers and WAN devices)
Sophisticated operating systems (Windows, Solaris, Novell OS, etc.)
LAN environments
Widespread digital networks
Voice, video and data integration
These types of applications require high bandwidth, followed by idle time on the network.
"Bursty" traffic is a characteristic well suited for a "shared" network like frame relay. Frame relay also
solves the trends for higher bandwidth and increased speed over the WAN.
7.10

FRAME-MODE CALL CONTROL

The steps for frame relay call control are as follows:

Establish a logical connection between two endpoints, and assign a unique DLCI to the connection
Exchange information in data frames. Each frame includes a DLCI field to identify the connection
Release the logical connection when call is over.

The establishment and release of a frame relay connection is accomplished by the exchange of Q.931/Q.933
messages over a logical connection dedicated to call control (DLCI = 0). A frame with DLCI=0 has a call control
message in the information field. Either side may request the establishment of a logical connection by sending a
SETUP message. The other side must reply either with a CONNECT message to accept the connection or with a
RELEASE COMPLETE message to refuse the connection. The side sending the SETUP message may choose the
DLCI for the connection by choosing an unused value; otherwise, the accepting side assigns the DLCI within the
CONNECT message. After the connection is established, data transfer can proceed. Clearing a connection is
accomplished by the exchange of DISCONNECT, RELEASE, and RELEASE COMPLETE messages.

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7.11

FRAME RELAY CONGESTION CONTROL

7.11.1 What is Congestion?

Congestion is a problem that occurs on shared networks when multiple users contend for access to the same
resources (bandwidth, buffers, and queues). Think about freeway congestion. Many vehicles enter the freeway
without regard for impending or existing congestion. As more vehicles enter the freeway, congestion gets worse.
Eventually, the on-ramps may back up, preventing vehicles from getting on at all.

Congestion typically occurs where multiple links feed into a single link, such as where internal LANs are
connected to WAN links. Congestion also occurs at routers in core networks where nodes are subjected to more
traffic than they are designed to handle. TCP/IP networks such as the Internet are especially susceptible to
congestion because of their basic connection- less nature. There are no virtual circuits with guaranteed bandwidth.
Packets are injected by any host at any time, and those packets are variable in size, which make predicting traffic
patterns and providing guaranteed service impossible. While connectionless networks have advantages, quality of
service is not one of them.
Two Congestion Control strategies supported in frame relay are:

Congestion avoidance procedures are used at the onset of congestion to minimize the effect on
the network
Congestion recovery procedures are used to prevent network collapse in the face of severe
congestion

Congestion Control in frame-relay networks includes the following mechanisms:

1. Admission Control. This provides the principal mechanism used in frame relay to ensure the guarantee of
resource requirement once accepted.
2. It also serves generally to achieve high network performance. The network decides whether to accept a
new connection request, based on the relation of the requested traffic descriptor and the network's residual
capacity. The traffic descriptor consists of a set of parameters communicated to the switching nodes at call
set-up time or at service-subscription time, and which characterizes the connection's statistical properties.
The traffic descriptor consists of three elements:
i.

Committed Information Rate (CIR). The average rate (in bit/s) at which the network guarantees to transfer
information units over a measurement interval T. This T interval is defined as: T = Bc/CIR.

ii. Committed Burst Size (BC). The maximum number of information units transmittable during the interval
T.
iii. Excess Burst Size (BE). The maximum number of uncommitted information units (in bits) that the
network will attempt to carry during the interval.
7.11.2

Congestion-Control Mechanisms

Frame Relay reduces network overhead by implementing simple congestion-notification mechanisms rather than
explicit, per-virtual-circuit flow control. Frame Relay typically is implemented on reliable network media, so data

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integrity is not sacrificed because flow control can be left to higher-layer protocols. Frame Relay implements two
congestion-notification mechanisms:

Forward-Explicit Congestion Notification (FECN)

Backward-Explicit Congestion Notification (BECN)

FECN and BECN each is controlled by a single bit contained in the Frame Relay frame header. The Frame Relay
frame header also contains a Discard Eligibility (DE) bit, which is used to identify less important traffic that can
be dropped during periods of congestion.

7.11.3 Frame Relay Discard Eligibility

The Discard Eligibility (DE) bit is used to indicate that a frame has lower importance than other frames. The DE
bit is part of the Address field in the Frame Relay frame header. DTE devices can set the value of the DE bit of a
frame to 1 to indicate that the frame has lower importance than other frames. When the network becomes
congested, DCE devices will discard frames with the DE bit set before discarding those that do not. This reduces
the likelihood of critical data being dropped by Frame Relay DCE devices during periods of congestion.
7.11.4 Frame Relay Error Checking
Frame Relay uses a common error-checking mechanism known as the cyclic redundancy check (CRC). The CRC
compares two calculated values to determine whether errors occurred during the transmission from source to
destination. Frame Relay reduces network overhead by implementing error checking rather than error correction.
Frame Relay typically is implemented on reliable network media, so data integrity is not sacrificed because error
correction can be left to higher-layer protocols running on top of Frame Relay.

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CHAPTER 8
ASYNCHRONOUS TRANSFER MODE (ATM)
8.1

Overview

8.1.1

What is ATM?

ATM is a cell relay, packet switching network and data link layer protocol which encodes data traffic into
small (53 bytes; 48 bytes of data and 5 bytes of header information) fixed-sized cells. ATM provides data link
layer services that run over Layer 1 links. This differs from other technologies based on packet-switched
networks (such as the Internet Protocol or Ethernet), in which variable sized packets (known as frames when
referencing to Layer 2) are used. ATM is a connection-oriented technology, in which a logical connection is
established between the two endpoints before the actual data exchange begins.
Circuit switching has the benefits of supporting high speed data as well as low and constant transmission
delays. Packet switching, on the other hand, is efficient for bursty traffic since the bandwidth is used only when
there is something to transmit. ATM is a cell-relay (NOTE: cell = packet, relay = switch, so it is a packet
switching) technology with statistical multiplexing. It is a protocol that transmits data as fixed sized packets.
This is the culmination of all the developments in switching and transmission of data in recent years.
8.1.2

ATM Design

ATM is a technology designed for the high-speed transfer of voice, video, and data through public and private
networks using cell relay technology. ATM is an International Telecommunications

Union -

Telecommunications Standardization Sector (ITU-T) Standard. The Standards for ATM were first developed in
the mid 1980s. The goal was to design a single networking strategy that could transport real-time video and
audio as well as image files, text and email. Two groups, the ITUT and the ATM Forum were involved in the
creation of the standards. ATM has been used primarily with telephone and IP networks.

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ATM was designed to make Broadband-ISDN (B-ISDN) a reality. B-ISDN was created conceptually as just an
extension of ISDN. It therefore functions as a communication network that can provide integrated broadband
services such as high-speed-data service, video phone, video conferencing, CATV services along with
traditional ISDN services such as phone and text.
The design intent is that ATM has the benefits of both circuit switching and packet switching.

8.1.3

ATM is Faster than X.25, more streamlined than frame relay

It supports data rates several orders of magnitude greater than frame relay
Data on logical connection is organized into fixed-size packets, called cells.
No link-by-link error control or flow control.

What Motivated the Development of ATM?

The emergence of fiber technology offered a transmission capacity that could easily handle high bit rates. This
led to the development of networks that can integrate all types of information services. As a result, ATM was
created in order to provide a network that is capable of handling data, voice, video, and image applications
independently of their bandwidth requirements. ATM was the standard developed to provide B-ISDN services,
that is, the next generation of ISDN services.
Before the ATM era, each application required its own type of network, mainly because different services
needed different requirements on the transmission medium. In order to meet these requirements, different
technologies were employed. This has yielded Time Division Multiplexing (TDM) and a vast range of
protocols, mainly based on the use of variable length packets (such as X.25, Frame Relay, IP). ATM enables
merging pure data networks and pure telephone networks into a single entity.
It is based on B-ISDN standard which was developed by ITU-T. Originally, it was supposed to transmit voice,
video and data through public networks only, but the ATM Forum made it possible for private networks, too.
8.1.4

Why ATM?

ATM was developed:

To support any type of traffic:
- Burtsy data (to multimegabit rates: files, images, multimedia)
- Intermittent data (interactive systems, low rate, delay intolerant)
- Voice (sustained data rate, 64 kbps)
- Video (sustained data rate, multimegabit rates)
To support transactions that use data, voice, and video simultaneously
To provide high bandwidth, which can't be found in other technologies
To provide a uniform architecture for fast LANs and scalable WANs of unrestricted sizes
To provide bandwidth on demand (pay for use)
To support multicast operations (video conferencing)

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To provide guaranteed quality of service

To provide a unified approach in network management - ATM Basics

To understand how ATM can be used, it is important to have a knowledge of how ATM packages and transfers
information. The following sections provide brief descriptions of the format of ATM information transfer and the
mechanisms on which ATM networking is based.
8.1.5

ATM Cell Basic Format

To understand how ATM works, it is required to have a knowledge of how ATM packages and transfers
information, and the mechanisms on which ATM networking is based. The basic unit of information used by ATM
is a fixed-size cell consisting of 53 octets, or bytes. The first 5 bytes contain header information, such as the
connection identifier, while the remaining 48 bytes contain the data, or payload as shown fig below. Because the
ATM switch does not have to detect the size of a unit of data, switching can be performed efficiently. The small
size of the cell also makes it well suited for the transfer of real-time data, such as voice and video. Such traffic is
intolerant of delays resulting from having to wait for large data packets to be loaded and forwarded.
The information that transfers through ATM is packed in fixed size units called cells. Each cell contains 53
bytes. The first 5 bytes contain cell-header information, and the remaining 48 contain the user information or
Payload.
Fig. 8.1 ATM Cell Basic
Format
Compared with synchronous procedures that have fixed assignment of time slots, the cells used by a particular
end station do not have a fixed position in the cell stream. A continuous stream of cells moves from the user to
the network and vice versa. If there are no data to be transmitted, idle cells that contain no data are inserted into
the stream. If the transmission bandwidth requirements increase, the ratio of the used cells to idle cells will
increase. This means that the bandwidth can be very easily adapted: from a few megabits per second (Mbps) to
many Gigabits per second (Gbps). The rate can be constant for the whole transmission, or variable, depending
on the type of information and the desired level of quality.
8.1.6 ATM Cell Header Format
An ATM cell header can be one of two formats: UNI or NNI. The UNI header is used for communication between
ATM endpoints and ATM switches in private ATM networks. The NNI header is used for communication between
ATM switches. The diagram below shows a basic ATM Cell format, the ATM UNI cell header format, and the
ATM NNI cell header format.

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Fig.8.2
Cell
Header
Format

An ATM Cell, ATM UNI Cell, and ATM NNI Cell Header Each Contain 48 Bytes of Payload.
Unlike the UNI, the NNI header does not include the Generic Flow Control (GFC) field. Additionally, the NNI
header has a Virtual Path Identifier (VPI) field that occupies the first 12 bits, allowing for larger trunks between
public ATM switches.

8.1.7

ATM Cell Header Fields

In addition to GFC and VPI header fields, several others are used in ATM cell header fields.
The following illustration descriptions summarize the ATM cell header fields shown in the diagram below:

Fig. 8.3
Cell Header
Fields

Generic Flow Control (GFC) Provides local functions, such as identifying multiple stations that share a
single ATM interface. This field is typically not used and is set to its default value of 0 (binary 0000).
Virtual Path Identifier (VPI) In conjunction with the VCI, identifies the Path (route) to the next
destination of a cell as it passes through a series of ATM switches (nodes) on the way to its destination.
Virtual Channel Identifier (VCI)In conjunction with the VPI, identifies the next destination of a cell as it
passes through a series of ATM switches on the way to its destination.

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 Payload Type (PT)Indicates in the first bit whether the cell contains user data or control data. If the cell
contains user data, the bit is set to 0. If it contains control data, it is set to 1. The second bit indicates congestion (0
= no congestion, 1 = congestion), and the third bit indicates whether the cell is the last in a series of cells that
represent a single AAL5 frame (1 = last cell for the frame).
Cell Loss Priority (CLP)Indicates whether the cell should be discarded if it encounters extreme
congestion as it moves through the network. If the CLP bit equals 1, the cell should be discarded in preference to
cells with the CLP bit equal to 0.
Header Error Control (HEC)Calculates checksum only on the first 4 bytes of the header. HEC can
correct a single bit error in these bytes, thereby preserving the cell rather than discarding it.
8.1.8

Summary of ATM Cell Fields

Generic Flow Control (GFC) is used for control of cell flow only at the local user-network interface, to
alleviate short-term overload conditions in the network.
Virtual Path Identifier (VPI) field constitutes a routing field for the network.
Virtual Channel Identifier (VCI) field is used for routing to and from the end user.
Payload Type (PT) field indicates the type of information in the information field.
Cell loss priority (CLP) bit is used to provide guidance to the network in the event of congestion.
Header Error Control (HEC) field is used to correct and detect errors in the header

8.2

ATM CONCEPTS

8.2.1

Why ATM cells?

The motivation for the use of small data cells was the reduction of jitter (delay variance, in this case) in the
multiplexing of data streams; reduction of this (and also end-to-end round-trip delays) is particularly important
when carrying voice traffic. This is because the conversion of digitized voice back into an analog audio signal is
an inherently real-time process, and to do a good job, the codec that does this needs an evenly spaced (in time)
stream of data items. If the next data item is not available when it is needed, the codec has no choice but to
produce silence or guess - and if the data is late, it is useless, because the time period when it should have been
converted to a signal has already passed.
8.2.1

ATM Cell Base

ATM cells are the smallest standardized information units within the ATM network. All user and signaling
information must be represented within this cell format. Each cell contains 53 bytes. The first 5 bytes contains
cell-header information, and the remaining 48 bytes are available for the user or signaling information. To
guarantee fast processing in the network, the header has very limited functionality.
Its main function is to identify cells belonging to the same virtual channel and to perform the appropriate
routing. The identification of the virtual connection is done by an identifier which is selected at call setup and
guarantees proper routing of each packet. In addition, it allows an easy multiplexing of different virtual
connections over a single link. The information field is relatively small in order to reduce the internal buffers
in the switching node and to limit the queuing delays in those buffers: small buffers guarantee a small delay

83

and a small delay jitter, as required in real time systems. The information field is carried transparently through
the network; no processing is performed on it within the network. ATM base cell structure,
8.2.2 ATM Switching Operations
The basic operation of an ATM switch is straightforward: The cell is received across a
link on a known VCI or VPI value. The switch looks up the connection value in a local translation table to
determine the outgoing port (or ports) of the connection and the new VPI/VCI value of the connection on that
link. The switch then retransmits the cell on that outgoing link with the appropriate connection identifiers.
Because all VCIs and VPIs have only local significance across a particular link, these values are remapped, as
necessary, at each switch.
8.3

ATM DEVICES

An ATM network is made up of one or more ATM switches and ATM endpoints. An ATM endpoint (or end
system) contains an ATM network interface adapter. Workstations, routers, Data Service Units (DSUs), LAN
switches, and video coder-decoders (CODECs) are examples of ATM end systems that can have an ATM
interface. The fig below illustrates several types of ATM end systemsrouter, LAN switch, workstation, and
DSU/CSU, all with ATM network interfacesconnected to an ATM switch through an ATM network to another
ATM switch on the other side.

Fig. 8.4 ATM Devices

8.4 ATM NETWORK END SYSTEMS

An ATM Network consists of an ATM switch and ATM end systems. The ATM switches handle transmission of
cells through the ATM network. Its functions are: accepting the incoming cell from an ATM end station or
another ATM switch, reading and updating the cell-header information and switching the cell towards its
destination as fast as possible. The ATM end system contains an ATM network interface adapter. Examples of
such end systems are workstations, routers and LAN switches.

Fig. 8.5 Basic ATM Network

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The diagram below illustrates a private ATM network and a public ATM network carrying voice, video, and data
traffic.

Fig.8.6
ATM
Private
&
Public
Networks

A Private ATM Network and a Public ATM Network Both Can Carry Voice, Video, and Data Traffic.

8.4.1

ATM Cell Switch

An ATM cell switch marks the beginning of a cycle. Any cell fully arrived when the clock ticks is eligible for
switching during that cycle. A cell not fully arrived has to wait until the next cycle. Cells arrive at ATM speed,
normally about 150 Mbps. This works out around 360000 cells/sec, the cycle time has to be about 2.7 sec. A
commercial switch might have from 16 to 1024 input lines. At 622 Mbps the cycle time has to be about 700 nsec.
The fact that the cells are fixed length and short makes it possible to build such switches. With longer variablelength packets, high speed switching would be more complex, that is why ATM uses short fixed-length cells.
All ATM switches have two common goals:

Switch all cells with as low a discard rate as possible - cells can be dropped just in emergencies, but the
loss rate should be as small as possible.
Never reorder the cells on a virtual circuit - this constraint makes switch design considerably more
difficult, but is required by the ATM standard.

A problem arises if the cells arriving at more input lines want to go to the same output port in the same cycle.
ch has some number of input lines and output lines (both numbers are usually the same). ATM switches are
generally synchronous in the sense that during a cycle, one cell is taken from each input line, passed into the
internal switching fabric, and eventually transmitted on the appropriate output line.

Fig. 8.7 Basic ATM Switches

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Switches may be pipelined, that is, it may take several cycles before an incoming cell appears on its output line.
Cells actually arrive on the input lines asynchronously, so there is a master clock that triggers the start of the
transmission of all first cells that arrive on time.
8.4.2

ATM Network Interfaces (UNI, NNI, B-ICI)

An ATM network is made of ATM switches connected by ATM interfaces. There are two main types of
interfaces: UNI and NNI. The UNI (User-Network Interface) connects ATM end-systems (such as hosts and
routers) to an ATM switch. The NNI (Network-Network Interface) connects two ATM switches. UNI and NNI
are also subdivided into public and private UNIs and NNIs. A private UNI connects an ATM end-system and a
private ATM switch. A public UNI connects an ATM end-system or private switch to a public switch. A private
NNI connects two private ATM switches. A public NNI connects two public ATM switches. A third type is the
B-ICI (Broadband Inter Exchange Carrier) which connects between ATM switches of different suppliers.
Depending on whether the switch is owned and located at the customer's premises or is publicly owned and
operated by the telephone company, UNI and NNI can be further subdivided into public and private UNIs and
NNIs. A private UNI connects an ATM endpoint and a private ATM switch.
Its public counterpart connects an ATM endpoint or private switch to a public switch. A private NNI connects two
ATM switches within the same private organization. A public one connects two ATM switches within the same
public organization. An additional specification, the broadband inter-carrier interface (B-ICI), connects two
public switches from different service providers. The diagram shows the ATM interface specifications for private
and public networks.

Fig. 8.8

ATM Network Interface

8.5

ATM ADVANTAGES AND DISADVANTAGES

8.5.1

What are the Advantages of ATM?

ATM has many benefits. These include:

Scalability in distance (suitable for both WAN and LAN technologies).

Scalability in speed (from Mbs to Gbs ratios).

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8.5.2

Suitable for multiple traffic types by supporting voice, video, audio and data, allowing
multimedia and mixed services over a single network.
Enables new applications.
Provides the best multiple service support
Supports delay close to that of dedicated services
High evolution potential, works with existing, legacy technolgies
Supports many users.
Compatibility to various physical networks - it can be transported over wide variety of physical
networks.
Simplified network management - by using the same technology for all levels of the network.
Incremental migration capability.
Standards based, resulting in a long architectural lifetime.
Supports the broadest range of burstiness, delay tolerance and loss performance through the
implementation of multiple QoS classes.
Provides the capability to support both connection-oriented and connectionless traffic using
AALs
Able to use all common physical transmission paths like SONET and SDH.
Cable can be twisted-pair, coaxial or fiber-optic
Ability to connect LAN to WAN
Legacy LAN emulation
Efficient bandwidth use by statistical multiplexing
Scalability
Higher aggregate bandwidth High speed Mbps and Gbps
Are there any Disadvantages of ATM?

ATM has also some disadvantages. These are:

8.5.3

Overhead of cell header (5 bytes per cell).

Complex mechanisms for achieving QoS (Quality of Service).
Flexible to efficiencys expense, at present, for any one application it is usually possible to find a more
optimized technology
Cost, although it will decrease with time
New customer premises hardware and software are required
Other Basic ATM Points To Remember

8.6

ATM cells are small enough to fit into spaces too small for larger packets or frames, thus enabling
both speed and efficiency.
Traffic routes are pre-planned.
Switching is done without the need for time-consuming software.
Payload error checking and correction is performed only at the destination node, not at every hop
along the way.

B-ISDN and ATM TECHNOLOGIES

Broadband ISDN is based on ATM technology, which has emerged as a popular method of simultaneously
transmitting audio, video, and data. ATM has transmission rates between 1.544 Mbps and 622 Mbps.

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What Drives the ATM?

The drives behind the ATM Technology include:
Technology:

Protocol enhancements: decentralized, peer-to-peer networking; High-performance digital

transmission: simple protocols for more reliable transmission medium; Worldwide industry support: most sectors
of industry support ATM; Power to the desktop: processing power for multimedia transmission; Standards and
interoperability: fast realization and industry-wide agreement on a common set of standards.
Applications: Consumer service applications; Entertainment imaging; Work at home: telecommuting; Home
shopping; Video-on-Demand; Multimedia Email; Interactive multimedia applications and games; Broadcast
public service applications;

Distance learning;

Digital library;

Video-on-demand for training;

Videoconferencing
Business:

Evolution from shared medium to high-performance switching;

Virtual networking;

Seamless

interworking (Mixed voice, video, data, wireless traffic); More bandwidth for less bucks; Future proofing
investment; Enabling new applications.
Benefits: Integration of multiple traffic types; Efficient bandwidth use by statistical multiplexing; Guaranteed
bandwidth and resource allocation; Dynamic bandwidth management; High service availability; Multiple
Quality of Service (QOS) class support; Suitability for both delay or loss sensitive and delay or loss insensitive
traffic; Seamless private and public network technology; Automatic configuration and failure recovery; Costeffective fixed length cell processing; Improved transmission utilization; Future-proof investment.
8.7 ATM CONNECTIONS: VIRTUAL CHANNELS AND VIRTUAL PATH
ATM networks are fundamentally connection oriented. This means that a virtual connection needs to be
established across the ATM network prior to any data transfer. ATM virtual connections are of two general types:

Virtual Path Connections (VPCs), identified by a Virtual Path Identifier (VPI).

Virtual Channel Connections (VCCs), identified by the combination of a VPI and a Virtual Channel Identifier
VCI.
A Virtual Channel represents a given path between the user and the destination. This can be represented by user to
user connections, user to network connections, or network to network connections. A Virtual Path is created by
multiples of Virtual Channels heading / routed towards the same destination. With Virtual Paths the multiple
Virtual Channels are switched together. This helps reduce network overheads and, therefore, helps keep network
speed up. A virtual path is a bundle of virtual channels, all of which are switched transparently across the ATM
network on the basis of the common VPI. A VPC can be thought of as a bundle of VCCs with the same VPI value
as shown below.
Fig. 8.9 Virtual Path Connections (VPCs)

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8.7.1

ATM Virtual Path and Virtual Channel Connections

Every cell header contains a VPI field and a VCI field, which explicitly associate a cell with a given virtual
channel on a physical transmission path or link. It is important to remember the following attributes of VPIs and
VCIs:

VPIs and VCIs are not addresses, such as MAC addresses used in LAN switching.

VPIs and VCIs are explicitly assigned at each segment of a connection and, as such, have only local
significance across a particular link. They are remapped, as appropriate, at each switching point.
A transmission path is the physical media that transports VCs and VPs. The Fig. below shows how VCs
concatenate to create VPs, which, in turn, traverse the media or transmission path.

Fig. 8.10 VCs Concatenate to Create VPs

PVC allows direct connectivity between sites. In this way, PVC is similar to a leased line. Among its advantages,
PVC guarantees availability of a connection and does not require call setup procedures between switches.
Disadvantages of PVCs include static connectivity and manual setup. Each piece of equipment between the
source and the destination must be manually provisioned for the PVC. Furthermore, no network resiliency is
available with PVC. An SVC is created and released dynamically and remains in use only as long as data is being
transferred. In this sense, it is similar to a telephone call. Dynamic call control requires a signaling protocol
between the ATM endpoint and the ATM switch. The advantages of SVCs include connection flexibility and call
setup that can be handled automatically by a networking device. Disadvantages include the extra time and
overhead required to set up the connection.
8.7.2

Why Virtual Circuits?

ATM is a channel-based transport layer, using Virtual circuits (VCs). This is encompassed in the concept of the
Virtual Paths (VP) and Virtual Channels. Every ATM cell has an 8- or 12-bit Virtual Path Identifier (VPI) and 16bit Virtual Channel Identifier (VCI) pair defined in its header. Together, these identify the virtual circuit used by
the connection. The length of the VPI varies according to whether the cell is sent on the user-network interface
(on the edge of the network), or if it is sent on the network-network interface (inside the network).
8.7.3

Advantages of Virtual Paths

Simplified network architecture

Increased network performance and reliability

Reduced processing and short connection setup time

Enhanced network services

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8.7.4

ATM is Fast and has: High Bandwidth, High-capacity multimedia capabilities for Voice, Video and Data
communications.
ATM VP / VC Characteristics

Improved Quality of service

Switched and semi-permanent virtual-channel connections

Cell sequence integrity

Traffic parameter negotiation and usage monitoring

VP Control Signaling: Semi-permanent VCs may be used for user-to-user exchange; no control
signaling is required.

Meta-signaling channel: A permanent low-data-rate channel used for a VC that can be used for call
control:

user-to-network signaling VC can then be used to set up VCs to carry user data

can also be used to set up a user-to-user signaling VC within a pre-established VC.

A virtual path can be established on a semi-permanent basis by prior agreement. No control signaling is
required.
Establishment/release may be customer controlled. Customer uses a signaling virtual channel to request
the virtual path from the network.
Establishment/release may be network controlled. Network establishes a virtual path (may be network-tonetwork, user-to-network, or user-to-user)

8.8 ATM PROTOCOLS ATM REFERENCE MODEL

What Are ATM Protocols (ATMP)?
ATMPs are Networking Standards for transferring data in cells of fixed sizes. ATM can be used in both LAN and
WAN. In the ATMP domain the socket interface provides an application with direct access to an ATM VC. The
ATMP is offered as a new address family. Each Protocol Data Unit (PDU) to be transmitted over the connection is
handed directly to the AAL for segmentation into ATM cells and subsequent transmission. Similarly, received
PDUs are handed up to the socket layer by the adaptation layer on completion of reassembly.
Standards for ATM UNI signalling provide a mechanism for the selection of the AAL and QoS to be used for an
ATM connection.
ATM Protocol Architecture (Diagram)

Fig. 8.11 ATMP Reference Model Planes

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User plane: Provides for user information transfer

Control plane: Call and connection control

Management plane: Plane management whole system functions

Layer management: Resources and parameters in protocol entities

8.9 TRANSMISSION OF ATM CELLS

8.9.1

Asynchronous Transmission

Unlike TDM in traditional telecommunication systems, ATM does not assign any time slots to a given user and it
is not tied to a master clock. Instead, ATM dynamically allocates ATM cells when user equipment wishes to
transmit information. Cells representing traffic generated by different users are multiplexed, as shown below, over
the physical transmission medium. However, ATM does not run directly over the physical medium, so the
physical layer technology is used to provide ATM with a set of necessary functions. Thus, ATM cells are often
transmitted in the asynchronous mode by using the synchronous physical layer.

Fig. 8.12 ATM Cell Transmission

8.9.2 Transmission in ATM Networks

ATM stands for Asynchronous Transfer Mode. This mode can be contrasted with the synchronous T1/E1 carrier
shown below. T1/ E1 frames are generated precisely every 125 sec. This rate is governed by a master clock. Slot
k of each frame contains 1 byte of data from the same source. ATM has no requirements that cells rigidly alternate
among the various sources. Cells arrive randomly from different sources. The stream of cells need not be
continuous. Gaps between the data are filled by special idle cells. ATM does not standardize the format for
transmitting cells. Cells are allowed to be sent individually, or they can be encased in a carrier such as T1/E2,
T3/E3, T4/E4, SONET, SDH or FDDI. For these examples, standards exist telling how cells are packed into the
frames. The transmission medium for ATM is normally fiber optics, but for runs under 100m, coax or cable
category 5 twisted pair are also acceptable. Each link goes between a computer and an ATM switch, or between
two ATM switches. So, all ATM links are point-to-point. Each link is unidirectional. For full-duplex operation,
two parallel links are needed.

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Fig. 8.13
ATM
Transmission
Path

8.10

ATM NETWORK SWITCH

8.10.1 The Function of an ATM Switch

An ATM network is made up of an ATM switch and ATM endpoint devices. An ATM switch is responsible for
cell transit through an ATM network. The job of an ATM switch is well defined: It accepts the incoming cell
from an ATM endpoint or another ATM switch. It then reads and updates the cell header information and quickly
switches the cell to an output interface toward its destination.
8.10.2

The Composition of an ATM network

An ATM endpoint (or end system) contains an ATM network interface adapter. Examples of ATM endpoints are
workstations, routers, digital service units (DSUs), LAN and WAN switches, and video coder-decoders
(CODECs). The diagram below shows an ATM network made up of ATM switches and ATM endpoints.

Fig. 8.14
An ATM Network
Comprising ATM
Switches and
Endpoints

8.11

ATM QUALITY OF SERVICE (QoS)

8.11.1 What is QoS?

In the field of packet-switched telecommunication networks, computer networking and in traffic engineering, the
term Quality of Service (QoS) refers to resource reservation control mechanisms rather than the achieved service
quality. Quality of service is the ability of a network to provide different priority to different applications, users,
or data flows, or to guarantee a certain level of network performance to a data flow. For example, a required bit
rate, delay, jitter, packet dropping and cell discarding probabilities and bit error rates may be guaranteed. Quality
of service guarantees are important if the network capacity is insufficient, especially for real-time streaming
multimedia applications such as voice over IP, online games and IP-TV, since these often require fixed bit rate
and are delay sensitive, and in networks where the capacity is a limited resource, for example in cellular data
communication. In the absence of network congestion, QoS mechanisms are not required. ATM supports QoS
guarantees comprising of traffic contract, traffic shaping, and traffic policing.

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A traffic contract specifies an envelope that describes the intended data flow. This envelope specifies values for
peak bandwidth, average sustained bandwidth, and burst size, among others. When an ATM end system connects
to an ATM network, it enters a contract with the network, based on QoS parameters.
8.11.2 Why Quality of Service
ATM provides 5 classes of service, which in turn can offer support to most existing data services and their various
traffic patterns. Certain services can tolerate some cell loss while others cannot. Other services have timing
constraints while some do not. And some may have multiple constraints while some have none. Thus, in order for
ATM to be able to support all the different services, while using available network resources efficiently and
providing specified and guaranteed levels of QoS, it needs to bound and control the different traffic parameters
associated with each service.
8.11.3 ATM QoS Parameters
A set of QoS Parameters are negotiated when a connection is set up in an ATM network. These parameters are
used to measure the QoS of a connection and quantify end-to-end network performance at the ATM layer. The
network should guarantee the negotiated QoS by meeting certain values of these parameters. These include:

Cell Transfer Delay (CTD). The delay experienced by a cell between the time it takes for the first bit of
the cell to be transmitted by the source and the last bit of the cell to be received by the destination.
Maximum Cell Transfer Delay (Max CTD) and Mean Cell Transfer Delay (Mean CTD) are used.
Peak-to-peak Cell Delay Variation (CDV). The difference between the maximum and minimum CTD
experienced during the connection. Peak-to-peak CDV and Instantaneous CDV values are used.

8.12 ATM ADAPTATION LAYER (AAL)

Practically, the AAL is a sub-layer of the ATM Reference Model. The ATM architecture uses a logical model to
describe the functionality that it supports. ATM functionality corresponds to the physical layer and part of the data
link layer of the OSI reference model as shown in the diagram below.

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Fig. 8.15 Mapping of ATM Reference Model Unto the OSI Reference Module
The ATM Reference Model Relates to the Lowest Two Layers of the OSI Reference Model. It is composed of the
following planes, which span all layers:

ControlThis plane is responsible for generating and managing signaling requests.

UserThis plane is responsible for managing the transfer of data.

ManagementThis plane contains two components:

Layer management:
problems.

manages layer-specific functions, such as the detection of failures and protocol

Plane management: manages and coordinates functions related to the complete system.

The ATM reference model is composed of the following ATM layers:

The ATM Physical layerAnalogous to the physical layer of the OSI reference model, the ATM physical
layer manages the medium-dependent transmission.
The ATM Layer
The ATM layer is responsible for:

Cell multiplexing and demultiplexing

Virtual path identification (VPI), and virtual channel identification (VCI)

Cell header generation/extraction

Generic Flow Control

The cell multiplexing functions enable multiple logical connections across a single interface. Cell header
information in appended by the sending device for use by the receiving device, this is handled by the cell header
generation/extraction functions. Placement of cells is handled by the generic flow control functions.
ATM layerCombined with the ATM adaptation layer, the ATM layer is roughly analogous to the data link
layer of the OSI reference model. The ATM layer is responsible for the simultaneous sharing of virtual circuits
over a physical link (cell multiplexing) and passing cells through the ATM network (cell relay). To do this, it uses
the VPI and VCI information in the header of each ATM cell.
ATM adaptation layer (AAL)Combined with the ATM layer, the AAL is roughly analogous to the data
link layer of the OSI model. The AAL is responsible for isolating higher-layer protocols from the details of the
ATM processes. The adaptation layer prepares user data for conversion into cells and segments the data into 48byte cell payloads.
Finally, the higher layers residing above the AAL accept user data, arrange it into packets, and hand it to the AAL.

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8.12.1 The ATM Physical Layer

The ATM physical layer has four functions: Cells are converted into a bitstream, the transmission and receipt of
bits on the physical medium are controlled, ATM cell boundaries are tracked, and cells are packaged into the
appropriate types of frames for the physical medium. For example, cells are packaged differently for SONET than
for DS-3/E-3 media types. The ATM physical layer is divided into two parts: the Physical Medium Dependent
(PMD) sublayer and the transmission convergence (TC) sublayer. The PMD sublayer provides two key functions.
First, it synchronizes transmission and reception by sending and receiving a continuous flow of bits with
associated timing information. Second, it specifies the physical media for the physical medium used, including
connector types and cable.
8.12.2 What is AAL?
AS explained diagrammatically above, the AAL is a sub layer of the ATM Reference Model, responsible for
isolating higher-layer protocols from the details of the ATM processes. The adaptation layer prepares user data for
conversion into cells and segments the data into 48-byte cell payloads. The AAL consists of two sublayers:

The Segmentation And Reassemble sublayer, SAR

Convergence Sublayer, CS.

The SAR sublayer packs the information received from CS into cells for transmission and handles unpacking at
the other end. The convergent sublayer provides specific application support for applications using AAL. This
sublayer is service dependent as applications attach to the AAL at specific service access points. The SAR
sublayer packs the information received from CS into cells for transmission and handles unpacking at the other
end. The SAR must pack all SAR header and trailer information as well as CS header and trailer information into
48-octet blocks.

8.12.3 Classification of AAL Types

The ITU-T has defined four types of service classifications. These are:

Type 1 - this classification requires constant bit rate, maintenance of timing relation, and is a connection
oriented classification.
Type 2 - This includes variable bit video, i.e., video conference. Here the application is connection
oriented, timing is important, however the bit rate will vary over time.

Type 3/4 - initially this was two classifications, however, ITU-T combined them due to similarity in
processing.
This type represents data transfer applications, and with varying bit rates, no timing
requirements with type three being connection oriented, and type four being connectionless orientated.

Type 5 is a new classification which was introduced to provide streamline transport facilities.

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Note: A PDU is a Protocol Data Unit. It is used to pack the information from the higher layers into a manageable
unit that will be passed to the ATM layer.
8.12.4
a)

AAL Protocol Types

AAL Type 1

Type one operations deal with a constant bit rate. It is therefore, the responsibility of the SAR protocol to pack
and unpack cells for transmission. Blocks contain a sequence number so that errored PDUs can be trapped. A
sequence number protection field is also contained in the cell to assist with error detection and correction. The CS
for type one deals with clocking and synchronisation, therefore no CS information is required to be transmitted
with the cell.
b)

AAL Protocol Type 2

This type deals with variable bit rate transmissions. Type two has been set up to deal with analog transmission of
video and sound. This type of transmission requires a constant connection, but data bit rates will change based on
the amount of activity. Initially, ITU-T provided a specification for the type two protocol, however, this has been
recalled and is currently under review. Due to this recall, there is no CS and SAR protocols.
c)

AAL Protocol Type 3/4

As with type two this type handles variable bit rate transmission. This type may be either connection based or
connectionless, and can be either message-mode or streaming-mode. As the ATM layer transfers data in cells of
48-octets, the AAL layer must provide segmentation and reassemble functions. For type 3/4, the higher layer
passes a block of data to the CS, then converts this to a PDU incorporating a CPCS (common part convergent
sublayer) header and trailer.
d)

AAL Protocol Type 5

This type was introduced to minimize protocol processing that exists in type 3 / 4. This is done by moving most of
the protocol information from the SAR PDU back to the CPCS PDU. Thus the CPCS PDU in type five contains a
32 bit CRC, user identification, interpretation identification, and a length field. This reduces the processing in the
SAR. The only function now performed by the SAR is the segmentation of the CPCS PDU into 48 octet cells for
passing to the ATM layer.
8.12.5

AAL SERVICES

ATM Service Categories

One of the main benefits of ATM is to provide distinct classes of service for the varying bandwidth, loss, and
latency requirements of different applications. Some applications require constant bandwidth, while others can
adapt to the available bandwidth, perhaps with some loss of quality. Still others can make use of whatever
bandwidth is available and use dramatically different amounts from one instant to the next.

Table 8.1 AAL Service Categories and Characteristics

Service Category

Traffic Parameters

QoS Characteristics

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Cell Loss

Cell Delay

CBRConstant Bit Rate

PCR

Low

lLw

VBR-RTVariable Bit Rate Real-Time

PCR, SCR, MBS

Low

Low

VBR-NRTVariable Bit Rate Non-Real Time

PCR, SCR, MBS

Low

Unspecified

ABRAvailable Bit Rate

PCR, MCR

Unspecified

Unspecified

UBRUnspecified Bit Rate

(no guarantees)

Unspecified

Unspecified

Note:
PCR
MCR

- Program Clock Reference;

- Minimum Cell Rate

Table 8.2

SCR, -

Sustainable Cell Rate; MBS

Maximum Burst Size;

AAL Service Classification

8.12.6 LATENCY / SPEED EFFECT

The key performance issues that relate to the requirements for ATM Traffic and Congestion Control are Latency /
Speed Effects and Cell Delay Variation. Consider the transfer of ATM Cells over a network at a data rate of 150
Mbps. At that rate, it takes (53 x 8 bits) / (150 X 10 6 bps) approximately 2.8 106 seconds to insert a single cell
into the network.
The time it takes to transfer the cell from the source to the destination user will depend on the number of
intermediate ATM switches, the switching time at each switch and the propagation time along all links in the path
from source to destination. There is a considerable delay to transfer multiples of ATM Cells over the network.
These delays are known as Latency, which introduce ATM network Cell Delays and Cell Delay Variations
which in turn slow down the designed speed of the network. This phenomenon is known as Speed Effect.
Cell Delay Variation
What is Cell Delay variation?

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Cells progressing along a virtual connection in an ATM network experience random delays, due to queuing in
multiplexing stages. These random delays generally induce an alteration in the initial time structure of any cell
stream passing through the network. This phenomenon is known as cell delay variation (CDV). As mention in the
QoS parameters section above, Cell Delay Variation (CDV) is the difference between the maximum and minimum
CTD experienced during the connection. Peak-to-peak CDV and Instantaneous CDV are used to measure quality
of service parameters. CDV is one of the QoS parameters that can be negotiated between applications and an
ATM network. The network should check during connection setup, as part of call admission control, whether it
can satisfy the requested CDV of an application. For this comparison, the network should estimate the end-to-end
CDV that it can support, by using local information about cell delays and delay variations in switches.
ATM NETWORK RESOURCE MANAGEMENT
What are ATM Network Resources?
ATM Network Resources include the combination of fixed cell sizes, the concept of VPs and VCs and five
different categories of ATM adaptation layer protocols make ATM one of the most flexible high speed networking
technologies implemented today. In addition to it's flexibility for supporting voice, video and data and ability to
scale to large networks, ATM is the first and so far the only widely implemented technology that can offer the
much needed QoS. Some of the ATM resources responsible for these achievements are discussed below.
ATM Network Resource Management
ATM Network Resource Management is the process of keeping track of the way link resources are allocated to
connections in broadband networks ATM networks. The two primary resources that are tracked by network
resource management are capacity (bandwidth) and connection identifiers. Network resource management keeps
track of the capacity and controls the allocation of capacity to connections when requested as part of the
connection setup process - ref. to traffic contract. Network resource management deals with protocols and
networks capable of performing a reservation of the available resources in order to guarantee a certain QoS.
Examples of these technologies are ATM and Multi-Protocol Label Switching (MPLS), which are usually used in
core networks. An important objective of network providers is to obtain the maximum profit from their resources;
hence there is a need for an efficient resource management.
ATM Network Resource Management is actually a congestion control technique concerned with ATM:
Cell loss ratio
Cell transfer delay
Cell delay variation

CHAPTER 9
TRAFFIC FLOW AND CONGESTION CONTROL

9.1

OVERVIEW
This Chapter will be discussing: Two Congestion Control Strategies, Admission Control
Mechanisms, Congestion-control Mechanisms, Traffic Rate Management,
The Traffic

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Management/Policing, Explicit Congestion Avoidance, Implicit Congestion Control, ECN

(Explicit Congestion Notification), ATM Connection Admission Control (CAC), Usage
Parameter Control, ATM Cell Priority Control, ATM Traffic Shaping And Traffic Policing.
Traffic Rate Management

9.1.1 What is Traffic Flow?

Basically, Traffic Flow is the phenomenon whereby information / data is propagated or sent (duplex)
through Network Nodes, Links, Channels, Routes, Routers, Switches, Hubs, Servers, Bridges,
Gateways, Etc.
9.1.2

Traffic Flow in Circuit Switching

This is where information or data is sent via the network elements listed above continuously.

9.1.3

Traffic Flow in Packet Switching

In Packet switching networks, Traffic Flow (Packet Flow or Network Flow) is a sequence of packets
from a source computer to a destination, which may be another host, a multicast group, or a broadcast
domain. Traffic flow is "an artificial logical equivalent to a call or connection. In other words, it is a
sequence of packets sent from a particular source to a particular Unicast, Anycast, or Multicast
destination that the source desires to label as a flow. A flow could consist of all packets in a specific
transport connection or a media stream. However, a flow is not necessarily 1:1 mapped to a transport
connection. Flow is also defined as a set of IP packets passing an observation point in the network
during a certain time interval.
9.2

Network Congestion

Basically, Network Congestion is the inability of a network to meet traffic demanded by users (due to insufficient
network resources) at a given time (practically during Busy Hours).
Causes of Network Congestion:
The basic cause of congestion is that the input traffic demands exceed the capacity of the network
In typical packet switching networks, this can occur quite easily when output links are slower than inputs
Multiple traffic sources competing for same output link at the same time, etc.
Effects of Congestion: Congestion is undesirable because it can cause increased delay, due to queuing within the
network, Packet loss, due to buffer overflow, reduced throughput, due to packet loss and retransmission during
rush hour traffic flow.
9.3

CONGESTION CONTROL

Congestion control is defined as the set of actions performed by the network to prevent or reduce congestion.
Congestion control is the most important part of the traffic management issue. Congestion control is a means of
minimizing congestion effects and preventing them from spreading. It can employ connection admission and or
usage parameter control and network parameter control to avoid overload situations. Congestion control is a state
of network elements in which traffic and/or control resource overload means that the network is not able to

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guarantee the negotiated quality of service to the established connections and to the new connection requests.
Congestion can be caused by unpredictable statistical fluctuations of traffic flows or a network fault.
Congestion control concerns controlling traffic entry into a telecommunications network, so as to avoid
congestive collapse by attempting to avoid oversubscription of any of the processing or link capabilities of the
intermediate nodes and networks and taking resource reducing steps, such as reducing the rate of sending packets.
It should not be confused with flow control, which prevents the sender from overwhelming the receiver.
Why Congestion Control
An ATM network needs Traffic Control capabilities to cope with the various service classes and to cope with
potential errors within the network at any time (e,g., a problem with the physical layer). The network should have
the following traffic control capabilities.
A network that controls congestion must:

9.4

be responsive to the different utility functions of the users

be able to manage the resources so that there is no loss of utility as the load increases

CONGESTION AVOIDANCE

Congestion avoidance is the capability of a network to anticipate congestion and to prevent / avoid it so that
congestion never occurs.
9.4.1

Explicit Congestion Avoidance

Explicit congestion notification is proposed as the congestion avoidance policy. It tries to keep the network
operating at its desired equilibrium point so that a certain QOS for the network can be met. To do so, special
congestion control bits have been incorporated into the address field of the frame relay: FECN and BECN. The
basic idea is to avoid data accumulation inside the network. FECN means Forward Explicit Congestion
Notification. FECN bit can be set to 1 to indicate that congestion was experienced in the direction of the frame
transmission, so it informs the destination that congestion has occurred. BECN means Backwards Explicit
Congestion Notification. BECN bit can be set to 1 to indicate that congestion was experienced in the network in
the direction opposite of the frame transmission, so it informs the sender that congestion has occurred.

9.4.2

Explicit Congestion Notification (ECN)

The problem with RED is that it drops packets. A more efficient technique would be for a router to set a
congestion notification bit in a packet, and then send the packet to the receiver. The receiver could then inform the
sender to slow down via a message in the ACK. All the while, the receiver gets its packet and we avoid using
packet drops to signal congestion. ECN is an end-to-end congestion avoidance mechanism that adopts this
technique. As the name implies,
ECN provides direct notification of congestion rather than indirectly signaling congestion via dropped packets.
ECN works when congestion is moderate. When congestion gets excessive, packet-drop techniques are used.

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ECN-enabled routers set a CD (congestion experienced) bit in the packet header of packets from ECN-capable
hosts when the length of a queue exceeds a certain threshold value. The packets are forwarded to the receiver,
which then sends an ACK to the sender that contains the congestion indicator. This ACK is called an ECN-Echo.
When the sender receives this explicit signal, it halves the rate at which it sends packets.
9.4.3 Implicit Congestion Control
Implicit Congestion Notification - Some upper layer protocols, such as Transport Control Protocol (TCP),
operating in the end devices have an implicit form of congestion detection. These protocols can infer that
congestion is occurring by an increase in round trip delay or by detection of the loss of a frame, for example.
Reliance on network traffic characteristics to indicate congestion is known as implicit congestion notification.
These upper layer protocols were developed to run effectively over networks whose capacity was undetermined.
Such protocols limit the rate at which they send traffic onto the network by means of a "window," which allows
only a limited number of frames to be sent before an acknowledgment is received.
When it appears that congestion is occurring, the protocol can reduce its window size, which reduces the load on
the network. As congestion abates, the window size is gradually increased. The same window-size adjustment is
also the normal way for the end-user devices to respond to explicit congestion notification FECN and BECN.
The ANSI standards state that implicit and explicit congestion notification are complementary and can be used
together for best results.
9.4.4

9.5

Collision Avoidance Technique

Each frame handler monitors its congestion and sets BECN and/or FECN as appropriate.
BECN notifies the user that congestion avoidance procedures should be initiated where applicable for
traffic in the opposite direction of the received frame. The notification indicates that frames transmitted
by the user on this logical connection may encounter congested resources.
User Response (UR) reduces transmitted frame rate until the signal ceases.
FECN notifies the user that congestion avoidance procedures should be initiated where applicable for
traffic in the same direction as the received frame. The notification indicates that this frame, on this
logical connection, has encountered congestion.
User Response notifies its peer user of this connection to restrict its flow of frames. This must be done at
a higher layer of protocol.

ATM CONNECTION ADMISSION CONTROL

What is Connection Admission Control (CAC)? In ATM networks, CAC is the set of actions taken by the
network during call set-up (or re-negotiation) phase to establish if a VP or VC can be accepted by the network. It
is the first line of defense against network congestion.

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A connection can only be established if sufficient network resources are available to establish the
connection from-end-to-end with the required QoS. The agreed QoS for any of the existing channels
must not be affected by the new connection.

Network accepts connection only if it can meet traffic demands. Users specify traffic characteristics for
new connection (VCC or VPC) by selecting a QoS.

Two classes of parameters are foreseen to support connection admission control :

a) A set of parameters describing the source traffic characteristics.
b) Another set of parameters to identify the quality of service required.

9.5.1 Connection-Oriented Networks and Connection Admission Control

Policing can be imposed at every node in an ATM network. Sources are required to ensure their traffic complies
with the traffic contract in force if they wish to avoid policing, and may implement traffic shaping to achieve this
optimally. Connection-oriented networks in ATM systems perform CAC based on traffic contracts. An
application that wishes to use a connection-oriented network to transport traffic must first request a connection
(through signalling, for example ITU-T Q.2931 Protocol), which involves informing the network about the
characteristics of the traffic and the QoS required by the application. This information is stored in a traffic
contract. If the connection request is accepted, the application is permitted to use the network to transport traffic.
9.5.2

Usage Parameter Control

Usage Parameter Control (UPC) and Network Parameter Control (NPC) do the same job at different interfaces.
The UPC function is performed at the User Network Interface (UNI), while the NPC function is performed at the
Network Node Interface (NNI). The main purpose of UPC/NPC is to protect the network resources from
malicious as well as unintentional traffic misbehaviour which can affect the quality of service of other already
established connections. What is actually carried out depends on the access network configuration. Usage
parameter control can simply discard cells that violate the negotiated traffic parameters. In addition a 'guilty'
connection may be released. A less rigorous measure would be to 'tag ' the cells and let them through if they do
not cause harm to the network.
9.5.3

ATM Cell Priority Control

ATM cells have an explicit Cell Loss Priority (CLP) bit in the header so at least two different ATM priority classes
can be distinguished. A single ATM connection can have both priority classes when the information to be
transmitted is classified into more and less important parts.
Each ATM cell header has a CLP bit used to identify cells as either conforming (to the contract) or
nonconforming. If cells are nonconforming - for example, more cells than the contract allows - the ATM switch
sets the CLP bit to one. This cell can now be transferred through the network only if there is sufficient network
capacity. If there is not enough bandwidth available, the nonconforming cell is discarded and may need to be
retransmitted. This process is known as ATM Cell Priority Control.
9.6

ATM TRAFFIC SHAPING AND TRAFFIC POLICING

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9.6.1

Traffic Shaping

Traffic shaping (also known as "packet shaping") is the control of network traffic in order to optimize or
guarantee performance, lower latency, and/or increase usable bandwidth by delaying packets that meet certain
criteria. More specifically, traffic shaping is any action on a set of packets (often called a stream or a flow) which
imposes additional delay on those packets such that they conform to some predetermined constraint (a contract or
traffic profile). Traffic shaping provides a means to control the volume of traffic being sent into a network in a
specified period (bandwidth throttling), or the maximum rate at which the traffic is seen. This control can be
accomplished in many ways and for many reasons; however traffic shaping is always achieved by delaying
packets. Traffic shaping is commonly applied at the network edges to control traffic entering the network, but can
also be applied by the traffic source (for example, computer or network card) or by an element in the network.
Traffic policing is the distinct but related practice of packet dropping and packet marking.
9.6.2

Traffic Policing

What is Traffic Policing? Traffic Policing is the process of monitoring network traffic for conformity with a
traffic contract and if required, dropping traffic to enforce compliance with that contract. Traffic sources which
are aware of a traffic contract sometimes apply Traffic Shaping in order to ensure their output stays within the
contract and is thus not dropped. The need for police action occurs in an ATM network when traffic flow exceeds
the negotiated rate (in the traffic contract) and the buffer overflows. To maintain network performance it is
possible to police virtual circuits against their traffic contracts. If a circuit is exceeding its traffic contract, the
network can either drop the cells or mark the Cell Loss Priority (CLP) bit (to identify a cell as discardable farther
down the line). Basic policing works on a cell by cell basis, but this is sub-optimal for encapsulated packet traffic
(as discarding a single cell will invalidate the whole packet). As a result, schemes such as Partial Packet Discard
(PPD) and Early Packet Discard (EPD) have been created that will discard a whole series of cells until the next
frame starts. This reduces the number of redundant cells in the network, saving bandwidth for full frames. EPD
and PPD work with AAL5 connections as they use the frame end bit to detect the end of packets.
9.7

Congestion Management Summary

Congestion management features allow us to control congestion by determining the order in which packets are
transmitted out an interface based on priorities assigned to those packets. Congestion management entails the
creation of queues, assignment of packets to those queues based on the packet's classification, and scheduling of
the packets in a queue for transmission. The congestion management QoS feature offers four types of queuing
protocols, each of which allows you to specify creation of a different number of queues, affording greater or lesser
degrees of differentiation of traffic and the order in which that traffic is transmitted. Congestion management is
crucial in meeting service level agreements (SLAs) and ensuring that mission-critical traffic is prioritized and
delivered with minimal delay. Congestion management involves the use of queues in the router to hold excess
packets during congestion until the interface is free to transmit them.
Presently, several different queuing algorithms are available. The proper use of queuing mechanisms can
influence the order in which the packets waiting in the queues are serviced.

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