INDEX

CHAPTER
1
2

CONTENTS

PAGE

OVERVIEW OF TELECOMMUNICATION NETWORKSI

OVERVIEW OF TELECOMMUNICATION NETWORKSII

23

PCM PRINCIPLES

40

DIGITAL SWITCHING

60

SIGNALLING IN TELECOMMUNICATION

73

FIBRE-OPTIC COMMUNICATION TECHNOLOGY

106

OF TRANSMISSION SYSTEMS & THEIR FEATURES

125

SDH

133

DWDM

145

10

MOBILE COMMUNICATION

160

11

CDMA TECHNOLOGY

194

12

INTRODUCTION TO BROADBAND

213

13

INTELLIGENT NETWORK

248

1
Dept of. EEE

Chapter-1
OVERVIEW OF TELECOMMUNICATION NETWORKS-I
a telecommunication device that is used to transmit and receive electronically or digitally
encoded speech between two or more people conversing. It is one of the most
common household appliances in the world today. Most telephones operate through
transmission of electric signals over a complex telephone network which allows almost
any phone user to communicate with almost any other user.
Telecommunication networks carry information signals among entities, which are
geographically far apart. An entity may be a computer or human being, a facsimile
machine, a teleprinter, a data terminal and so on. The entities are involved in the process
of information transfer that may be in the form of a telephone conversation (telephony) or
a file transfer between two computers or message transfer between two terminals etc.
With the rapidly growing traffic and untargeted growth of cyberspace,
telecommunication becomes a fabric of our life. The future challenges are enormous as
we anticipate rapid growth items of new services and number of users. What comes with
the challenge is a genuine need for more advanced methodology supporting analysis and
design of telecommunication architectures. Telecommunication has evaluated and growth
at an explosive rate in recent years and will undoubtedly continue to do so.
The communication switching system enables the universal connectivity. The
universal connectivity is realized when any entity in one part of the world can
communicate with any other entity in another part of the world. In many ways
telecommunication will acts as a substitute for the increasingly expensive physical
transportation.
The telecommunication links and switching were mainly designed for voice
communication. With the appropriate attachments/equipments, they can be used to
transmit data. A modern society

Introduction

Voice Signal Characteristics

Telecommunication is mainly concerned with the transmission of messages
between two distant points. The signal that contains the messages is usually converted
into electrical waves before transmission. Our voice is an analog signal, which has
amplitude and frequency characteristics.

Dept of. EEE

Voice frequencies: - The range of frequencies used by a communication device

determines the communication channel, communicating devices, and bandwidth or
information carrying capacity. The most commonly used parameter that characterizes an
electrical signal is its bandwidth of analog signal or bit rate if it is a digital signal. In
telephone system, the frequencies it passes are restricted to between 300 to 3400 Hz.
In the field of telecommunications, a Telephone exchange or a Telephone switch
is a system of electronic components that connects telephone calls. A central office is the
physical building used to house inside plant equipment including telephone switches,
which make telephone calls "work" in the sense of making connections and relaying the
speech information.

Switching system fundamentals

Telecommunications switching systems generally perform three basic functions:
they transmit signals over the connection or over separate channels to convey the identity
of the called (and sometimes the calling) address (for example, the telephone number),
and alert (ring) the called station; they establish connections through a switching network
for conversational use during the entire call; and they process the signal information to
control and supervise the establishment and disconnection of the switching network
connection.
In some data or message switching when real-time communication is not needed,
the switching network is replaced by a temporary memory for the storage of messages.
This type of switching is known as store-and-forward switching.

Signaling and control

The control of circuit switching systems is accomplished remotely by a specific
form of data communication known as signaling. Switching systems are connected with
one another by telecommunication channels known as trunks. They are connected with
the served stations or terminals by lines.
In some switching systems the signals for a call directly control the switching
devices over the same path for which transmission is established. For most modern
switching systems the signals for identifying or addressing the called station are received
by a central control that processes calls on a time-shared basis. Central controls receive
and interpret signals, select and establish communication paths, and prepare signals for
transmission. These signals include addresses for use at succeeding nodes or for alerting
(ringing) the called station.
Most electronic controls are designed to process calls not only by complex logic
but also by logic tables or a program of instructions stored in bulk electronic memory.
The tabular technique is known as translator. The electronic memory is now the most
accepted technique and is known as stored program control (SPC). Either type of control
Dept of. EEE

may be distributed among the switching devices rather than residing centrally.
Microprocessors on integrated circuit chips are a popular form of distributed stored
program control.

Switching fabrics
Space and time division are the two basic techniques used in establishing
connections. When an individual conductor path is established through a switch for the
duration of a call, the system is known as space division. When the transmitted speech
signals are sampled and the samples multiplexed in time so that high-speed electronic
devices may be used simultaneously by several calls, the switch is known as time
division.
In the early stages of development in telecommunication, manual switching
methods were deployed. But later on to overcome the limitations of manual switching;
automatic exchanges, having Electro-mechanical components, were developed. Strowger
exchange, the first automatic exchange having direct control feature, appeared in 1892 in
La Porte (Indiana). Though it improved upon the performance of a manual exchange it
still had a number of disadvantages, viz., a large number of mechanical parts, limited
availability, inflexibility, bulky in size etc. As a result of further research and
development, Crossbar exchanges,having an indirect control system, appeared in 1926 in
Sweden.
The Crossbar exchange improved upon many short- comings of the Strowger
system. However, much more improvement was expected and the revolutionary change
in field of electronics provided it. A large number of moving parts in Register, marker,
Translator, etc., were replaced en-block by a single computer. This made the exchange
smaller in size, volume and weight, faster and reliable, highly flexible, noise-free, easily
manageable with no preventive maintenance etc.

Network Architecture.
When electronic devices were introduced in the switching systems, a new concept
of switching evolved as a consequence of their extremely high operating speed compared
to their former counter-parts, i.e., the Electro-mechanical systems, where relays, the logic
elements in the electromechanical systems, have to operate and release several times
which is roughly equal to the duration of telephone signals to maintain required accuracy.
Research on electronic switching started soon after the Second World War, but
commercial fully electronic exchange began to emerge only about 30 years later.
However, electronic techniques proved economic for common control systems much
earlier. In electromechanical exchanges, common control systems mainly used switches
and relays, which were originally designed for use in switching networks. In common
controls, they are operated frequently and so wear out earlier. In contrast, the life of an
electronic device is almost independent of its frequency of operation. This gave a
motivation for developing electronic common controls and resulted in electronic
Dept of. EEE

replacements for registers, markers, translators etc. having much greater reliability than
their electromechanical predecessors.
In electromechanical switching, the various functions of the exchange are
achieved by the operation and release of relays and switch (rotary or crossbar) contacts,
under the direction of a Control Sub-System. These contracts are hard - wired in a
predetermined way. The exchange dependent data, such as subscribers class of service,
translation and routing, combination signaling characteristics are achieved by hard-ware
and logic, by a of relay sets, grouping of same type of lines, strapping on Main or
Intermediate Distribution Frame or translation fields, etc. When the data is to be
modified, for introduction of a new service, or change in services already available to a
subscriber, the hardware change ranging from inconvenient to near impossible, are
involved.
In an SPC exchange, a processor similar to a general-purpose computer is used to
control the functions of the exchange. All the control functions, represented by a series of
various instructions, are stored in the memory. Therefore the processor memories hold all
exchange dependent data. such as subscriber date, translation tables, routing and charging
information and call records. For each call processing step. e.g. for taking a decision
according to class of service, the stored data is referred to, Hence, this concept of
switching. The memories are modifiable and the control program can always be rewritten
if the behavior or the use of system is to be modified. This imparts and enormous
flexibility in overall working of the exchange.
Digital computers have the capability of handling many tens of thousands of
instructions every second, Hence, in addition to controlling the switching functions the
same processor can handle other functions also. The immediate effect of holding both the
control programme and the exchange data, in easily alterable memories, is that the
administration can become much more responsive to subscriber requirements. both in
terms of introducing new services and modifying general services, or in responding to the
demands of individual subscriber. For example, to restore service on payment of an
overdue bill or to permit change from a dial instrument to a multi frequency sender,
simply the appropriate entries in the subscriber data-file are to be amended. This can be
done by typing- in simple instructions from a teletypewriter or visual display unit. The
ability of the administration to respond rapidly and effectively to subscriber requirements
is likely to become increasingly important in the future.
The modifications and changes in services which were previously impossible be
achieved very simply in SPC exchange, by modifying the stored data suitably. In some
cases, the subscribers can also be given the facility to modify their own data entries for
supplementary services, such as on-demand call transfer, short code (abbreviated)
dialing, etc.
The use of a central processor also makes possible the connection of local and
remote terminals to carry out man-machine dialogue with each exchange. Thus, the
maintenance and administrative operations of all the SPC exchanges in a network can be
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performed from a single centralized place. The processor sends the information on the
performance of the network, such as, traffic flow, billing information, faults, to the
centre, which carries out remedial measures with the help of commands. Similarly, other
modifications in services can also be carried out from the remote centre. This allows a
better control on the overall performance of the network.
As the processor is capable of performing operations at a very high speed, it has
got sufficient time to run routine test programmes to detect faults, automatically. Hence,
there is no need to carry out time consuming manual routine tests.
In an SPC exchange, all control equipment can be replaced by a single processor.
The processor must therefore be quite powerful, typically it must process hundreds of
calls per second, in addition to performing other administrative and maintenance tasks.
However, totally centralized control has drawbacks. The software for such a central
processor will be voluminous, complex, and difficult to develop reliably. Moreover, it is
not a good arrangement from the point of view of system security, as the entire system
will collapse with the failure of the processor. These difficulties can be overcome by
decentralizing the control. Some routine functions such as scanning, signal distributing,
marking, which are independent of call processing, can be delegated to auxiliary or
peripheral processors.
Stored program control (SPC) has become the principal type of control for all types
of new switching systems throughout the world, including private branch exchanges, data
and Telex systems. Two types of data are stored in the memories of electronic switching
systems. One type is the data associated with the progress of the call, such as the dialed
address of the called line.
Another type, known as the translation data, contains infrequently changing information,
such as the type of service subscribed to by the calling line and the information required
for routing calls to called numbers. These translation data, like the program, are stored in
a memory, which is easily read but protected to avoid accidental erasure. This
information may be readily changed, however, to meet service needs. The flexibility of a
stored program also aids in the administration and maintenance of the service so that
system faults may be located quickly.
SPC exchanges can offer a wider range of facilities than earlier systems. In
addition, the facilities provided to an individual customer can be readily altered by
changing the customers class-of-service data stored in memory. Moreover, since the
processors stored data can be altered electronically,some of these facilities can be
controlled by customers. Examples include:1. Call barring (outgoing or incoming): The customer can prevent unauthorized
calls being made and can prevent incoming calls when wishing to be left in peace.
2. Call waiting: The Call waiting service notifies the already busy subscriber of a
third party calling him.
3. Alarm calls: The exchange can be instructed to call the customer at a pre-arranged
time (e.g. morning alarm).
Dept of. EEE

4. Call Forwarding: The subscriber having such a feature can enable the incoming
calls coming to his telephone to be transferred to another number during his
absence.
5. Conference calls: Subscriber can set up connections to more than one subscriber
and conduct telephone conferences under the provision of this facility.
6. Dynamic Barring Facility: Subscriber having STD/ISD facilities can dynamically
lock such features in their telephone to avoid misuse. Registering and dialing a
secret code will extend such such a facility.
7. Abbreviated Dialing: Most subscribers very often call only limited group of
telephone numbers. By dialing only prefix digit followed by two selection digits,
subscribers can call up to 100 predetermined subscribers connected to any
automatic exchange. This shortens the process of dialing all the digits.
8. Malicious call Identification: Malicious call identification is done immediately
and the information is obtained in the print out form either automatically or by
dialing an identification code.
9. Do Not Disturb: This facility enables the subscriber to free himself from attending
his incoming calls. Using this facility the calls coming to the subscriber can be
routed to an operator position or to an answering machine. The operator position
or the machine can inform the calling subscriber that the called subscriber is
temporarily inaccessible. Today SPC is a standard feature in all the electronic
exchanges.

Implementation of Switching Network.

In an electronic exchange, the switching network is one of the largest sub-system
in terms of size of the equipment. Its main functions are Switching (setting up temporary
connection between two or more exchange terminations), Transmission of speech and
signals between these terminations, with reliable accuracy.
There are two types of electronic switching system. viz. Space division and Time
Division.
Space Division switching System
In a space Division Switching system, a continuous physical path is set up
between input and output terminations. This path is separate for each connection and is
held for the entire duration of the call. Path for different connections is independent of
each other. Once a continuous path has been established., Signals are interchanged
between the two terminations. Such a switching network can employ either metallic or
electronic cross points. Previously, usage of metallic cross-points using reed relays and
all were favored. They have the advantage of compatibility with the existing line and
trunk signaling conditions in the network.

Time Division Switching System

In Time Division Switching, a number of calls share the same path on time
division sharing basis. The path is not separate for each connection, rather, is shared
Dept of. EEE

sequentially for a fraction of a time by different calls. This process is repeated

periodically at a suitable high rate. The repetition rate is 8 KHz, i.e. once every 125
microseconds for transmitting speech on telephone network, without any appreciable
distortion. These samples are time multiplexed with staggered samples of other speech
channels, to enable sharing of one path by many calls. The Time Division Switching was
initially accomplished by Pulse Amplitude
Modulation (PAM) Switching. However, it still could not overcome the
performance limitations of signal distortion noise, cross-talk etc. With the advent of Pulse
Code Modulation (PCM), the PAM signals were converted into a digital format
overcoming the limitations of analog and PAM signals. PCM signals are suitable for both
transmission and switching. The PCM switching is popularly called Digital Switching.
Digital Switching Systems
A Digital switching system, in general, is one in which signals are switched in
digital form. These signals may represent speech or data. The digital signals of several
speech samples are time multiplexed on a common media before being switched through
the system.
To connect any two subscribers, it is necessary to interconnect the time-slots of
the two speech samples, which may be on same or different PCM highways. The
digitalized speech samples are switched in two modes, viz., Time Switching and Space
Switching. This Time Division Multiplex Digital Switching System is popularly known
as Digital Switching System.
The ESS No.1 system was the first fully electronic switching system but not
digital. But later came ESS No.4 system which was digital for trunk portion only. When
designed, the cost of A/D conversion (CODEC) on each subscriber line was seen as
prohibitive. So the ESS No.4 system was acting as a Trunk/Tandem exchange but not as
a local exchange. So the main difficulty for implementing a digital local exchange was
the implementation of the subscriber line interface. This was solved by the introduction
of Integrated Circuits, which made the digital local exchange economically feasible. This
implementation handles the following functions:
B-Battery feed
O-Over-voltage protection (from lightning and accidental power line contact)
R-Ringing
S-Supervisory Signaling
C-Coding (A/D inter conversion & low pass filtering)
H-Hybrid (2W to 4W conversion)
T-Testing the connectivity of Subscriber
Examples of digital exchanges (switching systems) include CDOT, OCB, AXE, EWSD,
5ESS etc.

Dept of. EEE

The general architecture of a Digital Switching System is depicted in

fig2General architecture of Digital Switching System
Nx2
Mbps
links

Subs interface

Digital Switch

Trunks interface
Other
exchanges

CONTROL
PROCESSOR
Other auxiliary inter faces
Such as,
(a) Tone generator
(b) Frequency receives
(c) Conference call facility
(d) CCS# 7 Protocol
Manager
(e) V 5.2 access manager

Operation &
Maintenance

Figure-2
The next evolutionary step was to move the PCM codec from the
exchange end of the customers line to the customers end. This provides digital
transmission over the customers line, which can have a number of advantages. Consider
data transmission. If there is an analog customers line, a modem must be added and data
can only be transmitted at relatively slow speeds. If the line is digital, data can be
transmitted by removing the codec (instead of adding a modem). Moreover, data can be
transmitted at 64 kbit/s instead of at, say, 2.4 kbit/s. Indeed, any form of digital signal
can be transmitted whose rate does not exceed 64 kbit/s.

Dept of. EEE

Dept of. EEE

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Local and trunk Network

Trunk Lines
The term Trunk Line in telecommunications refers to the high-speed connection
between telephone central offices in the Public Switched Telephone Network (PSTN).
Trunk lines are always digital. The wiring between central offices was originally just
pairs of twisted copper wire (the twists in the wiring prevented things known as crosstalk
and noise). Because it is expensive to string up (or lay trenches for buried cables), the
phone company researched ways in which to carry more data over the existing copper
lines. This was achieved by using time-division multiplexing. Later, when fiber-optic
technology became available, phone companies upgraded their trunk lines to fiber optics
and used statistical time-division multiplexing, synchronous digital heirarchy, coarse or
dense wave division multiplexing and optical switching to further improve transmission
speeds.
The signaling information exchanged between different exchanges via inter
exchange trunks for the routing of calls is termed as Inter exchange Signaling. Earlier in
band /out of band frequencies were used for transmitting signaling information. Later on,
with the emergence of PCM systems, it was possible to segregate the signaling from the
speech channel. A trunk line is a circuit connecting telephone switchboards (or other
switching equipment), as distinguished from local loop circuit which extends from
telephone exchange switching equipment to individual telephones or information
origination/termination equipment.
When dealing with a private branch exchange (PBX), trunk lines are the phone
lines coming into the PBX from the telephone provider. This differentiates these
incoming lines from extension lines that connect the PBX to (usually) individual phone
sets. Trunking saves cost, because there are usually fewer trunk lines than extension lines,
since it is unusual in most offices to have all extension lines in use for external calls at
once. Trunk lines transmit voice and data in formats such as analog, T1, E1, ISDN or
PRI. The dial tone lines for outgoing calls are called DDCO (Direct Dial Central Office)
trunks.
A signal travelling over a trunk line is not actually flowing any faster. The
electrical signal on a voice line takes the same amount of time to traverse the wire as a
similar length trunk line. What makes trunk lines faster is that the signal has been altered
to carry more data in less time using more advanced multiplexing and modulation
techniques. If you compared a voice line and a trunk line and put them side by side and
observed them, the first pieces of information arrive simultaneously on both the voice and
trunk line. However, the last piece of information would arrive sooner on the trunk line.
No matter what, you can't break the laws of physics. Electricity over copper or laser light
over fiber optics, you cannot break the speed of light--though that has rarely stopped
uneducated IT or IS managers from demanding that cabling perform faster instead of
upgrading equipment.

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Trunk lines can contain thousands of simultaneous calls that have been combined
using time-division multiplexing. These thousands of calls are carried from one central
office to another where they can be connected to a de-multiplexing device and switched
through digital access cross connecting switches to reach the proper exchange and local
phone number.

Local and trunk Network

s

TR

09

TR

TR

TR

CI

CT

S
L
TR
CID
CIA
CTI

: Remote line unit

: Local subscriber exchange
: Transit exchange
: Outgoing international exchange
: Incoming international exchange
: International transit exchange

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CI

What is Trunking?
In telecommunications systems, trunking is the aggregation of multiple user circuits into
a single channel. The aggregation is achieved using some form of multiplexing. Trunking
theory was developed by Agner Krarup Erlang, Erlang based his studies of the statistical
nature of the arrival and the length of calls. The Erlang B formula allows for the
calculation of the number of circuits required in a trunk based on the Grade of Service
and the amount of traffic in Erlangs the trunk needs cater for.

Definition
In order to provide connectivity between all users on the network one solution is to build
a full mesh network between all endpoints. A full mesh solution is however impractical, a
far better approach is to provide a pool of resources that end points can make use of in
order to connect to foreign exchanges. The diagram below illustrates the where in a
telecommunication network trunks are used.

A Modern Telephone Network Indicating where trunks are used. SLC - Subscriber
line concentrator

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LE

Local Exchange

TDM TAX II

Level II Tax

TDM TAX I

Level I Tax

Level I Taxs are connected to the Gateway.

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Call routing
Routing in the PSTN is the process used to route telephone calls across the public
switched telephone network. This process is the same whether the call is made between
two phones in the same locality, or across two different continents.

Relationship between exchanges and operators

Telephone calls must be routed across a network of multiple exchanges, potentially
owned by different telephone operators. The exchanges are all are inter-connected
together using trunks. Each exchange has many "neighbours", some of which are also
owned by the same telephone operator, and some of which are owned by different
operators. When neighbouring exchanges are owned by different operators, they are
known as interconnect points.
This means that there is really only one virtual network in the world that enables any
phone to call any other phone. This virtual network comprises many interconnected
operators, each with their own exchange network. Every operator can then route calls
directly to their own customers, or pass them on to another operator if the call is not for
one of their customers.
The PSTN is not a fully meshed network with every operator connected to every other that would be both impractical and inefficient. Therefore calls may be routed through
intermediate operator networks before they reach their final destination. One of the major
problems in PSTN routing is determining how to route this call in the most cost effective
and timely manner.
Call routing
Each time a call is placed for routing, the destination number (also known as the called
party) is entered by the calling party into their terminal. The destination number generally
has two parts, a prefix which generally identifies the geographical location of the
destination telephone, and a number unique within that prefix that determines the specific
destination terminal. Sometimes if the call is between two terminals in the same local
area (that is, both terminals are on the same telephone exchange), then the prefix may be
omitted.
When a call is received by an exchange, there are two treatments that may be applied:
Either the destination terminal is directly connected to that exchange, in which
case the call is placed down that connection and the destination terminal rings.
Or the call must be placed to one of the neighbouring exchanges through a
connecting trunk for onward routing.
Each exchange in the chain uses pre-computed routing tables to determine which
connected exchange the onward call should be routed to. There may be several
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alternative routes to any given destination, and the exchange can select dynamically
between these in the event of link failure or congestion.
The routing tables are generated centrally based on the known topology of the network,
the numbering plan, and analysis of traffic data. These are then downloaded to each
exchange in the telephone operators network. Because of the hierarchical nature of the
numbering plan, and its geographical basis, most calls can be routed based only on their
prefix using these routing tables.
Some calls however cannot be routed on the basis of prefix alone, for example nongeographical numbers, such as toll-free or freephone calling. In these cases the Intelligent
Network is used to route the call instead of using the pre-computed routing tables.
In determining routing plans, special attention is paid for example to ensure that two
routes do not mutually overflow to each other, otherwise congestion will cause a
destination to be completely blocked.
According to Braess' paradox, the addition of a new, shorter, and lower cost route can
lead to an increase overall congestion[. The network planner must take this into account
when designing routing paths.
One approach to routing involves the use of Dynamic Alternative Routing (DAR). DAR
makes use of the distributed nature of a telecommunications network and its inherent
randomness to dynamically determine optimal routing paths. This method generates a
distributed, random, parallel computing platform that minimises congestion across the
network, and is able to adapt to take changing traffic patterns and demands into account.
Routing can be loosely described as the process of getting from here to there. Routing
may be discussed in the context of telephone networks or computer networks. In
telephone networks, routing is facilitated by switches in the network, whereby in
computer networks routing is performed by routers in the network.
Definition: Routing in telephone networks
Routing in the context of telephone networks is the selection of a specific circiut group,
for a given call or traffic stream, at an exchange in the network . "The objective of
routing is to establish a successful connection between any two exchangesin the network"
. By selecting routes that meet the constraints set by the user traffic and the network,
routing determines which network resources (circuit group) should be used to transport
which user traffic.
Different networks employ different routing techniques, but all communication networks
share a basic routing functionality based on three core routing functions
Assembling and distributing information on the state of the network and user traffic that
is used to generate and select routes.

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Generating and selecting feasible and optimal routes based on network and user traffic
state information.
Forwarding user traffic along the selected routes.
The public switched telephone network (PSTN) architecture is made up of a hierarchy of
exchanges (e.g local and regoinal exchanges) with each level of the hierarchy performing
different functions . Two adjacent exchanges in the network may be connected by several
direct routes consisting of one or more circuits .
In circuit-switched networks, such as the PSTN, switching and transmission resources are
dedicated to a call along the path from source to destination for the complete duration of
the call. Routing decisions are imperative in facilitating this process as they determine the
most efficient links to use to connect users for a call . Routing in the PSTN is done using
a hop-by-hop approach . When a user wants to make a call, they dial the destination
number to which the call should be routed. This destination number is made up of a
prefix (area code or national destination network), which identifies the geographical
location of the called party, and a unique number (the subscriber number) linked to the
prefix that identifies the exact destination to which the call should be routed The end
exchange to which the calling party is connected (the originating exchange) uses the area
code to identify the outgoing circuit group connecting to the first choice adjacent
exchange en-route This circuit group is called the first choice route and is obtained using
a routing table at the originating switch . The function of the switch at the originating end
exchange is to connect the switch input port to which the calling user is connected to a
free outgoing circuit group in the first choice group . If all the circuits along the first
choice route are fully occupied, the switch then attempts to use an alternative route circuit
group to route the call to the destination exchange . The originating exchange then
forwards the address to the adjacent exchange (first choice or alternate route), and the
procedure is repeated at the adjacent exchange in order to reach the destination end
exchange to which the called party is connected . When the address reaches the
destination exchange, it only needs to process the last part of the address to identify the
switch input port that the called party is connected .
Routing directs forwarding . Forwarding of traffic can be done using connection-oriented
or connectionless approaches . In connection-oriented forwrding, forwarding instructions
are installed in all the switches along a designated route before the route can be used to
transport traffic . Traffic forwarded using the connectionless approach carries its own
forwarding information either as precise routing commands for each switch along a route
or as hints that may be autonomously interpreted by any switch in the network .
In PSTN, forwarding of traffic is based on the connection-oriented approach. Call routing
is achieved using pre-computed routing tables, containing all the possible pre-defined
routes for a connection, at each switch .The pre-defined routes specified in the routing
table include information of a direct route (or routes) to be used under normal traffic and
network conditions (e.g no link failure or network congestion) as well as alternative
routes that should be used in the event that all circuits along the direct route are fully
occupied . An alternative route may be an indirect route consisting of several circuit
groups connecting two exchanges via other exchanges . The following example illustrates
the use of an alternative route to connect two exchanges in the event of the direct route
being congetsed.

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A Typical Telephone Exchange -OCB-283

FUNCTIONAL ARCHITURE
The Alcatel E10 system is located at the heart of the telecommunication networks
concerned. It is made up of three independent functional units:
- The Subscriber Access Subsystem which carries out connection of analogue and
digital subscriber lines,
-

Connection and Control which carries out connections and processing of calls,

Operation and Maintenance which is responsible for all functions needed by the
network operating authority.

Each functional unit is equipped with softwares which are appropriate for handling the
functions for which it is responsible.
Synchronization and Time Base Station STS
Time base (BT)
The BT ensures times distribution for LR and PCM to provide the synchronization, and
also for working out the exchange clock.Time distribution is tripled.
Time generation can be either autonomous or slaved to an external rhythm with a view to
synchronise the system with the network
Auxiliary Equipment Control Station SMA
Auxiliary equipment manager (ETA)
The ETA Supports:
-

The tone generators (GT).

The frequency receiving and generation (RGF) devices,

Conference circuits (CCF),

The exchange clock

CCS7 protocol handler (PUPE) and CCS7 controller (PC): CCITT No. 7 protocol
processing
For connection of 64 kbit/s signaling channels, semi- permanent connections are
established via the connection matrix, to the PUPE which processes the CCITT No. 7
protocol.
More precisely, the PUPE function carries out the following:
-

signaling channel Level 2 processing,

the message routing function

(Part of Level 3). The PC carries out:

-

the network management function (part of Level 3),

PUPE defence,

Various observation tasks which are not directly linked to CCITT No. 7.

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18

CCITT N 7
SIGNALLING
NETWORK

SUBCRIBER
CONNECTION

ACCESS

AND
SUBSYSTE
M

CONTROL

TELEPHONE
NETWORK

DATA
NETWORK

NT
VALUE ADDED
NETWORK

OCB 283
OP E R AT I O N
AND
MAINTENANCE

PABX

ALCATEL 1000 E10

OCB 283

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OPERATION AND
MAINTENANCE
NETWORK

Host switching matrix (SMX)

The SMX is a square connection matrix with a single time stage, T, duplicated in full,
which enables up to 2048 matrix links (LR) to be connected.
A matrix link LR is an internal PCM, with 16 bits per channel (32 channels). The MCX
can execute the following:
1)

an unidirectional connection between any incoming channel and any out going
channel. There can be as many simultaneous connections as there are outgoing
channels. It should be remembered that a connection consists of allocating the
information contained within an incoming channel to an outgoing channel,

2)

connection between any incoming channel and any M outgoing channels,

3)

connection of N incoming channels belonging to one frame structure of any

multiplex onto N outgoing channels which belong to the same frame structure,
abiding to the integrity and sequencing of the frame received. This function is
referred to as connection with N x 64 kbit/s.

The MCX is controlled by the COM function (matrix switch controller) to ensure the:
-

set up and breakdown of the connections by access to the matrix command memory.
This access is used to write at the output T.S. address the incoming T.S. address

defense of the connections. Security of the connections in order to assure a good data
switching.

Truck Control Station SMT

PCM controller (URM)
The URM provides the interface between external PCMs and the OCB283. These PCM
come from either:
- a remote subscriber digital access unit (CSN) or from a remote electronic
concentrator CSE,
-

another switching centre, on channel-associated signalling or CCITT No.7,

the digital recorded announcement equipment

satellite

In particular, the URM carries out the following functions:

-

HDB3 conversion to binary (PCM

matrix link),

binary conversion to HDB3 (matrix link

extraction and pre-processing of the channel-associated signalling of T.S.16 (PCM

command),

transmission of channel-associated signalling in T.S.16 (command

PCM),

PCM).

Main Control Station SMC

Call handler (MR)
The MR is responsible for the establishment and breaking off of communications.

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The call handler takes the decisions necessary for processing of communications in terms
of the signaling received, after consultation of the subscriber and analysis database
manager (TR) if necessary. The call handler processes new calls and handling-up
operations, releases equipment, commands switching on and switching off etc.
In addition, the call handler is responsible for different management tasks (control of tests
of circuits, sundry observations).
Operation and maintenance function (OM) SMM
The functions of the operation and maintenance subsystem are carried out by the
operation and maintenance software OM).
The operating authority accesses all hardware and software equipment of the Alcatel
1000 E10 system via computer terminals belonging to the operation and maintenance
subsystem: consoles, magnetic media, intelligent terminal. These functions can be
grouped into 2 categories:
-

operation of the telephone application,

operation and maintenance of the system.

In addition, the operation and maintenance subsystem carries out:

-

loading of softwares and of data for connection and command and for the
subscriber digital access units,
temporary backup of detailed billing information,

centralisation of alarm data coming from connection and control stations, via
alarm rings,

central defence of the system.

Finally, the operation and maintenance subsystem permits two-way communication with
operation and maintenance networks, at regional or national level (TMN).
CSN - digital satellite center
The digital satellite center [CSN center satellite numerique) is a subscriber connection
unit on which both analogue and digital subscribers can be connected.
Its design and composition enable the CSN to fit into an existing network and can be
connected to time-based systems using the CCITT N 7 type of semaphore signalling.
The CSN is a connection unit designed to adapt to a variety of geographical situation: it
can be either local [CSNL] or distant [CSND] with respect to the connecting switch.

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A Typical Telephone Exchange -OCB-283

CS
SMX
SMT
( 1 TO 28) X 2

Circuits and
announcemen
t machine

SMA
( 2 TO 37)
1 TO 4 MAS

SMC
2 TO 14

SMM
1x2

CSN

Digital satellite center

SMC :

Main Control Station

SMA :

Auxiliary Equipment Control Station

SMT :

Truck Control Station

SMX :

Matrix Control Station

SMM :

Maintenance Station

STS

Synchronization and Time Base Station

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STS
1x3

Chapter-2
Overview of Telecommunications Networks II
Institutional mechanism and role
Introduction: All industries operate in a specific environment which keeps changing and
the firms in the business need to understand it to dynamically adjust their actions for best
results.
Like minded firms get together to form associations in order to protect their common
interests. Other stake holders also develop a system to take care of their issues.
Governments also need to intervene for ensuring fair competition and the best value for
money for its citizens. This handouts gives exposure on the Telecom Environment in
India and also dwells on the role of international bodies in standardizing and promoting
Telecom Growth in the world.
Institutional Framework: It is defined as the systems of formal laws, regulations, and
procedures, and informal conventions, customs, and norms, that broaden, mold, and
restrain socio-economic activity and behaviour. In India, The Indian telegraph act of 1885
amended from time to time governs the telecommunications sector. Under this act, the
government is in-charge of policymaking and was responsible for provisioning of
services till the opening of telecom sector to private participation. The country has been
divided into units called Circles, Metro Districts, Secondary Switching Areas (SSA),
Long Distance Charging Area (LDCA)

and Short Distance Charging Area (SDCA).

Major changes in telecommunications in India

began in the 1980s. The initial phase of telecom reforms began in 1984 with the creation
of Center for Department of Telematics (C-DOT) for developing indigenous technologies
and private manufacturing of customer premise equipment. Soon after, the Mahanagar
Telephone Nigam Limited (MTNL) and Videsh Sanchar Nigam Limited (VSNL) were
set up in 1986.
The Telecom Commission was established in 1989. A crucial aspect of the institutional
reform of the Indian telecom sector was setting up of an independent regulatory body in
1997 the Telecom Regulatory Authority of India (TRAI), to assure investors that the
sector would be regulated in a balanced and fair manner. In 2000, DoT corporatized its

services

wing and created Bharat Sanchar Nigam Limited. Further changes in the

regulatory system took place with the TRAI Act of 2000 that aimed at restoring
functional clarity and improving regulatory quality and a separate disputes settlement
body was set up called Telecom Disputes Settlement and Appellate Tribunal (TDSAT) to
fairly adjudicate any dispute between licensor and licensee, between service provider,
between service provider and a group of consumers. In October 2003, Unified Access
Service Licenses regime for
basic and cellular services was introduced. This regime enabled services providers to
offer fixed and mobile services under one license. Since then, Indian telecom has seen
unprecedented customer growth crossing 600 million connections. India is the fourth
largest telecom market in Asia after China, Japan and South Korea. The Indian telecom
network is the eighth largest in the world and the second largest among emerging
economies. A brief on telecom echo system and various key elements in institutional
framework is given below:

Department of Telecommunications: In India, DoT is the nodal agency for taking care
of telecom sector on behalf of government. Its basic functions are:
Policy Formulation
Review of performance

Licensing
Wireless spectrum management
Administrative monitoring of PSUs
Research & Development
Standardization/Validation of Equipment
International Relations
Main wings within DoT:
Telecom Engineering Center (TEC)
USO Fund
Wireless Planning & Coordination Wing (WPC)
Telecom Enforcement, Resource and Monitoring (TERM) Cell
Telecom Centers of Excellence (TCOE)
Public Sector Units
Bharat Sanchar Nigam Limited(BSNL) Indian
Telephone Industries Limited (ITI) Mahanagar
Telephone Nigam Limited(MTNL)
Telecommunications Consultants India Limited(TCIL)
R & D Unit
Center for development of Telematics (C-DoT)
The other key governmental institutional units are TRAI & TDSAT. Important units are
briefed below:
Telecom Engineering Center (TEC): It is a technical body representing the interest of
Department of Telecom, Government of India. Its main functions are:
Specification of common standards with regard to Telecom networkequipment, services
and interoperability.
Generic Requirements (GRs), Interface Requirements (IRs)
Issuing Interface Approvals and Service Approvals
Formulation of Standards and Fundamental Technical Plans
Interact with multilateral agencies like APT, ETSI and ITU etc. for standardisation
Develop expertise to imbibe the latest technologies and results of R&D Provide
technical support to DOT and technical advice to TRAI & TDSAT Coordinate
with C-DOT on the technological developments in the Telecom Sector

for policy planning by DOT www.tec.gov.in

Universal Service Obligation Fund (USO): This fund was created in 2002. This fund is
managed by USO administrator. All telecom operators contribute to this fund as per
government policy. The objective of this fund is to bridge the digital divide i.e. ensure
equitable growth of telecom facilities in rural areas. Funds are allocated to operators who
bid lowest for providing telecom facilities in the areas identified by USO administrator.
WIRELESS PLANNING & COORDINATION (WPC) This unit was created in 1952 and

is the National Radio Regulatory Authority responsible for Frequency Spectrum

Management, including licensing and caters for the needs of all wireless users
(Government and Private) in the country. It exercises the statutory functions of the
Central Government and issues licenses to establish, maintain and operate wireless
stations. WPC is divided into major sections like Licensing and Regulation (LR), New
Technology Group (NTG) and Standing Advisory Committee on Radio Frequency
Allocation (SACFA). SACFA makes the recommendations

on major frequency

allocation issues, formulation of the frequency allocation plan, making

recommendations on the various issues related to International Telecom Union (ITU), to
sort out problems referred to the committee by various wireless users, Siting clearance of
all wireless installations in the country etc.
Telecom Enforcement, Resource and Monitoring (TERM) Cell: In order to ensure
that service providers adhere to the licence conditions and for taking care of telecom
network security issues, DoT opened these cells in 2004 and at present 34 cells are
operating in various Circles and big districts in the country. Key functions of these units
are Inspection of premises of Telecom and Internet Service Providers, Curbing illegal
activities in telecom services, Control over clandestine / illegal operation of telecom
networks by vested interests having no license, To file FIR against culprits, pursue the
cases, issue notices indicating violation of conditions of various Acts in force from time
to time, Analysis of call/subscription/traffic data of various licensees, arrangement for
lawful interception / monitoring of all communications passing through the licensees
network, disaster management, network performance monitoring, Registration of OSPs
and Telemarketers in License Service Areas etc..
Telecom Centers of Excellence (TCOE): (www.tcoe.in) The growth of Indian

Telecommunications sector has been astounding, particularly in the last decade. This
growth has been catalysed by telecommunications sector liberalization and reforms.
Some of the areas needing immediate attention to consolidate and maintain the growth
are:
Capacity building for industry talent pool
Continuous adaptation of the regulatory environment to facilitate induction/
adoptation of high potential new technologies and business models
Bridging of high rural - urban teledensity/digital divide
Faster deployment of broadband infrastructure across the country
Centres of Excellence have been created to work on (i) enhancing talent pool,
(ii)technological innovation, (iii) secure information infrastructure and (iv) bridging of
digital divide. These COEs are also expected to cater to requirements of South Asia as
regional leaders. The main sponsor (one of the telecom operators), the academic institute
where the Centers are located and the tentative field of excellence are enumerated in the
table below:
Field of Excellence in Telecom

Associated Institute

Sponsor

Next Generation Network & Network

Technology
Telecom Technology &
Management
Technology Integration, Multimedia &
Computational Maths
Telecom Policy, Regulation, Governance,
Customer Care & Marketing
Telecom Infrastructure & Energy

IIT, Kharagpur

Vodafone Essar

IIT ,Delhi

Bharti Airtel

IIT, Kanpur

BSNL

IIM, Ahmedabad

IDEA Cellular

IIT, Chennai

Reliance

Disaster Management of Info systems &

Information Security
Rural Application

IISc, Bangalore

Aircel

IIT Mumbai

Tata Telecom

Spectrum Management (Proposed)

WPC, Chennai

Govt with
Industry
consortium

Telecom Regulatory Authority of India (TRAI): TRAI was established under TRAI
Act 1997 enacted on 28.03.1997. The act was amended in 2000. Its Organization setup
consists of One Chairperson, Two full-time members & Two part-time members. Its
primary role is to deals with regulatory aspects in Telecom Sector & Broadcasting and
Cable services.
TRAI has two types of functions as mentioned below:
Mandatory Functions
Tariff policies
Interconnection policies
Quality of Service
Ensure implementation of terms and conditions of license
Recommendatory Functions
New license policies
Spectrum policies
Opening of sector
www.trai.gov.in
Telecom Dispute Settlement Appellate Tribunal (TDSAT): TDSAT was established in
year 2000 by an amendment in TRAI act by transferring the functions of dispute handling
to new entity i.e. TDSAT. The organization setup consists of one Chairperson & two fulltime members. Its functions are:
Adjudicate any dispute between
licensor and licensee
two or more licensees
group of consumers
Hear & dispose off appeal against any direction, decision or order of the Authority
under TRAI Act www.tdsat.nic.in
Key International Standardization Bodies for Telecom sector:
ITU is the leading United Nations agency for information and communication technology
issues, and the global focal point for governments and the private sector in developing
networks and services. For nearly 145 years, ITU has coordinated the shared global use
of the radio spectrum, promoted international cooperation in assigning satellite orbits,

worked to improve telecommunication infrastructure in the developing world, established

the worldwide standards that foster seamless interconnection of a vast range of
communications systems and addressed the global challenges of our times, such as
mitigating climate change and strengthening cybersecurity. Vast spectrum of its work
area includes broadband Internet to latest-generation wireless technologies, from
aeronautical and maritime navigation to radio astronomy and satellite-based meteorology,
from convergence in fixed-mobile phone, Internet access, data, voice and TV
broadcasting to next-generation networks. ITU also organizes worldwide and regional
exhibitions and forums, such as ITU TELECOM WORLD, bringing together the most
influential representatives of government and the telecommunications and ICT industry
to exchange ideas, knowledge and technology for the benefit of the global community,
and in particular the developing world. ITU is based in Geneva, Switzerland, and its
membership includes 191 Member States and more than 700 Sector Members and
Associates. On 1 January 2009, ITU employed 702 people from 83 different countries.
The staff members are distributed between the Union's Headquarters in Geneva,
Switzerland and eleven field offices located around the world. www.itu.int
Asia Pacific Telecommunity: Headquartered at Bangkok, the APT is a unique
organization of Governments, telecom service providers, manufactures of communication
equipment,research & development organizations and other stake holders active in the
field of communication and information technology. APT serves as the focal organization
for communication and information technology in the Asia Pacific region. The APT has
34 Members, 4 Associate Members and 121 Affiliate Members. The objective of the
Telecommunity is to foster the development of telecommunication services and
information infrastructure throughout the region with a particular focus on the expansion
thereof in less developed areas. APT has been conducting HRD Programme for
developing the skills of APT Members to meet the objectives of APT. The topics include
Information
Communication Technologies (ICT), Network and Information Security, Finance and
Budget,Telecommunication

Management,

Mobile

Communications,

Multimedia,

Satellite Communication, Telecommunications and ICT Policy and Regulation,

Broadband Technologies, e-Applications, Rural Telecommunications Technologies, IP
Networks and Services, Customer Relations, etc. www.aptsec.org
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The

European

Telecommunications

Standards

Institute

(ETSI)

produces

globallyapplicable standards for Information and Communications Technologies (ICT),

including fixed, mobile, radio, converged, broadcast and internet technologies. It is
officially recognized by the European Union as a European Standards Organization. ETSI
is a not-forprofit organization with more than 700 ETSI member organizations drawn
from 62 countries across 5 continents world-wide. ETSI unites Manufacturers, Network
operators, National Administrations , Service providers, Research bodies, User groups ,
Consultancies. This cooperation has resulted in a steady stream of highly successful ICT
standards in mobile, fixed, and radio communications and a range of other standards that
cross these boundaries, including Security, Satellite, Broadcast, Human Factors, Testing
& Protocols, Intelligent transport, Power-line telecoms, eHealth, Smart Cards,
Emergency communications, GRID & Clouds, Aeronautical etc. ETSI is consensus-based
and conducts its work through

Technical Committees, which produce standards and

specifications, with the ETSI General Assembly and Board. www.etsi.org

BSNL: Bharat Sanchar Nigam Limted was formed in year 2000 and took over the service
providers role from DoT. Today, BSNL has a customer base of over 9 crore and is the
fourth largest integrated telecom operator in the country. BSNL is the market leader in
Broadband, landline and national transmission network. BSNL is also the only operator
covering over 5 lakh village with telecom connectivity. Area of operation of BSNL is all
India except Delhi &Mumbai.
MTNL: Mahanagar Telephone Nigam Limited, formed in 1984 is the market leader in
landline and broadband in its area of operation. www.mtnl.net.in
TCIL: TCIL, a prime engineering and consultancy company, is a wholly owned
Government of India Public Sector Enterprise. TCIL was set up in 1978 for providing
Indian telecom expertise in all fields of telecom, Civil and IT to developing countries
around the world. It has its presence in over 70 countries. www.tcil-india.com
ITI: Indian telephone Industries is the oldest manufacturing unit for telephone
instruments. To keep pace with changing times, it has started taking up manufacturing of
new technology equipment such as GSM, OFC equipment, Invertors, Power plants,
Defense equipments, Currency counting machines etc. www.itiltd-india.com
Centre for Development of Telematics (CDoT): This is the R & D unit under DoT setup
in 1984. The biggest contribution of this centre to Indian telecom sector is the
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development of low capacity (128 port) Rural automatic Exchange (RAX) which enabled
provisioning of telephone in even the smallest village. This was specially designed to suit
Indian environment, capable of withstanding natural temperature and dusty conditions.
Prominent Licenses provided by DoT:
Access Service (CMTS & Unified Access Service): The Country is divided into 23
Service Areas consisting of 19 Telecom Circle Service Areas and 4 Metro Service Areas
for providing Cellular Mobile Telephone Service (CMTS). Consequent upon
announcement of guidelines for Unified Access (Basic& Cellular) Services licenses on
11.11.2003, some of the CMTS operators have been permitted to migrate from CMTS
License to Unified Access Service License (UASL). No new CMTS and Basic service
licenses are being awarded after issuing the guidelines for Unified access Service
Licence(UASL). As on 31st March 2008, 39 CMTS and 240 UASL licenses operated.
o 3G & BWA (Broadband Wireless Access): Department of Telecom started the auction
process for sale of spectrum for 3G and BWA (WiMax) in April 2010 for 22 services
areas in the country. BSNL & MTNL have already been given spectrum for 3G and
BWAand they need to pay the highest bid amount as per auction results. BSNL & MTNL
both are providing 3G services. BSNL has rolled out its BWA service by using WiMax
technology.
Mobile Number Portability (MNP) Service: Licenses have been awarded to two
perators to provide MNP in India. DoT is ensuring the readiness of all mobile operators
and expects to start this service any time after June 2010.
Infrastructure Provider: There are two categories IP-I and IP-II. For IP-I the applicant
company is required to be registered only. No license is issued for IP-I. Companies
registered as IP-I can provide assets such as Dark Fibre, Right of Way, Duct space and
Tower. This was opened to private sector with effect from 13.08.2000. An IP-II license
can lease / rent out /sell end to end bandwidth i.e. digital transmission capacity capable to
carry a message. This was opened to private sector with effect from 13.08.2000. Issuance
of IP-II Licence has been discontinued w.e.f. 14.12.05
INMARSAT : INMARSAT (International Maritime Satellite Organisation) operates a
constellation of geo-stationary satellites designed to extend phone, fax and data
communications all over the world. Videsh Sanchar Nigam Ltd (VSNL) is permitted
to provide Inmarsat services in India under their International Long Distance(ILD)
31

Dept of. EEE

licence granted by Department of Telecommunications(DoT). VSNL has commissioned

their new Land Earth Station (LES) at Dighi, Pune compatible with 4th generation
INMARSAT Satellites (I-4) and INMARSAT-B, M, Mini-M & M-4 services are now
being provided through this new LES after No Objection Certificate (NOC) is issued by
DoT on case by case basis.
National Long Distance: There is no limit on number of operators for this service and
license is for 20 years.
International Long Distance: This was opened to private sector on 1st April 2002 with
no limit on number of operators. The license period is 20 years.
Resale of IPLC: For promoting competition and affordability in International Private
Leased Circuits (IPLC) Segment, Government permitted the Resale of IPLC by
introducing a new category of License called as Resale of IPLC Service License
witheffect from 24th September 2008. The Reseller can provide end-to-end IPLC
between India and country of destination for any capacity denomination. For providing
the IPLC service, the Reseller has to take the IPLC from International Long Distance
(ILD) Service Providers licensed and permitted to enter into an arrangement for leased
line with Access Providers, National Long Distance Service Providers and International
Long Distance Service Providers for provision of IPLC to end customers.
Sale of International Roaming SIM cards /Global Calling Cards in India: The
cards being offered to Indian Customers will be for use only outside India. However, if it
is essential to activate the card for making test calls/emergent calls before the departure
of customer and /or after the arrival of the customer, the same shall be permitted for forty
eight (48) hours only prior to departure from India and twenty four (24) hours after
arrival in India.
Internet without Telephony: The Internet Service Provider (ISP) Policy was announced
in November, 98. ISP Licenses , which prohibit telephony on Internet ,are being
issued starting from 6.11.98 on non-exclusive basis. Three category of license exist
namely A,B and C. A is all India, B is telecom Circles, Metro Districts and major districts
where as C is SSA wide.
Internet with Telephony: Only ISP licensees are permitted, within their service
area, to offer Internet Telephony service. The calls allowed are PC to PC in India, PC in
India to PC/Telephone outside India, IP based calls from India to other countries.
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VPN: Internet Service Providers (ISPs) can provide Virtual Private Network (VPN)
Services. VPN shall be configured as Closed User Group(CUG) only and shall carry only
the traffic meant for the internal use of CUG and no third party traffic shall be carried on
the VPN. VPN shall not have any connectivity with PSTN / ISDN / PLMN except when
the VPN has been set up using Internet access dial-up facility to the ISP node. Outward
dialing facility from ISP node is not permitted.
VSAT & Satellite Communication: There are two types of CUG VSAT licenses : (i)
Commercial CUG VSAT license and (ii) Captive CUG VSAT license. The commercial
VSAT service provider can offer the service on commercial basis to the subscribers by
setting up a number of Closed User Groups (CUGs) whereas in the captive VSAT
service only one CUG can be set up for the captive use of the licensee. The scope of the
service is to provide data connectivity between various sites scattered within territorial
boundary of India via INSAT Satellite System using Very Small Aperture
Terminals (VSATs). However, these sites should form part of a Closed User Group
(CUG). PSTN connectivity is not permitted.
Radio Paging: The bids for the Radio Paging Service in 27 cities were invited in 1992,
the licenses were signed in 1994 and the service was commissioned in 1995. There was
a provision for a fixed license fee for first 3 years and review of the license fee
afterwards.The license was for 10 years and in 2004 Govt offered a extended 10 years
license with certain license fee waivers but with the wide spread use of mobile phones,
this service has lost its utility.
PMRTS: Public Mobile Radio Trunking service allows city wide connectivity through
wireless means. This service is widely used by Radio Taxi operators and companies
whose workforce is on the move and there is need to locate the present position of
employee for best results. PSTN connectivity is permitted.
INSAT MSS: INSAT Mobile Satellite System Reporting Service (INSAT MSS
Reporting Service) is a one way satellite based messaging service available through
INSAT. The basic nature of this service is to provide a reporting channel via satellite to
the group of people, who by virtue of their nature of work are operating from remote
locations without any telecom facilities and need to send short textual message or short
data occasionally to a central station.
Voice Mail/ Audiotex/ UMS (Unified Messaging Service): Initially a seprate license
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was issued for these services. For Unified Messaging Service, transport of Voice Mail
Messages to other locations and subsequent retrieval by the subscriber must be on a
nonreal time basis. For providing UMS under the licence, in addition to the licence for
Voice Mail/Audiotex/UMS, the licensee must also have an ISP licence. The ISP licence
as well as Voice Mail/Audiotex/ UMS licence should be for the areas proposed to be
covered by UMS service. Since start of NTP-99, all access provider i.e. CMTS, UASL,
Fixed service providers are also allowed to provide these services as Value Added
Service (VAS) under their license conditions.
Telemarketing: Companies intending to operate as Telemarketes need to obtain this
license from DoT. Other Service Provider (including BPO): As per New Telecom Policy
(NTP) 1999, Other Service Providers (OSP), such as tele-banking, tele-medicine, teletrading, ecommerce, Network Operation Centers and Vehicle Tracking Systems etc are
allowed to operate by using infrastructure provided by various access providers for nontelecom services.
Telecom Operators: Interested companies obtain license for various services to get
authorization to provide licensed telecom services in India. While hundreds of license
holders exists in India for various services, major operators are BSNL, Bharti (Airtel),
Vodafone, Reliance, Aircel, Idea and Tata etc. There is a stiff competition in the market
and operators struggle to provide innovative services earlier than others, at rates lower
than rivals, continuously find ways to extend better customer care and improve profit
margins by managing costs. A typical diagram depicting various macro level activities
performed by a telecom service provider is given below:

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In todays fast growing customer base in telecom market, rising expectations of

customers for prompt service support, very efficient & powerful software solutions are a
must. For this purpose, over the years, OSS (Operations Support Systems) & BSS
(Business Support Systems) software solutions have been developed to manage these
activities. The term OSS most frequently describes "network systems" dealing with the
telecom network itself, supporting processes such as maintaining network inventory,
provisioning services, configuring network components, and managing faults. Business
Support Systems or BSS typically refers to "business systems" dealing with customers,
supporting processes such as taking orders, processing bills, and collecting payments.
The two systems together are often abbreviated BSS/OSS or simply B/OSS. Many
proprietary software solutions are available from different vendors. A standardization
initiative has been taken up by Telecom Management forum, an international
membership organization of communications service providers and suppliers to the
communications industry. TM Forum is regarded as the most authoritative source for
standards and frameworks in OSS. TM Forum has been active in proving a framework
and discussion forum for advancements in OSS and BSS. A typical architecture of
OSS/BSS application is given below:

Optical-OFC, DWDM etc., Transport-SDH,PDH, ATM,PSTN, DSL etc., IP-MPLS,

Internet, IP TV, Multicast etc., Fixed/Wireless-PSTN, GSM, CDMA, WiMax, 3G etc.,
System-Windows, Unix etc.
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Sector Specific industry associations:

The Cellular Operators Association of India (COAI) was constituted in 1995 as a
registered, non-profit, non-governmental society dedicated to the advancement of
communication, particularly modern communication through Cellular Mobile Telephone
Services. COAI represents Indian Cellular industry and on its behalf it interacts with the
policy maker, the licensor, the regulator, the spectrum management agency and the
industry (telecom /non-telecom) associations. www.coai.com
Key Objectives of the COAI
To improve standards and competitiveness in the Cellular Industry and attain the
status of world class infrastructure. To facilitate affordable mobile telephony services
for Indians.
To study the best practices & research of the industry as well as to analyse the
Cellular Experience worldwide.
To assist relevant authorities by providing them information about the industry to help
them formulate suitable policies for the industry's growth.
To improve standards and quality of services in consultation with GSM India - the
Indian chapter of the GSM Association.
To maintain and upgrade services in terms of speech transmission, access, coverage,
security etc, to enable expansion of cellular services.
To help address problems of cellular operators relating to operational, regulatory,
financial, or licensing through interaction with the Ministry of Communications & IT,
Ministry of Finance, Ministry of Commerce, Department of telecommunications,
Telecom Regulatory Authority of India, Financial Institutions etc.
Association of Unified Telecom Service Providers of India (AUSPI) is the representative
industry body of Unified Access Service Licensees providing CDMA & GSM Mobile
Services, Fixed Line Services as well as Value Added Services throughout the length and
breadth of the country. AUSPI is a registered society and works as a non-profit
organization with the aim of delivering the promise of improved Access, Coverage and
Teledensity in India. The objectives of the Association include collection and
dissemination of knowledge and information for promotion and healthy growth of
telecom services, enunciating a telecom vision for India, fueling unprecedented domestic
investment, improving teledensity and bringing value for customers. The Association
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36

interacts on policy and regulatory issues with various Government bodies such as the
Department of Telecommunications, Telecom Regulatory Authority of India, apex
industry organizations like ASSOCHAM, Confederation of Indian Industry (CII) and
Federation of Indian Chambers of Commerce & Industry (FICCI), technical institutions,
financial analysts and other institutions of world repute. The Association formulates
expert opinion on industry issues and submits whenever necessary, recommendations to
the concerned authorities. www.auspi.in
TEMA Established in 1990, Telecom Equipment Manufacturers Association of
India
(TEMA) is recognized by the Government of India as the National Apex body to
represent telecom Technology Providers, Global and Indian, Private and Government
owned companies. TEMA has membership of more than 150 member companies
covering almost 80 per cent of Indian Telecom Equipment Manufacturing. Services
offered to TEMA members include, interaction with Government, Policy makers,
interaction with various National Confederations of Industries, overseas Delegations,
Exhibition Organizers, Market Development Assistance Authorities, Tender Information,
Excise and Customs Departments, Telecom Engineering Center for product specifications
etc. Our members are exporting a variety of Telecom Equipments to South America,
Middle-East, Africa, SAARC, CIS andSouth East Asian Countries. TEMA also has an
Export Promotion Forum set up by theMinistry of Commerce, Government of India to
promote Export of Telecom Equipments and Services. The Forum also make various
recommendations to the Government for making necessary changes in various policies
and procedures for promotion of Exports and Services.
Key Industry/ Trade Associations influencing the Telecom Market
The Confederation of Indian Industry (CII) works to create and sustain an environment
conducive to the growth of industry in India, partnering industry and government alike
through advisory and consultative processes.CII is a non-government, not-for-profit,
industry led and industry managed organisation, playing a proactive role in India's
development process. Founded over 115 years ago, it is India's premier business
association, with a direct membership of over 7800 organisations from the private as well
as public sectors, including SMEs and MNCs, and an indirect membership of over 90,000
companies from around 396 national and regional sectoral associations. With 64 offices
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in India, 9 overseas in Australia, Austria, China, France, Germany, Japan, Singapore,

UK, and USA, and institutional partnerships with 221 counterpart organisations in 90
countries, CII serves as a reference point for Indian industry and the international
business community. www.cii.in
The Associated Chambers of Commerce and Industry of India (ASSOCHAM), India's
premier apex chamber covers a membership of over 2 lakh companies and professionals
across the country. It was established in 1920 by promoter chambers, representing all
regions of India. As an apex industry body, ASSOCHAM represents the interests of
industry and trade, interfaces with Government on policy issues and interacts with
counterpart

international

organizations

to

promote

bilateral

economic

issues.

ASSOCHAM is represented on all national and local bodies and is, thus, able to proactively convey industry viewpoints, as also communicate and debate issues relating to
public-private partnerships for economic development. www.assocham.org
FICCI: Established in 1927, FICCI is the largest and oldest apex business organisation in
India. FICCI plays a leading role in policy debates that are at the forefront of social,
economic and political change. Its publications are widely read for their in-depth research
and policy prescriptions. FICCI works closely with the government on policy issues,
enhancing efficiency, competitiveness and expanding business opportunities for industry
through a range of specialised services and global linkages. It also provides a platform for
sector specific consensus building and networking. www.ficci.com
Job opportunities in Telecom Sector
Government sector: Every year UPSC conducts Indian Engineering Services exam for
recruitment to fill up vacancies notified by various departments such as Broadcasting,
Military Engineering Service, Indian Telecom Service, Indian Railways, Wireless
Planningetc. Numbers of vacancies vary year to year.
Entry level engineers with Telecom Operators: All operators recruit thousands on fresh
engineers every year owing to the high growth in telecom market. BSNL recruits of the
order of thousand fresh graduates every year at Junior Telecom Officer level.
Sales Engineers: Many Telecom solutions are very sophisticated and technical. Such
sales need to be handled by telecom engineers.

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Manufacturing Sector: Most of the MNCs have set up factories in India for
manufacturing telecom network equipment as well as Customer premises equipment.
There is enough job potential with these firms.
Support jobs in Non-Telecom sector: In todays scenario, all industries use many
telecom facilities for faster and efficient communication. All such activities require
maintenance professionals. Even in medical sector, growing use of telemedicine has
created a new marketfor telecom professionals.
Research & Development: Many MNCs have outsourced R & D in telecom to Indian
firms.
For example Nokia has outsourced its product design to M/s TCS. All such deals create
jobopportunities for telecom engineers.
IT sector: The core of BPO sector is the telecom network. IT sectors generates huge
telecomjobs.
Education sector: Government of Indias mission mode project on Education such as
Sarva shiksha Abhiyan, connecting all libraries in India, providing broadband to all
schools etc. requires telecom professionals to install and manage this huge network.
National E-Governance Project: The ambitions plan of India to network each nook &
corner of the country and provide a citizen centric, single window service counter
requires creation of vast telecom network across the country. Each State is implementing
State Wide Area Network (SWAN). All such projects create demand for telecom
professionals.
Research executives with Consultancy Firms: Telecom growth impacts a countrys
economy.
Many consultancy firms thrive on generating reports on business models, future potential
andextending guidance to existing and new entrants in telecom market. There is a
significant need for telecom professionals with such firms also.
Pay range: Entry level engineer can get a starting annual package ranging from 2-4 lakh
depending on the nature of job & employer firm.

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Chapter-3
PCM PRINCIPLES
INTRODUCTION
A long distance or local telephone conversation between two persons could be provided
by using a pair of open wire lines or underground cable as early as early as mid of 19th
century. However, due to fast industrial

development

and

an

increased

telephone

awareness, demand for trunk and local traffic went on increasing at a rapid rate. To cater
to the increased demand of traffic between two stations or between two subscribers at the
same station we resorted to the use of an increased number of pairs on either the open
wire alignment, or in underground cable. This could solve the problem for some time
only as there is a limit to the number of open wire pairs that can be installed on one
alignment

due

to

headway

consideration

and

maintenance

problems. Similarly increasing the number of open wire pairs that can be installed on one
alignment due to headway consideration and maintenance problems. Similarly increasing
the number of pairs to the underground

cable

is

uneconomical

and

leads

to

maintenance problems.
It, therefore, became imperative to think of new technical innovations

which could

exploit the available bandwidth of transmission media such as open wire lines or
underground cables to provide more number of circuits on one pair. The technique used
to provide a number of circuits using a single transmission link is called Multiplexing.
MULTIPLEXING TECHNIQUES
There are basically two types of multiplexing techniques
i.
ii

Frequency Division Multiplexing (FDM)

Time Division Multiplexing (TDM)

Frequency Division Multiplexing Techniques (FDM)

The FDM techniques is the process of translating individual speech circuits (300-3400
Hz) into pre-assigned frequency slots within the bandwidth of the transmission medium.
The frequency translation is done by amplitude modulation of the audio frequency with
an appropriate carrier frequency. At the output of the modulator a filter network is
connected to select either a lower or an upper side band. Since the intelligence is carried
in either side band, single side band suppressed carrier mode of AM is used. This results
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in substantial saving of bandwidth mid also permits the use of low power amplifiers.
Please refer Fig. 1.

FDM techniques usually find their application in analogue transmission systems. An

analogue transmission system is one which is used for transmitting continuously varying
signals.
Time Division Multiplexing
Basically,

time

division

multiplexing

involves

nothing

more

than

sharing

a transmission medium by a number of circuits in time domain by establishing a sequence

of time slots during which individual channels (circuits) can be transmitted. Thus the
entire bandwidth is periodically available to each channel. Normally all time slots1 are
equal in length. Each channel is assigned a time slot with a specific common repetition
period called a frame interval. This is illustrated in Fig. 2.
Each channel is sampled at a specified rate and transmitted for a fixed
duration.All channels are sampled one by,thecycleisrepeatedagainand again. The
channels are connected to individual gates which are opened one by one in a fixed
sequence. At the receivingendalsosimilar gates are opened in unision with the gates at the
transmitting end.
The signal received at the receiving end will be in the form of discrete
samples and these are combined to reproduce the original signal. Thus, at a given instant
of time, onty one channel is transmitted through the medium, and by sequential sampling
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a number of channels can be staggered in time as opposed to transmitting all the channel
at the same time as in EDM systems. This staggering of channels in time sequence for
transmission over a common medium is called Time Division Multiplexing (TDM).

PULSE CODE MODULATION SYSTEM

It was only in 1938, Mr. A.M. Reaves (USA) developed a Pulse Code Modulation (PCM)
system to transmit the spoken word in digital form. Since then digital speech transmission
has become an alternative to the analogue systems.
PCM systems use TDM technique to provide a number of circuits on the same
transmission medium viz open wire or underground cable pair or a channel provided by
carrier, coaxial, microwave or satellite system.
Basic Requirements For PCM System
To develop a PCM signal from several analogue signals, the following processing steps
are required

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Filtering

Sampling
42
42

Quantisation

Encoding

Line Coding

FILTERING
Filters are used to limit the speech signal to the frequency band 300-3400 Hz.
SAMPLING
It is the most basic requirement for TDM. Suppose we have an analogue signal Fig. 3 (b),
which is applied across a resistor R through a switch S as shown in Fig. 3 (a) . Whenever
switch S is closed, an output appears across R. The rate at which S is closed is called the
sampling frequency because during the make periods of S, the samples of the analogue
modulating signal appear across R. Fig. 3(d) is a stream of samples of the input signal
which appear across R. The amplitude of the sample is depend upon the amplitude of the
input signal at the instant of sampling. The duration of these sampled pulses is equal to
the duration for which the switch S is closed. Minimum number of samples are to be sent
for any band limited signal to get a good approximation of the original analogue signal
and the same is defined by the sampling Theorem.

FIG. 3 : SAMPLING PROCESS

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Sampling Theorem
A complex signal such as human speech has a wide

range of frequency

components with the amplitude of the signal being different at different frequencies. To
put it in a different way, a complex signal will have certain amplitudes for all frequency
components of which the signal is made. Let us say that these frequency components
occupy a certain bandwidth B. If a signal does not have any value beyond this bandwidth
B, then it is said to be band limited. The extent of B is determined by the highest
frequency components of the signal.
Sampling Theorem States
"If a band limited signal is sampled at regular intervals of time and at a rate equal to or
more than twice the highest signal frequency in the band, then the sample contains all the
information of the original signal." Mathematically, if fH is the highest frequency in the
signal to be sampled then the sampling frequency Fs needs to be greater than 2 fH.
i.e. Fs>2fH
Let us say our voice signals are band limited to 4 KHz and let sampling
frequency be 8 KHz.
Time period of sampling Ts =

1 sec

8000
or Ts = 125 micro seconds
If we have just one channel, then this can be sampled every 125 microseconds and
the resultant samples will represent the original signal. But, if we are to sample N
channels one by one at the rate specified by the sampling theorem, then the time available
for sampling each channel would be equal to Ts/N microseconds.
Fig. .4 shows how a number of channels can be sampled and combined.
The channel gates (a, b ... n) correspond to the switch S in Fig. 3. These gates are opened
by a series of pulses called "Clock pulses". These are called gates because, when closed
these actually connect the channels to the transmission medium during the clock period
and isolate them during the OFF periods of the clock pulses. The clock pulses are
staggered so that only one pair of gates is open at any given instant and, therefore, only
one channel is connected to the transmission medium. The time intervals during which
the common transmission medium is allocated to a particular channel is called the Time
Slot for that channel. The width of.this time slot will depend, as stated above, upon the
number of channels to be combined and the clock pulse frequency i.e. the sampling
frequency.

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FIG. 4: SAMPLING & COMBINING CHANNELS

In a 30 channel PCM system. TS i.e. 125 microseconds are divided into
32 parts. That is 30 time slots are used for 30 speech signals, one time
slot

for

signalling

of

all

the

30

chls,

and

one

time

slot

for

synchronization between Transmitter & Receiver.

The time available per channel would be Ts/N = 125/32
= 3.9 microseconds
Thus in a 30 channel PCM system, time slot is 3.9 microseconds and time period
of sampling i.e..the interval between 2 consecutive samples of a channel is 125
microseconds. This duration i.e. 125 microseconds is called Time Frame.
The signals on the common medium (also called the common highway)
of a TDM system will consist of a series of pulses, the amplitudes of
which

are

proportional

to

the

amplitudes

of

the

their respective sampling instants. This is illustrated in Fig. 5

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45

individual

channels

at

i
FIG 5 : PAM OUTPUT SIGNALS
The original signal for each channel can be recovered at the receive end by
applying gate pulses at appropriate instants and passing the signals through low pass
filters. (Refer Fig. 6)

Fig. 6 : RECONSTRUCTION OF ORIGINAL SIGNAL

QUANTISATION
In FDM systems we convey the speech signals in their analogue electrical form. But
in PCM, we convey the speech in discrete form. The sampler selects a number of points
on the analogue speech signal (by sampling process) and measures their instant values.
The output of the sampler is a PAM signal as shown in Fig. 3; The transmission of PAM
signal will require linear amplifiers at trans and receive ends to recover distortion less

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signals. This type of transmission is susceptible to all the disadvantages of AM signal

transmission. Therefore, in PCM systems, PAM signals are converted into digital form by
using Quantization Principles. The discrete level of each sampled signal is quantified
with reference to a certain specified level on an amplitude scale.
The process of measuring the numerical values of the samples and giving them a
table value in a suitable scale is called "Quantising". Of course, the scales and the number
of points should be so chosen that the signal could be effectively reconstructed after
demodulation.
Quantising, in other words, can be defined as a process of breaking down a
continuous amplitude range into a finite number of amplitude values or steps.
A sampled signal exists only at discrete times but its amplitude is drawn from a
continuous range of amplitudes of an analogue signal. On this basis, an infinite number of
amplitude values is possible. A suitable finite number of discrete values can be used to
get an. approximation of the infinite set. The discrete value of a sample is measured by
comparing it with a scale having a finite number of intervals and identifying the interval
in which the sample falls. The finite number of amplitude intervals is called the
"quantizing interval". Thus, quantizing means to divide the analogue signal's total
amplitude range into a number of quantizing intervals and assigning a level to each
intervals.
For example, a 1 volt signal can be divided into 10mV ranges like 10-20mV, 3040mV and so on. The interval 10-20 mV, may be designated as level 1, 20-30 mV as
level 2 etc. For the purpose of transmission, these levels are given a binary code. This is
called encoding. In practical systems-quantizing and encoding are a combined process.
For the sake of understanding, these are treated separately.
Quantizing Process
Suppose we have a signal as shown in Fig. 7 which is sampled at instants a, b, c, d
and e. For the sake of explanation, let us suppose that the signal has maximum amplitude
of 7 volts.
In order to quantize these five samples taken of the signal, let us say the total amplitude is
divided into eight ranges or intervals as shown in Fig. 7. Sample (a) lies in the 5th range.
Accordingly, the quantizing process will assign a binary code corresponding to this i.e.
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101, Similarly codes are assigned for other samples also. Here the quantizing intervals
are of the same size. This is called Linear Quantizing.

FIG. 7 : QUANTIZING-POSITIVE SIGNAL

Assigning an interval of 5 for sample 1, 7 for 2 etc. is the quantizing
process.Giving,the assigned levels of samples,th ebinary codeis called coding of the
quantized samples.
Quantizing is done for both positive and negative swings. As shown in
Fig.6,eight quantizing levels are used for each direction of the analogue

signal.

To

indicate whether a sample is negative with reference to zero or is positive with reference
zero, an extradigitisadded to the binary code. This extra digit is called the "sign bit". In
Fig.8. positive values have a sign bit of '1' and negative values have sign
bit of'0'.

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FIG. 8 : QUANTIZING - SIGNAL WITH + Ve & - Ve VALUES

Relation between Binary Codes and Number of levels.

Because the quantized samples are coded in binary form, the quantization
intervals will be in powers of 2. If we have a 4 bit code, then we can have 2" = 16 levels.
Practical PCM systems use an eight bit code with the first bit as sign bit. It means we can
have 2" = 256 (128 levels in the positive direction and 128 levels in the negative
direction) intervals for quantizing.
Quantization Distortion
Practically in quantization we assign lower value of each interval to a sample falling in
any particular interval and this value is given as

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Table-1 : Illustration of Quantization Distortion

Analogue

Signal Quantizing Interval Quantizing Level

Binary Code

Amplitude Range

(mid value)

0-10 mv

5 mv

1000

10-20mv

15mv

1001

20-30 mv

25 mv

1010

30-40 mv

35 mv

1011

40-50 mv

45 mv

1100

If a sample has an amplitude of say 23 mv or 28 mv, in either case it will be

assigned \he \eve\ "2". This Is represented in binary code 1010. When this is decoded at
the receiving end, the decoder circuit on receiving a 1010 code will convert this into an
analogue signal of amplitude 25 mv only. Thus the process' of quantization leads to an
approximation of the input signal with the detected signal having some deviations in
amplitude from the actual values. This deviation between the amplitude of samples at the
transmitter and receiving ends (i.e. the difference between the actual value & the
reconstructed value) gives rise to quantization distortion.
If V represent the step size and 'e' represents the difference in amplitude
fe' must exists between - V/2 & + V/2) between the actual signal level and its quantized
equivalent then it can be proved that mean square quantizing error is equal to (V2). Thus,
we see that the error depends upon the size of the step.

12

In linear quantization, equal step means equal degree of error for all input
amplitudes. In other words, the signal to noise ratio for weaker signals will be poorer.
To reduce error, we, therefore, need to reduce step size or in other words,
increase th,e number of steps in the given amplitude range. This would
increase

the transmission

bandwidth

however,

because bandwidth B = fm log L. where L is

the number of quantum steps and fm is the highest signal frequency. But as we knows
from speech statistics that the probability of occurrence of a small amplitude is much
greater than large one, it seems appropriate to provide more quantum levels (V = low
value) in the small amplitude region and only a few (V = high value) in the region of
higher amplitudes. In this case, provided the total number of specified levels remains
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50

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unchanged, no increase in transmission bandwidth will be required. This will also try to
bring about uniformity in signal to noise ratio at all levels of input signal. This type of
quantization is called non-uniform quantization.
In practice, non-uniform quantization is achieved using segmented quantization
(also called companding). This is shown in Fig. 9 (a). In fact, there are equal number of
segments for both positive and negative excursions. In order to specify the location of a
sample value it is necessary to know the following :
1.

The sign of the sample (positive or negative excursion)

2.

The segment number

3.

The quantum level within the segment

As seen in Fig. 9 (b), the first two segment in each polarity are collinear, (i.e. the slope is
the same in the central region) they are considered as one segment. Thus the total number
of segment appear to be 13. However, for purpose of analysis all the 16 segments will be
taken into account.
ENCODING
Conversion of quantised analogue levels to binary signal is called encoding. To
represent 256 steps, 8 level code is required. The eight bit code is also called an eight bit
"word".

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The 8 bit word appears in the form

P

ABC

Polarity bit 1
for + ve 'O' for - ve.

WXYZ

Segment Code

Linear encoding

in the segment

The first bit gives the sign of the voltage to be coded. Next 3 bits gives the segment
number. There are 8 segments for the positive voltages and 8 for negative voltages. Last
4 bits give the position in the segment. Each segment contains 16 positions.
Referring to Fig. 9(b), voltage Vc will be encoded as 1 111 0101.

FIG. 9 (b) : ENCODING CURVE WITH COMPRESSION 8 BIT CODE

The quantization and encoding are done by a circuit called coder. The coder
converts PAM signals (i.e. after sampling) into a 8 bit binary signal. The coding is done
as per Fig. 9 which shows a relationship between voltage V to be coded and equivalent
binary number N. The function N = f(v) is not linear.
The curve has the following characteristics.

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It is symmetrical about the origins. Zero level corresponds to zero voltage to be encoded.
It is logarithmatic function approximated by 13 straight segments numbered 0 to 7 in
positive direction and 'O' to 7 in the negative direction. However 4 segments 0, 1, 0, 1
lying between levels + vm/64 -vm/64 being colinear are taken as one segment.
The voltage to be encoded corresponding to 2 ends of successive segments are in the
ratio of 2. That is vm, vm/2, vm/4, vm/8, vm/16, vm/32, vm/64, vm/128 (vm being the
maximum voltage).
There are 128 quantification levels in the positive part of the curve and 128 in the
negative part of the curve.
In a PCM system the channels are sampled one by one by applying the sampling
pulsqs to the sampling gates. Refer Fig. 10. The gates open only when a pulse is applied
to them and pass the analogue signals through them for the duration for which the gates
remain open. Since only one gate will be activated at a given instant, a common encoding
circuit is used for all channels. Here the samples are quantized and encoded. The encoded
samples of all the channels and signals etc are combined in the digital combiner and
transmitted.

53

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The reverse process is carried out at the receiving end to retreive the original
analogue signals. The digital combiner combines the encoded samples in the form of
"frames". The digital separator decombines the incoming digital streams into individual
frames. These frames are decoded to give the PAM (Pulse Amplitude Modulated)
samples. The samples corresponding to individual channels are separated by operating
the receive sample gates in the same sequence i.e. in synchronism with the transmit
sample gates.

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CONCEPT OF FRAME
In Fig. 10, the sampling pulse has a repetition rate of Ts sees and a pulse width of
"St". When a sampling pulse arrives, the sampling gate remains opened during the time
"St" and remains closed till the next pulse arrives. It means that a channel is activated for
the duration "St". This duration, which is the width of the sampling puse, is called the
"time slot" for a given channel.
Since Ts is much larger as compared to St. a number of channels can be sampled
each for a duration of St within the time Ts. With reference to Fig. 10, the first sample of
the first channel is taken by pulse 'a', encoded and is passed on the combiner. Then the
first sample of the second channel is taken by pulse 'b' which is also encoded and passed
on to the combiner, Likewise the remaining channels are also sampled sequentially and
are encoded before being fed to the combiner. After the first sample of the Nth channel is
taken and processed, the second sample of the first channel is taken, this process is
repeated for all channels. One full set of samples for all channel taken within the duration
Ts is called a "frame". Thus the set of all first samples of all channels is one frame; the
set of all second samples is another frame and so on.
As already said in para 5.3.5, Ts in a 30 channel PCM system is 125
microseconds and the signalling information of all the channels is transmitted through a
separate time slot. To maintain synchronization between
the

synchronization

data

transmit and receive

ends,

is transmitted through another time slot. Thus for a 30 chl

PCM system, we have 32 time slots.

Thus the time available per channel would be 3.9 microsecs.
Thus for a 30 chl PCM system,
Frame = 125 microseconds
Time slot per chl = 3.9 microseconds.
Structure of Frame
A frame of 125 microseconds duration has 32 time slots. These slots are
numbered Ts 0 to Ts 31.

55

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Information for providing synchronization between trans and receive ends is passed
through a separate time slot. Usually the slot Ts 0 caries the synchronizsation signals.
This slot is also called Frame alignment word (FAW).
The signalling informatiori is transmitted through time slot Ts 16.
Ts 1 to Ts 15 are utilized for voltage signal of channels 1 to 15 respectively.
Ts 17 to Ts 31 are utilized for voltage signal of channels 16 to 30 respectively.
SYNCHRONIZATION
The output of a PCM terminal will be a continuous stream of bits. At the
receiving end, the receiver has to receive the incoming stream of bits and discriminate
between frames and separate channels from these. That is, the receiver has to recognise
the start of each frame correctly. This operation is called frame alignment or
Synchronization and is achieved by inserting a fixed digital pattern called a "Frame
Alignment Word (FAW)" into the transmitted bit stream at regular intervals. The receiver
looks for FAW and once it is detected, it knows that in next time slot, information for
channel one will be there and so on.
The digits or bits of FAW occupy seven out of eight bits of Ts 0 in the following
pattern.
Bit position of Ts 0
FAW digit value

B1
X

B2
0

B3
1

B4
1

B5
0

B6
1

B7

B8

The bit position B1 can be either '1' or '0'. However, when the PCM system is to
be linked to an international network, the B1 position is fixed at '1'.
The FAW is transmitted in the Ts O of every alternate frame.
Frame which do not contain the FAW, are used for transmitting supervisory and
alarm signals.
To distinguish the Ts 0 of frame carrying supervisory/alarm signals from those carrying
the FAW, the B2 bit position of the former are fixed at T. The FAW and alarm signals are
transmitted alternatively as shown in Table - 2.

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TABLE-2
Frame

Remark

Numbers

B1

B2

B3

B4

B5

B6

B7

B8

FO

FAW

F1

ALARM

F2

FAW

F3 etc

ALARM

In frames 1, 3, 5, etc, the bits B3, B4, B5 denote various types of alarms. For example, in
B3 position, if Y = 1, it indicate Frame synchronisation alarm. If Y = 1 in B4, it indicates
high error density alarm. When there is no alarm condition, bits B3 B4 B5 are set 0. An
urgent alarm is indicated by transmitting "all ones". The code word for an urgent alarm
would be of the form.
X
111
SIGNALLING IN PCM SYSTEMS

1111

In a telephone network,-the signalling information is used for proper routing of

a call between two subscribers, for providing certain status information like dial tone,
busy tone, ring back. NU tone, metering pulses, trunk offering signal etc. All these
functions are grouped under the general terms "signalling" in PCM systems. The
signaling information can be transmitted in the form of DC pulses (as in step by step
exchange) or multifrequency pulses (as in cross bar systems) etc.
The signalling pulses retain their amplitude for a much longer period than
the pulses carrying speech information. It means that the signalling information
is a slow varying signal in time compared to the speech signal which is fast changing in
the time domain. Therefore, a signalling channel can be digitized with less number of bits
than a voice channel.
In a 30 chl PCM system, time slot Ts 16 in each frame is allocated for
carrying signalling information.
The

time

slot

16

of each

frame

carries

the

signalling

data

corresponding to two VF channels only. Therefore, to cater for 30 channels, we must

transmit 15 frames, each having 125 microseconds duration.

For

carrying

synchronization data for all frames, one additional frame is used. Thus a group of
57

Dept of. EEE

16 frames (each of 125 microseconds) is formed to make a "multiframe". The duration of

a multiframe is 2 milliseconds. The multiframe has 16 major time slots of 125
microseconds duration. Each of these (slots) frames has 32 time slots carrying, the
encoded samples of all channels plus the signaling and synchronization data. Each
sample has eight bits of duration 0.400 microseconds (3.9/8 = 0.488) each. The
relationship between the bit duration frame and multiframe is illustrated in Fig. 11 (a) &
11 (b).

Dept of. EEE

FIG. 11 (B) 2.048 Mb/s PCM MULTIFRAME

We have 32 time slots in a frame, each slot carries an 8 bit word.
The total number of bits per frame = 32 x 8 = 256
The total number of frames per seconds is 8000
The total number of bits per second are 256 x 8000 = 2048 K/bits.
Thus, a 30 chP PCM system has 2048 K bits.
Multiframe Structure
In the time slot 16 of FO, the first four bits (positions 1 to 4) contain the
multiframe alignment signal which enables the receiver to identify a multiframe.
The other four bits (no. 5 to 8) are spare. These may be used for carrying alarm signals.
Time slots 16 of frames F1 to FT5 are used for carrying the signalling information. Each
frame carries signalling, data for two VF channels. For instance, time slot Ts 16 of frame
F1 carries the signal data for VF channel 1 in the first four bits. The next four bits are
used for carrying signalling information
for channel 16. Similarly, time slot Ts16 of F2 carries signalling data of chls 2 .and 17.
Thus in multiframe structure, four signalling bits are provided for each VF channels.
As each multiframe includes 16 frames, each with a sacnqtoq -per sec.,.the.signalling of
each channel will occur at a rate of 500 per sec.

Dept of. EEE

CHAPTER - 4
DIGITAL SWITCHING
Introduction
A Digital switching system, in general, is one in which signals are switched in
digital form. These signals may represent speech or data. The digital signals of several
speech samples are time multiplexed on a common media before being switched through
the system.
To connect any two subscribers, it is necessary to interconnect the time-slots of the two
speech samples which may be on same or different PCM highways. The digitalised
speech samples are switched in two modes, viz., Time Switching and Space Switching.
This Time Division Multiplex Digital Switching System is popularly known as Digital
Switching System.
In this handout, general principles of time and space switching are discussed. A
practical digital switch, comprising of both time and space stages, is also explained.
Time and Space Switching
Generally, a digital switching system several time division
multiplexed (PCM) samples. These PCM samples are conveyed on PCM highways (the
common path over which many channels can pass with separation achieved by time
division.). Switching of calls in this environment , requires placing digital samples from
one time-slot of a PCM multiplex in the same or different time-slot of another PAM
multiplex.
For example, PCM samples appearing in TS6 of I/C PCM HWY1 are transferred to TS18
of O/G PCM HWY2, via the digital switch, as shown in Fig1.

FIG 1 DIGITAL SWITCH

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60

The interconnection of time-slots, i.e., switching of digital signals can be achieved using
two different modes of operation. These modes are: I. Space Switching
ii. Time switching
Usually, a combination of both the modes is used.
In the space-switching mode, corresponding time-slots of I/C and O/G PCM
highways are interconnected. A sample, in a given time-slot, TSi of an I/C HWY, say
HWY1, is switched to same time-slot, TSi of an O/G HWY, SAY HWY2. Obviously
there is no delay in switching of the sample from one highway to another highway since
the sample transfer takes place in the same time-slot of the PCM frame.
Time Switching, on the other hand, involves the interconnection of different timeslots on the incoming and outgoing highways by re-assigning the channel sequence. For
example, a time-slot TSx of an I/C Highway can be connected to a different time-slot.,
TSy, of the outgoing highway. In other words, a time switch is, basically, a time-slot
changer.
Digital Space Switching
Principle
The Digital Space Switch consists of several input highways, X1, X2,...Xn and
several output highways, Y1, Y2,.............Ym, inter connected by a crosspoint matrix of
n rows and m columns. The individual crosspoint consists of electronic AND gates. The
operation of an appropriate crosspoint connects any channel, a , of I/C PCM highway to
the same channel, a, of O/G PCM highway, during each appropriate time-slot which
occurs once per frame as shown in Fig 2. During other time-slots, the same crosspoint
may be used to connect other channels. This crosspoint matrix works as a normal space
divided matrix with full availability between incoming and outgoing highways during
each time-slot.
Each crosspoint column, associated with one O/G highway, is assigned a column
of control memory. The control memory has as many words as there are time-slot per
frame in the PCM signal. In practice, this number could range from 32 to 1024. Each
crosspoint in the column is assigned a binary address, so that only one crosspoint per
column is closed during each time-slot. The binary addresses are stored in the control
memory, in the order of time-slots. The word size of the control memory is x bits, so that
2x = n, where n is the number of cross points in each column .

A new word is read from the control memory during each time-slot, in a
cyclic order. Each word is read during its corresponding time-slot, i.e.,Word 0
(corresponding to TS0), followed by word 1 (corresponding to TS1) and so on.
The word contents are contained on the vertical address lines for the duration of
the time-slot. Thus, the cross point corresponding to the address, is operated

during a particular time-slot. This cross point operates every time the particular
time-slot appears at the inlet in successive frames. normally, a call may last for
around a million frames.
As the next time-slot follows, the control memory is also advanced by one
step, so that during each new time-slot new corresponding words are read from
the various control memory columns. This results in operation of a completely
different set of cross points being activated in different columns. Depending upon
the number of time-slots in one frame, this time division action increases the
utilisation of cross point 32 to 1024 times compared with that of conventional
space-divided switch matrix.
Illustration
Consider the transfer of a sample arriving in TS7 of I/C HWY X1 to O/G
HWY Y3. Since this is a space switch, there will be no reordering of time i.e., the
sample will be transferred without any time delay, via the appropriate cross point.
In other words, the objective is to connect TS7 of HWY X1 and TS7 of HWY Y3.
The

central

control

(CC)

selects

the

control

memory

column

corresponding output highway Y3. In this column, the memory location

corresponding to the TS7 is chosen. The address of the cross point is written in
this location, i.e., 1, in binary, is written in location 7, as shown in fig 2.This cross
point remains operated for the duration of the time-slot TS7, in each successive
frame till the call lasts.
For disconnection of call, the CC erases the contents of the control memory
locations, corresponding to the concerned time-slots. The AND gates, therefore,
are disabled and transfer of samples is halted.
Practical Space Switch
In a practical switch, the digital bits are transmitted in parallel rather than
serially, through the switching matrix.
In a serial 32 time-slots PCM multiplex, 2048 Kb/s are carried on a single wire
sequentially, i.e., all the bits of the various time-slots follow one another. This
single wire stream of bits, when fed to Serial to Parallel Converter is converted
into 8-wire parallel output. For example, all 8 bits corresponding to TS3 serial
input are available simultaneously on eight output wires (one bit on each output
wire), during just one bit period, as shown in fig.3. This parallel output on the
eight wires is fed to the switching matrix. It can be seen that during one full timeslot period, only one bit is carried on the each output line, whereas 8 bits are

carried on the input line during this period. Therefore, bit rate on individual
output wires, is reduced to 1/8th of input bit rate=2048/8=256Kb/s
Due to reduced bit rate in parallel mode, the cross point is required to be
operated only for 1/8th of the time required for serial working. It can, thus, be
shared by eight times more channels, i.e., 32 x 8 = 256 channels, in the same
frame.
However, since the eight bits of one TS are carried on eight wires, each
cross point have eight switches to interconnect eight input wires to eight output
wires. Each cross point (all the eight switches) will remain operated now for the
duration of one bit only, i.e., only for 488 ns (1/8th of the TS period of 3.9 s)

Fig 3 Serial parallel converter

For example, to connect 40 PCM I/C highways, a matrix of 40x 40 = 1600
cross points each having a single switch, is required in serial mode working.
Whereas in parallel mode working, a matrix of (40/8 x 40/8) = 25 cross point is
sufficient. As eight switches are required at each cross point 25 x 8 = 200
switches only are required. Thus, there is a reduction of the matrix by 1/8th in
parallel mode working, hence reduction in size and cost of the switching matrix.

Digital Time Switch

Principle
A Digital Time Switch consists of two memories, viz., a speech or buffer
memory to store the samples till destination time-slots arrive, and a control or
connection or address memory to control the writing and reading of the samples
in the buffer memory and directing them on to the appropriate time-slots.
Speech memory has as many storage locations as the number of time-slots
in input PCM, e.g., 32 locations for 32 channel PCM system.
The writing/reading operations in the speech memory are controlled by the
Control Memory. It has same number of memory locations as for speech memory,
i.e., 32 locations for 32 channel PCM system. Each location contains the address
of one of the speech memory locations where the channel sample is either written
or read during a time-slot. These addresses are written in the control memory of
the CC of the exchange, depending upon the connection objective.
A Time-Slot Counter which usually is a synchronous binary counter, is
used to count the time-slots from 0 to 31, as they occur. At the end of each frame,
It gets reset and the counting starts again. It is used to control the timing for
writing/reading of the samples in the speech memory.
Illustration
Consider the objective that TS4 of incoming PCM is to be connected to
TS6 of outgoing PCM. In other words, the sample arriving in TS4 on the I/C
PCM has to be delayed by 6 - 4 = 2 time-slots, till the destination time-slot, viz.,
TS6 appears in the O/G PCM. The required delay is given to the samples by
storing it in the speech memory. The I/C PCM samples are written cyclically i.e.
sequentially time-slot wise , in the speech memory locations. Thus, the sample in
TS4 will be written in location 4, as shown in fig.4.
The reading of the sample is controlled by the Control Memory. The
Control

Memory location corresponding to output time-slot TS6, is 6. In this

location, the CC writes the input time-slot number, viz.,4, in binary. These
contents give the read address for the speech memory, i.e., it indicates the speech
memory locations from which the sample is to be read out, during read cycle.
When the time-slot TS6 arrives, the control memory location 6 is read. Its
content addresses the location 4 of the speech memory in the read mode and
sample is read on to the O/G PCM.

In every frame, whenever time-slot 4 comes a new sample will be written in

location 4. This will be read when TS6 occurs. This process is repeated till the
call lasts.
For disconnection of the call, the CC erases the contents of the control
memory location to halt further transfer of samples.
Time switch can operate in two modes, viz.,
I.

Output associated control

ii.

Input associated control

Output associated control

In this mode of working , 2 samples of I/C PCM are written cyclically in
the speech memory locations in the order of time-slots of I/C PCM, i.e., TS1 is
written in location 1, TS2 is written in location 2, and so on, as discussed in the
example of Sec.4.2.
The contents of speech memory are read on output PCM in the order
specified by control memory. Each location of control memory is rigidly
associated with the corresponding time-slot of the O/G PCM and contains the
address of the TS of incoming PCM to be connected to. The control memory is
always read cyclically, in synchronism with the occurrence of the time-slot. The
entire process of writing and reading is repeated in every frame, till the call is
disconnected.

FIG 4 OUTPUT ASSOCIATED CONTROL SWITCH

It may be noticed that the writing in the speech memory is sequential and
independent of the control memory, while reading is controlled by the control
memory, i.e., there is a sequential writing but controlled reading.
Input associated control
Here, the samples of I/C PCM are written in a controlled way, i.e., in
the order specified by control memory, and read sequentially.
Each location of control memory is rigidly associated with the corresponding
TS of I/C PCM and contains the address of TS of O/G PCM to be connected to.
The previous example with the same connection objective of connecting TS4
of I/C PCM to TS6 of O/G PCM may be considered for its restoration. The
location 4 of the control memory is associated with incoming PCM TS4. Hence, it
should contain the address of the location where the contents of TS4 of I/C PCM
are to be written in speech memory. A CC writes the number of the destination
TS, viz., 6 in this case, in location 4 of the control memory. The contents of TS4
are therefore, written in location of speech memory, as shown in fig5.
The contents of speech memory are read in the O/G PCM in a sequential
way, i.e., location 1 is read during TS1, location 2 is read during TS2, and so on.
In this case, the contents of location 6 will appear in the output PCM at TS6. Thus
the input PCM TS4 is switched to output PCM TS6. In this switch, there is
sequential reading but controlled writing.

FIG 5 INPUT ASSOCIATED CONTROLLED TIME SWITCH

Time Delay Switching

The writing and reading, of all time-slots in a frame, has to be completed
within one frame time period (before the start of the next frame). A TS of
incoming PCM may, therefore, get delayed by a time period ranging from 1 TS to
31 TS periods, before being transmitted on
outgoing PCM. For example,
consider a case when TS6 of incoming PCM is to be switched to TS5 in outgoing
PCM. In this case switching can be completed in two consecutive frames only,
i.e., 121 microseconds for a 32 channel PCM system. However, this delay is
imperceptable to human beings.
Non-Blocking feature of a Time Switch
In a Time Switch, there are as many memory locations in the control and
speech memories as there are time-slots in the incoming and outgoing PCM
highways, i.e., corresponding to each time-slot in incoming highway, there is a
definite memory location available in the speech and control memories. Similarly,
corresponding to each time-slot in the outgoing highway there is a definite
memory location available in the control and speech memories. This way,
corresponding to free incoming and outgoing time-slots, there is always a free
path available to interconnect them. In other words, there is no blocking in a time
switch.
Two Dimensional Switching
Though the electronic cross points are not so expensive, the cost
of accessing and selecting them from external pins in a Space Switch, becomes
prohibitive as the switch size increases. Similarly, the memory location
requirements rapidly go up as a Time Switch is expanded, making it
uneconomical. Hence, it becomes necessary to employ a number of stages, using
small switches as building blocks to build a large network. This would result in
necessity of changing both the time-slot and highway in such a network. Hence,
the network, usually, employs both types of switches viz., space switch and time
switch, and. therefore, is known as two dimensional network. These networks can
have various combinations of the two types of switches and are denoted as TS,
STS, TSST,etc.
Though to ensure full availability, it may be desirable to use only
T stages. However, the networks having the architecture of TT, TTT, TTTT, etc.,
are uneconomical, considering the acceptability of tolerable limits of blocking, in
a practical network. Similarly, a two-stage two-dimensional network, TS or ST, is
basically suitable for very low capacity networks only. The most commonly used
architecture has three stages, viz., STS or TST. However, in certain cases, their
derivatives, viz., TSST, TSSST, etc., may also be used.

An STS network has relatively simpler control requirements and hence,

is still being favoured for low capacity networks, viz., PBX exchanges. As the
blocking depends mainly on the outer stages, which are space stages, it becomes
unsuitable for high capacity systems.
A TST network has lesser blocking constraints as the outer stages are time
stages which are essentially non-blocking and the space stage is relatively smaller.
It is, therefore, most cost-effective for networks handling high traffic, However,
for still higher traffic handling capacity networks, e.g., tandom exchanges, it may
be desirable to use TSST or TSSST architecture.
The choice of a particular architecture is dependent on other factors also,
viz., implementation complexity, modularity, testability, expandability, etc. As a
large number of factors favour TST structure, it is most widely used.
TST Network
As the name suggests, in a TST network, there are two time stages
separated by a space stage. The former carry out the function of time-slot
changing, whereas the latter performs highway jumping. Let us consider a
network having n input and n output PCM highways. Each of the input and output
time stages will have n time switches and the space stage will consist of an n x n
cross point matrix. The speech memory as well as the control memory of each
time switch and each column of a control memory of the space switch will have m
locations, corresponding to m time-slots in each PCM. Thus, it is possible to
connect any TS in I/C PCM to any TS in O/G PCM.
In the case of a local exchange, the network will be of folded type, i.e.,
the O/G PCM highways, via a suitable hybrid. Whereas, for a transit exchange,
the network will be non-folded, having complete isolation of I/C and O/G PCM
highways. However, a practical local exchange will have a combination of both
types of networks.
For the sake of explanation, let us assume that there are only four I/C and
O/G PCM highways in the network. Hence, there will be only four time switches
in each of the T-stages and the space switch will consist of 4x4 matrix. let us
consider an objective of connecting two subscribers through this switching
network of local exchange, assuming that the CC assigns TS4 on HWY0 to the
calling party and TS6 on HWY3 to the called party
The speech samples of the calling party have to be carried from TS4 of
I/C HWY 0 and to TS6 of O/G HWY3 and those of the called party from TS6 of
I/C HWY 3 to TS4 of O/G HWY 0 , with the help of the network. The CC
establishes the path, through the network in three steps. To introduce greater
flexibility, it uses an intermediate time-slot, TSx, which is also known as internal

time-slot. The three switching steps for transfer of speech sample of the calling
party to the called party are as under:
Step 1 Input Time Stage (IT) TS4 HWY0 to TSx HWY0
Step 2 Space stage (S)Tsx HWY0 to Tsx HWY3
Step 3 Output Time Stage (OT)Tsx HWY3 to TS6 HWY3
As the message can be conveyed only in one direction through this
path, another independent path, to carry the massage in the other direction is also
established by the CC, to complete the connection. Assuming the internal timeslots to be TS10 and TS11, the connection may be established as shown in fig 6.

FIG 6 T S T SWITCH
Let us now consider the detailed switching procedure making some more
assumptions for the sake of simplicity. Though practical time switches can handle
256 time-slots in parallel mode, let us assume serial working and that there are
only 32 time-slots in each PCM. Accordingly, the speech and control memories in
time switches and control memory columns in space switch, will contain 32
locations each.
To establish the connection, the CC searches for free internal timeslots. Let us assume that the first available time-slots are TS10 and TS11, as
before. To reduce the complexity of control, the first time stage is designed as
output-controlled switch, whereas the second time stage is input-controlled.

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70

FIG 7 T S T SWITCH STRUCTURE

For transfer of speech samples from the calling party to the called party of
previous example, CC orders writing of various addresses in location 10 of
control memories of IT-10, OT-3 and column 3 of CM-S of corresponding to O/G
highway, HWY3. Thus, 4 corresponding to I/C TS4 is written in CM-IT-0, 6
corresponding to O/G TS6 is written in CM-OT-3 and 0 corresponding to I/C
HWY 0 is written in column 3 of CM-S, as shown in fig. 7.
As the first time switch is output-controlled, the writing is done
sequentially. Hence, a sample, arriving in TS4 of I/C HWY 0,

is stored in

location 4 of SM-IT-0. It is readout on internal HWY 0 during TS10 as per the

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71

control address sent by CM-IT-0. In the space switch, during this internal TS10,
the cross point 0 in column 3 is enabled, as per the control address sent by column
3 of CM-S, thus, transferring the sample to HWY3. The second time stage is input
controlled and hence, the sample, arriving in TS10, is stored in location 6 of SMOT-3, as per the address sent by the CM-OT-3. This sample is finally, readout
during TS6 of the next frame, thus, achieving the connection objective.
Similarly, the speech samples in the other direction, i.e., from the called
party to the calling party, are transferred using internal TS11. As soon as the call
is over, the CC erases the contents in memory locations 10 and 11 of all the
concerned switches, to stop further transfer of message. These locations and timeslots are, then, avialable to handle next call.
Switching Network Configuration of some Modern Switches
E10B

- T-S-T

EWSD

- T-S-S-S-T

AXE10

- T-S-T

CDOT(MBM)

- T-S-T

5ESS

- T-S-T

OCB 283

-T

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CHAPTER 5
SIGNALLING IN TELECOMMUNICATION
1

Introduction
A telecommunication network establishes and realizes temporary connections, in

accordance with the instructions and information received from subscriber lines and inter
exchange trunks, in form of various signals.

Therefore, it is necessary to interchange

information between an exchange and it external environment i.e. between subscriber

lines and exchange, and between different exchanges. Though these signals may differ
widely in their implementation they are collectively known as telephone signals.
A signalling system uses a language which enables two switching equipments to converse
for the purpose of setting up calls. Like any other language. it possesses a vocabulary of
varying size and varying precision, ie. a list of signals which may also vary in size and a
syntax in the form of a complex set of rules governing the assembly of these signals.This
handout discusses the growth of signalling and various type of signalling codes used in
Indian Telecommunication.
Telephony started with the invention of magneto telephone which used a magneto to
generate the ringing current, the only signal, sent over a dedicated line between two
subscribers. The need for more signals was felt with the advent of manual switching. Two
additional signals were, therefore, introduced to indicate call request and call release.
The

range of signals increased further with the invention of electro-mechanical

automatic exchanges and is still growing further at a very fast pace, after the advent of
SPC electronic exchanges.
The interchange of signaling information can be illustrated with the help of a typical call
connection sequence. The circled number in Fig. 1 correspond to the steps listed below
i.

A request for originating a call is initiated when the calling subscriber

lifts the handset.

ii.

The exchange sends dial-tone to the calling subscriber to indicate to him to start
dialing.

iii.

The called number is transmitted to the exchange, when the calling subscriber
dials the number.

iv.

If the number is free, the exchange sends ringing current to him.

v.

Feed-back is provided to the calling subscriber by the exchange by sending,

a) Ring-back tone, if the called subscriber is free(shown in fig.1)
b) Busy tone if the called subscriber is busy ( not shown in the figure), or
c)Recorded message, if provision exists, for non completion of call due to
some other constraint ( not shown in figure).
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73

vi.

The called subscriber indicates acceptance of the incoming call by lifting the

handset
vii.

The exchange recognizing the acceptance terminates the ringing current and the
ring-back tone, and establishes a connection between the calling and called
subscribers.

viii.

The connection is released when either subscriber replaces the handset.When the
called subscriber is in a different exchange, the

following

inter-exchange

trunk. signal functions are also involved, before the call can be set up.
ix

The originating exchange seizes an idle inter exchange trunk, connected to a digit
register at the terminating exchange.

x.

The originating exchange sends the digit. The steps iv to viii are then performed
to set up the call.

Types of Signalling
Subscriber Line signalling
Calling Subscriber Line Signaling

In automatic exchanges the power is fed over the subscribers loop by the centralized
battery at the exchange. Normally, it is 48 V. The power is fed irrespective of the
state of the subscriber, viz., idle, busy or talking.
Call request
When the subscriber is idle, the line impedance is high. The line impedance falls, as soon
as, the subscriber lifts the hand-set, resulting in increase of line current. This is
detected as a new call signal and the exchange after connecting an appropriate
equipment to receive the address information sends back dial-tone signal to the
subscriber.
Address signal
After the receipt of the dial tone signal, the subscriber proceeds to send the address digits.
The digits may be transmitted either by decade dialing or by multifrequency
pushbutton dialling.
1. Decadic Dialling
The address digits may be transmitted as a sequence of interruption of the DC loop by a
rotary dial or a decadic push-button key pad. The number of interruption (breaks)
indicate the digit, exept0, for which there are 10 interruptions. The rate of such
interruptions is 10 per second and the make/break ration is 1:2. There has to be a
inter-digital pause of a few hundred milliseconds to enable the exchange to

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distinguish between consecutive digits. This method is, therefore, relatively slow
and signals cannot be transmitted during the speech phase.
2. Multifrequency Push-button Dialling
This method overcomes the constraints of the decadic dialling. It uses two sets of four
voice frequencies. pressing a button (key), generates a signal comprising of two
frequencies. one from each group. Hence, it is also called Dual-Tone Multifrequency (DTMF) dialling. The signal is transmitted as long as the key is kept
pressed. This provides 16 different combinations. As there are only 10 digits, at
present the highest frequency, viz., 1633 Hz, is not used and only 7 frequencies
are used, as shown in Fig.2.
By this method, the dialling time is reduced and almost 10 digits can be
transmitted per second. As frequencies used lie in the speech band, information
may be transmitted during the speech phase also, and hence, DTMF telephones
can be used as access teminals to a variety of systems, such as computers with
voice output. The tones have been so selected as to minimize harmonic
interference and probability of simulation by human voice.
HIGH FREQUENCY GROUP
1209 Hz
1336 Hz
097 Hz

ABC
2

DEF
3

GHI
4

JKL
5

MNO
6

PRS
7

TUV
8

WXY
9

170 Hz

662 Hz
OPER
0

FIGURE 2. TONE-DIALLING FREQUENCY GROUPS.

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75

End of selection signal

The address receiver is disconnected after the receipt of complete address. After the
connection is established or if the attempt has failed the exchange sends any one of the
following signals.
1.

Ring-back tone to the calling subscriber and ringing current to

the

called

subscriber, if the called line is free.

2.

Busy-tone to the calling subscriber, if the called line is busy or

otherwise

inaccessible.
3.

Recorded announcement to the calling subscriber, if the

indicate reasons for call failure, other than

provision

exists,

to

called line busy.

Ring back, tone and ringing current are always transmitted from the called subscriber
local exchange and busy tone and recorded announcements, if any, by the equipment as
close to the calling subscriber as possible to avoid unnecessary busying of equipment and
trunks.
Answer Back Signal
As soon as the called subscriber lifts the handset, after ringing, a battery reversal signal
is transmitted on the line of the calling subscriber. This may be used to operate special
equipment attached to the calling subscriber, e.g., short-circuiting the transmitter of a
CCB, till a proper coin is inserted in the coin-slot.
Release signal
When the calling subscriber releases i.e., goes on hook, the line impedance goes high.
The exchange recognizing this signal, releases all equipment involved in the call. This
signal is normally of more than 500 milliseconds duration.
Permanent Line (PG) Signal
Permanent line or permanent glow (PG) signal is sent to the calling subscriber if he fails
to release the call even after the called subscriber has gone on-hook and the call is
released after a time delay. The PG signal may also be sent, in case the subscriber takes
too long to dial. It is normally busy tone.
Called subscriber line signals.
Ring Signal
On receipt of a call to the subscriber whose line is free, the terminating exchange sends
the ringing current to the called telephone. This is typically 25 or 50Hz with suitable
interruptions. Ring-back tone is also fed back to the calling subscriber by the terminating
exchange.

Answer Signal
When the called subscriber, lifts the hand-set on receipt of ring, the line impedance goes
low. This is detected by the exchange which cuts off the ringing current and ring-back
tone.
Release Signal
If after the speech phase, the called subscriber goes on hook before the calling subscriber,
the state of line impedance going high from a low value, is detected. The exchange sends
a permanent line signal to the calling subscriber and releases the call after a time delay, if
the calling subscriber fails to clear in the meantime.
Register Recall Signal
With the use of DTMF telephones, it is possible to enhance the services, e.g., by dialing
another number while holding on to the call

in progress, to set up a call to a third

subscriber. The signal to recall the dialling phase during the talking phase, is called
Register Recall Signal. It consists of interruption of the calling subscribers loop for
duration less than the release signal. it may be of 200 to 320 milliseconds duration.
Inter-exchange Signaling
Inter-exchange signaling can be transmitted over each individual inter exchange
trunk. The signals may be transmitted using the same frequency band as for speech
signals (inband signaling), or using the frequencies outside this band (out-of-band
signaling). The signaling may be
i. Pulsed
The signal is transmitted in pulses. Change from idle condition to one of active states for
a particular duration characterizes the signal, e.g., address information
ii. Continuous
The signal consists of transition from one condition to another, a steady state condition
does not characterizes any

signal.

iii. Compelled
It is similar to the pulsed mode but the transmission is not of fixed duration but condones
till acknowledgement of the receiving unit is received back at the sending unit. It is a
highly reliable mode of signal transmission of complex signals.
Line signals
DC Signaling
The simplest cheapest, and most reliable system of signaling on
trunks, was DC signaling, also known as metallic loop signaling, exactly the same
as used between the subscriber and exchange, i.e.,

i.

Circuit seizure/release corresponding to off/on-hook signal of the

subscriber.
ii.

Address information in the from of decade pulses.

In-Band and Out-of-Band Signals

Exchanges separated by long distance cannot use any form of DC line signaling.
Suitable interfaces have to be interposed between them, for conversion of the
signals into certain frequencies, to enable them to be carried over long distance. A
signal frequency (SF) may be used to carry the on/off hook information. The
dialing pulses can also be transmitted by pulsing of the states. The number of
signals is small and they can be transmitted in-band or out-of band. The states
involved are shown in Table 1.
TABLE 1. SINGLE FREQUENCY SIGNALING STATES TONE SIGNAL
CONDITION
State
Idle (On hook)

Forward
On

Backward
On

Seizure(off hook)

off

on

Release (on hook)

on

off/on

Answer(off hook)

off

off

Clear Back (on hook)

off

on

Blocking (off hook)

on

off

FORWARD

BACKWARD

For in band signaling the tone frequency is chosen to be 2600Hz. or 2400 Hz. As
the frequency lies within the speech band, simulation of tone-on condition
indicating end-of call signal by the speech, has to be guarded against, for premature disconnection.
Out-of- Band signaling overcomes the problem of tone on condition imitation by
the speech by selecting a tone frequency of 3825 Hz which is beyond the speech
band. However, this adds up to the hard-ware costs.
5.3.1.3 E & M Signals
E & M lead signaling may be used for signaling on per-trunk basis. An additional
pair of circuit, reserved for signaling is employed. One wire is dedicated to the
forward signals ((M-Wire for transmit or mouth) which corresponds to receive or
R-lead of the destination exchange, and the other wire dedicated to the backward

signals (E-wire for receive or ear) which corresponds transmit or send wire or SLead of the destination exchange. The signaling states are shown in table2.
TABLE 2. E & M SIGNALING STATES
State

Outgoing Exchange

Incoming Exchange

M- lead

M- lead

Idle(On hook)

Earth

FORWARD

Battery

E-lead
Open
Open

Elead

Earth

Open

Earth

Earth

Battery/Earth

Open

seizure(off hook)
Release(On hook)

Earth

Earth/open

BACKWARD
Answer(off hook)

battery

Earth

Battery

Earth

Clear Back

battery

Open

Earth

earth

(On hook)
Blocking

Earth

Earth

Battery

Open

This type of signaling is normally used in conjunction with an interface to change

the E & M signals into frequency signal to be carried along with the speech.
Register Signals
It was, however felt that the trunk service could not be managed
properly
without the trunk register which basically is an address digit receiver, with such
development, the inter-exchange signaling was sub- divided into two categories.
1. Line signaling in which the signals operate throughout the duration of call, and
2. Register signaling during the relatively short phase of setting up the call,
essentially for transmitting the address information.

forward
signal

outgoing register
incomming
register

time

2-and-2only
signal recognition
acknowledgement
backward
signal and request for
next signal

time

signal cessation
signal cessation
recognition
recognition

compelled signal
sequence

next forward
signal

acknowledgement
backward signal
Sending

Fig.3. Compelled signalling procedure

In other words, register signals are interchanged between registers during a phase
between receipt of trunk seizure signal and the exchange switching to the speech
phase. These signals are proceed-to-send (PTS) signals, address, signals, and
signals indicating the result of the call attempt.
The register signals may be transmitted in band or out of band. however, in the
latter case, the signaling is relatively slow and only limited range of signals may
be used. For example, a

single out-of-band frequency may be selected and

information sent as pulses.

In-band transmission can be used easily as there can be no possible interference
with the speech signals. To reduce transmission time and to increase reliability, a
number of frequencies are used in groups. Normally 2 out of 6 frequencies are
used. To make the system more reliable compelled sequence is used. Hence, this
system is normally called compelled sequence Multi-frequency (CSMF) signaling
as shown in Fig.3. In CCITT terminology it is termed as R2 system. As the
frequencies need be transmitted only for a short duration to convey the entire
information, the post dialling delay is reduced.
When more than two exchanges are involved in setting up the connections the
signaling
i.

may be done in either of the two modes

End-to-end signaling

80
80

The signaling is always between the ends of the connection, as the call progresses.
Considering a three exchanges,

A-B-C,

connection, initially the signaling is

between A-B, then between A-C after the B-C connection is established.
ii.

Link-By-Link signaling

The signaling is always confined to individual links. Hence, initially the signaling
is between A-B, then between B-C after the B-C connection is established.
Generally supervisory (or line) and subscriber signaling is necessarily on link-bylink basis. Address component may be signalled either by end-to-end or link-bylink depending upon the network configuration.
R2 Signalling
CCITT standardized the R2 signaling system to be used on national and
international routes. However, the Indian environment requires lesser number of
signals and hence, a slightly modified version is being used.
There is a provision for having 15 combinations using two out of six frequencies
viz., 1380, 1500, 1620, 1740, 1860 and 1980 Hz, for forward signals and another
15 combination using two out of six frequencies viz., 1140,1020, 900, 780, 660
and 540 Hz, for backward signals. In India, the higher frequency in the forward
group i.e., 1980 Hz, and the lower frequency in the backward group, i.e., 540 hz,
are not used. Thus, there are 10 possible combinations in both the directions. The
weight codes for the combinations used are indicated in Table 3 and the
significance of each signal is indicated in Table 4 and 5.

TABLE 3- SIGNAL FREQUENCY INDEX AND WEIGHT CODE

Signal Frequency (Hz)

Forward

1380

1500

1620

1740

1860

Backward

1140

1020

900

780

660

f0

f1

f2

f3

f4

Index
Weight Code

1
2

TABLE 4-FORWARD SIGNALS

Weight
Group I
0+1
Digit 1
0+2
Digit2

3
4
5
6
7

1+2
0+4
1+4
2+4
0+7

Digit3
Digit4
Digit5
Digit6
Digit7

8
9
10

1+7
2+7
4+7

Digit8
Digit9
Digit0

Signal

Signal
No.
1

Weight
Code
0+1

2
3

0+2
1+2

0+4

5
6

1+4
2+4

TABLE 5 -BACKWARD SIGNALS

Group A
Send next digit
Restart
Address complete, Changeover to
reception of group B signals
Calling line identification for malicious
calls
send calling subscribers category
Set up speech connection

7
0+7
Send last but 1 digit
8
1+7
Send last but 2 digit
9
2+7
Send last but 3 digit
10
4+7
Spare
Note : Signals A2, and A7 to A9 are used in Tandem working only.
It can be seen from the tables that

Group II
Ordinary subscriber
Subscriber
with
priority Test / Mtce,
equipment
Spare
STD Barred
Spare
CCB
Changed Number to
Operator
Closed Number
Closed Number
Spare

Group B
Called line free with out
metering
Changed number
Called line busy
Local congestion
Number unobtainable
called line fee, with
metering
Route congestion
Spare
Route Breakdown
Malicious call blocking

1. Forward signals are used for sending the address information of the called
subscriber, and category and address, information of the calling subscriber.
2. Backward signals are used for demanding address information and

callers

category and for sending condition and category of called line.

R2 signaling is fully compelled and the backward signal is transmitted as an
acknowledgement to the forward signal. This speeds up the interchange of
information, reducing the call set up time. However, the satellite circuits are an

exception and semi-compelled scheme may only be used due to long propagation
time.
Register signals may be transmitted on end-to-end basis. It is a self checking
system. Each signal is acknowledgement appropriately at the other end after the
receiver checks the presence of only 2 and only 2 out of 5 proper frequencies.
An example of CSMF signaling between two exchanges may be illustrated
by considering a typical case. The various signals interchanged after seizure of the
circuit using DC signaling are
1.

originating exchange sends first digit

2.

Receipt of the digit is acknowledged by the terminating exchanges by sending A5

(demanding the callers category).

3.

A5 is acknowledgement by sending any11-1 to 11-5 by the originating exchange

4.

Terminating exchange acknowledges this by A1, demanding for next digit.

5.

Originating exchange, acknowledges A1 by sending any of 1-1to 1-10 sending the

digit.
6.

The digits are sent in succession by interchange of steps v and vi.

7.

On receipt of last digit, the terminating exchange carries out group and line

selection and then sends A3, indicating switching over to group B signals.
8.

This is acknowledgement by the originating exchange by sending the callers

category again.
9.

The terminating exchange acknowledgements by sending the called line condition

by sending any of B2 to B6.

10.

In response to B6, the originating exchanges switches through the speech path and

the registers are released. Alternatively, in

response to B2 to B5, the registers are

released and appropriate tone is fed to the calling subscriber by the originating
exchange.
3

Digital Signalling
All, the systems discussed so far, basically, are on per line or per trunk basis, as
the signals are carried on the same line or trunk. With the emergence of PCM
systems, it was possible to segregate the signaling from the speech channel.
Inter exchange signalling can be transmitted over a channel directly associated
with the speech channel, channel-associated signalling (CAS) ,

or

over a

dedicated link common to a number of channels, common channel signalling

(CCS). The information transmitted for setting up and release of calls is same in
both the cases. Channel associated signalling requires the exchanges, to have
access to each trunk via the equipment which may be decentralised, whereas, in

common channel signalling, the exchange is connected to only a limited number

of signalling links through a special terminal.
Channel- Associated signalling
In the PCM systems the signalling information is conveyed on a separate channel which
is rigidly associated with the speech channel. Hence, this method is known as channel
associated signalling (CAS). Though the speech sampling rate is 8 Khz, the signals do not
change as rapidly as speech and hence, a lower sampling rate of 500 Hz, for digitisation
of signals can suffice. Based on this concept, TS 16 of each frame of 125 microseconds is
used to carry signals of 2 speech channels, each using 4 bits.
Hence, for a 30 channel PCM system, 15 frames are required to carry all the signals. To
constitute a 2 millisecond multiframe of 16 frames. F 0 to F 15 TS 16 of the frame F 0 is
used for multiframe synchronisation. TS 16 of F1 contains signal for speech channels 1
and 16 being carried in TS 1 and TS 17, respectively, TS16 of F2 contains signals of
speech channels 2 and 17 being carried in TS2 and TS 18, respectively and so on, Both
line signals and address information can be conveyed by this method.
Although four bits per channel are available for signalling only two bits are used. As the
transmission is separate in the forward and backward direction, the bits in the forward
link are called af and bf, and those in the backward link are called ab and bb. Values for
these bits are assigned as shown in Table 6.
As the dialling pulses are also conveyed by these conditions, the line state recognition
time is therefore, above a threshold value. The bit bf is normally kept at 0, and the value
1 indicates a fault.
However, the utilisation of such a dedicated channel for signalling for each speech
channel is highly inefficient as it remains idle during the speech phase. Hence, another
form of signalling known as common-channel signalling evolved.
State

Bit Value
Backward.

Forward

af

Idle
Seizure
Seizure acknowledge
Answer
Clear Forward
Clear Back

1
0
0
0
1
0

bf

0
0
0
0
0
0

ab

1
1
1
0
0/1
1

bb

0
0
1
1
1
1

COMMON CHANNEL SIGNALING SYSTEM No. 7 (CCS#7)

1.

Introduction
Communication

networks

generally

connect

two

subscriber

terminating

equipment units together via several line sections and switches for message
exchange (e.g. speech, data, text or images). Control information has to be
transferred between the exchanges for call control and for the use of facilities. In
analog communication networks, channel-associated signaling systems have so far
been used to carry the control information. Fault free operation is guaranteed with
the channel-associated signaling systems in analog communication networks, but
the systems

do not

meet

requirements

in

digital,

processor-controlled

communication network. Such networks offer a considerably larger scope of

performance as compared with the analog communication networks due, for
instance, to a number of new services and facilities. The amount and variety of the
information to be transferred is accordingly larger. The information can no longer
be economically transported by the conventional channel-associated signaling
systems. For this reason, a new, efficient signaling system is required in digital,
processor-controlled communication networks.
The CCITT has, therefore, specified the common channel signalling system no.7
(CCS-7). CCS-7 is optimised for application in digital networks. It is
characterised by the following main features :
internationally standardized (national variations possible).
suitable for the national, international and intercontinental network level.
suitable for various communication services such as telephony, text services, data
services digital network (ISDN).
high performance and flexibility along with a future-oriented concept which well
meet new requirements.
high reliability for message transfer.
processor-friendly structure of messages (signal units of multiples of 8
bits).
signalling on separate signalling links; the bit rate of the circuits is,
therefore, exclusively for communication.
signalling links always available, even during existing calls.
use of the signalling links for transferring user data also.
used on various transmission media
- cable (copper, optical fiber)

- radio relay
- satellite (up to 2 satellite links)
use of the transfer rate of 64 Kbit/s typical in digital networks.
used also for lower bit rates and for analog signalling links if necessary.
automatic supervision and control of the signalling network.
2.

CC#7 Signalling terminology

2.1

Signalling Network

In contrast to channel-associated signalling, which has been standard practice until

now, in CCS7 the signalling messages are sent via separate signalling links (See
Fig. 1). One signalling link can convey the signalling messages for many circuits
The CCS7 signalling links connect signalling points (SPs) in a communication
network. The signalling points and the signalling links form an independent
signalling network which is overlaid over the circuit network.

Fig 1. Signalling via a Common Channel Signalling link

2.2.

Signalling Points (SP)

A distinction is made between signalling points (SP) and signalling transfer points
(STP).
The SPs are the sources (originating points) and the sinks (destination points) of
signalling traffic. In a communication network these are primarily the exchanges.
The STPs switch signalling messages received to another STP or to a SP on the
basis of the destination address. No call processing of the signalling messages

occurs in a STP. A STP can be integrated in a SP (e.g. in an exchange) or can form

a node of its own in the signalling network. One or more levels of STPs are
possible in a signalling network, according to the size of the network.
All SPs in the signalling network are identified by means of a code within the
framework of a corresponding numbering plan and, therefore, can be directly
addressed in a signalling message.
2.3. Signalling links
A signalling link consists of a signalling data link (two data channels operating
together in opposite directions at the same date rate) and its transfer control
functions. A channel of an existing transmission link (e.g. a PCM30 link) is used
as the signalling data link. Generally, more than one signalling link exists between
two SPs in order to provide redundancy. In the case of failure of a signalling link,
functions of the CCS7 ensure that the signalling traffic is rerouted to fault-free
alternative routes. The routing of the signalling links between two SPs can differ.
All the signalling links between two SPs are combined in a signalling link set.
2.4.

Signalling Modes

Two different signalling modes can be used in the signalling networks for CCS7,
viz. associated mode and quasi-associated mode.
In the associated mode of signalling, the signalling link is routed together with
the circuit group belonging to the link. In other words, the signalling link is
directly connected to SPs which are also the terminal points of the circuit group
(See Fig.2). This mode of signalling is recommended when the capacity of the
traffic relation between the SPs A and B is heavily utilized.

Fig. 2 Associated Mode of Signalling

In the quasi-associated mode of signalling, the signalling link and the speech
circuit group run along different routes, the circuit group connecting the SP A
directly with the SP B. For this mode, the signalling for the circuit group is carried
out via one or more defined STPs (See Fig. 3.3). This signalling mode is
favourable for traffic relations with low capacity utilization, as the same signalling
link can be used for several destinations.

Fig. 3 Quasi-associated mode

2.5 Signalling Routes

The route defined for the signalling between an originating point and a destination
point is called the signalling route. The signalling traffic between two SPs can be
distributed over several different signalling routes. All signalling routes between
two SPs are combined in a signalling route set.
2.6 Network Structure
The signalling network can be designed in different ways because of the two
signalling modes. It can constructed either with uniform mode of signalling
(associated or quasi-associated) or with a mixed mode (associated and quasiassociated).
The worldwide signalling network is divided into two levels that are functionally
independent of each other; an international level with an international network and
a national level with many national networks. Each network has its own
numbering plans for the SPs.
3. Planning Aspects
Economic, operational and organizational aspects must be considered in the
planning of the signalling network for CCS7. An administration should also have
discussions with the other administrations at an early stage before CCS7 is
introduced in order to make decisions, for example, on the following points :
(a) Signalling network
- mode of signalling
- selection of the STPs
- signalling type (en block or overlap)
- assignment of the addresses to SPs.

(b) Signalling data links, e.g. 64 kbit/s digital or 4.8 kbit/s analog
(c) Safety requirements
- load sharing between signalling links
- diverting the signalling traffic to alternative routes in event of faults.
- error correction
(d) Adjacent traffic relations
The signalling functions in CCS7 are distributed among the following parts :
- message transfer part (MTP)
- function specific user parts (UP)
The MTP represents a user-neutral means of transport for messages between the
users. The term user is applied here for all functional units which use the transport
capability of the MTP.
Each user part encompasses the functions, protocols and coding for the signalling
via CCS7 for a specific user type (e.g. telephone service, data service, ISDN). In
this way, the user parts control the set-up and release of circuit connections, the
processing of facilities as well as administration and maintenance functions for the
circuits.
The functions of the MTP and the UP of CCS7 are divided into 4 levels. Levels to
3 are allotted to the MTP while the UPs form level 4 .

Fig. 4 Functional Levels of CCS7

The message transfer part (MTP) is used in CCS7 by all user parts (UPs) as a
transport system for message exchange. Messages to be transferred from one UP
to another are given to the MTP (See Fig.5). The MTP ensures that the messages
reach the addressed UP in the correct order without information loss, duplication
or sequence alteration and without any bit errors.
4.

Functional Levels

Fig. 5 Message exchange between two Signalling Points with CCS7

4.1 Level I (Signalling Data Link) defines the physical, electrical and functional
characteristics of a signalling data link and the access units. Level 1 represents the
bearer for a signalling link. In a digital network, 64-kbit/s channels are generally
used as signalling data links. In addition, analog channels (preferablywith a bit rate
of 4.8 kbit/s) can also be used via modems as a signalling data
link.
4.2

Level 2 (Signalling Link) defines the functions and procedures for a correct
exchange of user messages via a signalling link. The following functions must be
carried out at level 2 :
- delimitation of the signal units by flags.
- elimination of superfluous flags.
- error detection using check bits.
- error correction by re-transmitting signal units.
- error rate monitoring on the signalling data link.
90
90

- restoration of fault-free operation, for example, after disruption of the signalling

data link.
4.3

Level 3 (Signalling Network) defines the inter-working of the individual

signalling links. A distinction is made between the two following functional areas
:
- message handling, i.e. directing the messages to the desired signalling line, or to
the correct UP.
- signalling network management, i.e. control of the message traffic, for example,
by means of changeover of signalling links if a fault is detected and changeback to
normal operation after the fault is corrected.
The various functions of level 3 operate with one another, with functions of other
levels and with corresponding functions of other signalling of other SPs.
5.

CCS#5 Signalling messages Common terms

5.1 Signal Units (SU)
The MTP transport messages in the form of SUs of varying length. A SU is
formed by the functions of level 2. In addition to the message it also contains
control information for the message exchange. There are three different types of
SUs :
- Message Signal Units (MSU).
- Link Status Signal Units (LSSU).
- Fill-in Signal Units (FISU).
Using MSUs the MTP transfers user messages, that is, messages from UPs (level
4) and messages from the signalling network management (level 3). The structure
of the three types of message units is shown in Fig.6.
The LSSUs contain information for the operation of the signalling link (e.g. of the
alignment).
The FISUs are used to maintain the acknowledgement cycle when no user
messages are to be sent in one of the two directions of the signalling link.
5.2 Protocol Information Bits
Flag (F) : (8 bits) The SUs are of varying length. In order to clearly separate them
from one another, each SU begins and ends with a flag. The closing flat of one
SUs is usually also the opening flag of the next SU. However, in the event of
overloading of the signalling link, several consecutive flags can be sent. The flag
is also used for the purpose of alignment. The bit pattern of a flg is 01111110.
5.3

Backward Sequence Number (BSN) : (7 bits) The BSN is used as an

acknowledgement carrier within the context of error control. It contains the

91
91

forward sequence number (FSN) of a SU in the opposite direction whose

reception is being acknowledged. A series of SUs can also be acknowledged with
one BSN.
5.4

Backward Indicator Bit (BIB) : (1 bit) The BIB is needed during general

error correction. With this bit, faulty SUs are requested to be retransmitted for
error correction.

Fig. 6 Format of Various Signal Units

5.5

Forward Sequence Number (FSN) : (7 bits) A FSN is assigned

consecutively to each SU to be transmitted. On the receive side, it is used for

supervision of the correct order for the SUs and for safeguarding against
transmission errors. The numbers 0 to 127 are available for the FSN.
5.6

Forward Indicator Bit (FIB) : (1 bit) The FIB is needed during general

error correction. It indicates whether a SU is being sent for the first time or
whether it is being retransmitted.
5.7

Length Indicator (LI) : (6 bits) The LI is used to differentiate between

the three SUs. It gives the number of octets between the check-bit (CK) field and
the LI field. The LI field contains different values according to the type of SU; it
is 0 for FISU, 1 or 2 for LISU and is greater than 2 for MSU.
The maximum value in the length indicator fields is 63 even if the signalling
information field (SIF) contains more than 63 octets.
92
92

5.8

Check bits (CK) : (16 bits) The CKs are formed on the transmission side

from the contents of the SU and are added to the SUs as redundancy. On the receive side,
the MTP can determine with the CKs whether the SU was transferred without any errors.
The SUs acknowledged as either positive or faulty on the basis of the check.
5.9 Fields specific to MSUs :
5.9.1 Service Information Octet (SIO) : (8 bits) It contains the Service
Indicator (SI, 4 bits) and Subservice field (SSF, 4 bits) whose last 2 bits are Network
Indicator (NI).
An SI is assigned to each user of the MTP. It informs the MTP which UP has sent
the message and which UP is to receive it. Four SI bits can define 16 UPs (3-SCCP, 4TUP, 5-ISUP, 6-DATAUP, 8-MTP test, etc.). The NI indicates whether the traffic is
international (00,01) or national (10,11). In CCS7 a SP can belong to both national and
international network at the same time. So SSF field indicate where the SP belongs.
5.9.2

Signalling Information Fields (SIF) : (2 to 272 octets) It contains the

actual user message. The user message also includes the address (routing label, 40 bits)
of the destination to which the message is to be transferred. The maximum length of the
user message is 62 octets for national and 272 octets for international networks (one octet
= 8 bits). The format and coding of the user message are separately defined for each UP.
5.10 Fields Specific to LSSUs
5.10.1 Status Field (SF) : (1 to 2 octets) It contains status indications for the
alignment of the transmit and receive directions. It has 1 or 2 octets, out of which only 3
bits of first octet are defined by CCITT, indicating out (000), normal (001), Emergency
(010) alignments, out-of-service (011), Local processor outage (100) status, etc.
5.10.2 Addressing of the SUs (in SIF)
A code is assigned to each SP in the signalling network according to a numbering
plan. The MTP uses the code for message routing. The destination of a SU is specified in
a routing label. The routing label is a component of every user message and is transported
in the SIF. The routing label in a MSU consists of the following (See Fig. 7).

Fig. 7 Routing Label of a Message Signal Unit

93
93

5.10.3 Destination Point Code (DPC) : (14 bits) identifies the SP to which this
message is to be transferred.
5.10.4 Originating Point Code (OPC) : (14 bits) specifies the SP from which
the message originates.
The coding of OPC and DPC is pure binary and using 14 bits linear encoding, it is
possible to identify 16,384 exchanges. The number of exchanges in DOT network having
CCS7 capability are expected to be within this limit.
5.10.5 Signalling Link Selection (SLS) field : (4 bits) The contents of the SLS field
determine the signaling route (identifying a particular signalling link within s link set or
link sets) along which the message is to be transmitted. In this way, the SLS field is used
for load sharing on the signalling links between two SPs.
The SIO contains additional address information. Using the SI, the destination
MTP identifies the UP for which the message is intended. The NI, for example, enables a
message to be identified as being for national or international traffic.
LSSUs and FISUs require no routing label as they are only exchanged between
level 2 of adjacent MTPs.
The message sent from a user to the MTP for transmission contains : the user
information, the routing label, the SI, the NI and a LI. The processing of a user message
to be transmitted in the MTP begins in level 3 (See Fig.8).
The MTP is responsible for (a) transmitting, (b) receiving SUs, (c) for correcting
transmission errors, (d) for the signalling network management, and (e) for the alignment.
Its functions are spread over the functional levels 1, 2 and 3.
5.11

The message routing (level 3) determines the signalling link on which the

user message is to be transmitted. To do this, it analyzes the DPC and the SLS field in the
routing label of the user message, and then transfers the message to the appropriate
signalling link (level 2).
5.12

The transmission control (level 2) assigns the next FSN and the FIB to

the user message. In addition, it includes the BSN and the BIB as an acknowledgement
for the last received MSU. The transmission control simultaneously enters the part of the
MSU formed so far in the transmission and retransmission buffers. All MSUs to be
transmitted are stored in the retransmission buffer until their fault-free reception is
acknowledged by the receive side. Only then are they deleted.
5.13

The check bit and flag generator (level 2) generates CKs for

safeguarding against transmission errors for the MUS and sets the flag for separating the
SUs. In order that any section of code identical to the flag (01111110) occurring by
chance is not mistaken for the flag, the user messages are monitored before the flag is
94
94

added to see if five consecutive ones (1) appear in the message. A zero (0) is
automatically inserted after five consecutive 1s. On the receive side, the zero following
the five 1s is then automatically removed and the user message thereby regains its
original coding.
The check-bit and flag generator transfers a complete MSU to level 1. In level 1,
the MUS is sent on the signalling data link.
The bit stream along a signalling data link is received in level 1 and transferred to
level 2. Flag detection (level 2) examines the received bit stream for flags. The bit
sequence between two flags corresponds to one SU. The alignment detection (level 2)
monitors the synchronism of transmit and receive sides with the bit pattern of the flags.

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95

Using the CKs transmitted, error detection (level 2) checks whether the SU was
correctly received. A fault-free SU is transferred to the receive control, while a
faulty SU is discarded. The reception of a faulty SU is reported to error rate
monitoring, in order to keep a continuous check on the error rate on the receive
side of the signalling link. If a specified error rate is exceeded, this is reported to
the signalling link status control by error rate monitoring. The signalling link status
control then takes the signalling link out of service and sends a report to level 3.
5.14

The receive control (level 2) checks whether the transferred SU contains

the expected FSN and the expected FIB. If this is the case and if it is a MSU, the
receive control transfers the user message to level 3 and causes the reception of
the MSU to be positively acknowledged. If the FSN of the transferred MSU does
not agree with that expected, the receive control detects a transmission error and
causes this and all subsequent MSU to be retransmitted (see subheading
"Correction of Transmission Errors").
5.15

The message discrimination (level 3) accepts the correctly received user

message. It first determines whether the user message is to be delivered to one of

the immediately connected UPs or to be transferred to the another signalling link
(quasi-associated message). This pre selection is achieved in the message
discrimination by evaluation of the DPC. A user message which only passes
through a SP (STP) is transferred by the message discrimination to the message
routing, where it is treated as a user message to be transmitted.
If a received user message is intended for one of the connected UPs (SP), it is
transferred to message distribution (level 3). The message distribution evaluates
the SIO, thereby determining the UP concerned, and delivers the user message
there.
6. Signalling Network Management
The signalling network management is a function of level 3. It controls the
operation and the inter working of the individual signalling links in the signalling
network. To this end, the signalling network management exchanges messages
and control instructions with the signalling links of level 2, sends message to the
UPs and works together with the signalling network management in adjacent SPs.
For the inter working with other SPs the signalling network management uses the
transport function of the MTP.

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96

Management messages are transferred in MSUs like user messages. For

discrimination, the management messages have their own SI. The signalling
network management contains 3 function blocks :
(a)

The signalling link management controls and monitors the individual

signalling links. It receives the messages concerning the alignment and status of
the individual signalling links, or concerning operating irregularities and effects
any changes in status which may be necessary. In addition, the signalling link
management controls the putting into service of signalling links, including initial
alignment and automatic realignment of signalling links after failures or alignment
losses due to persistent faults. If necessary, the signalling link management
transfers messages to the signalling traffic management or receives instructions
from there.
(b)

The signalling route mangement controls and monitors the operability of

signalling routes. It exchanges messages with the signalling route management in

the adjacent STPs for this purpose. The signalling route management receives, for
example, messages concerning the failure or re-availability of signalling routes or
the overloading of STPs. In cooperation with the signalling traffic management, it
initiates the appropriate actions in order to maintain the signalling operation to the
signalling destinations involved.
(c)

The signalling traffic management controls the diversion of the

signalling traffic from faulty signalling links or routes to fault-free signalling links
or routes. It also controls the load distribution on the signalling links and routes.
To achieve this, it can initiate the following actions :
- changeover; on failure of a signalling link the signalling traffic management
switches the signalling traffic from the failed signalling link to a fault-free
signalling link.
- changeback; when signalling link becomes available again after a fault has been
corrected, the signalling traffic management reverse the effect of the changeover.
- rerouting; when SP can no longer be reached on a normal route, the signalling
traffic management diverts the signalling traffic to a predefined alternative route.
When overloading occurs, the signalling traffic management sends messages to
the users in its own SP in order that they reduce the load. The management also
informs the adjacent SPs of the overloading in its own SP and requests them to
also reduce the load.
The signalling traffic management accomplishes its functions by
- receiving messages from the signalling link and signalling route management.
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97

- sending control instructions to signalling link and signalling route management.

- directly accessing the signalling links, e.g. during emergency alignment.
- modifying the message routing on failure of signalling routes.
- exchanging management messages with the signalling traffic management in
adjacent SPs.
As discussed earlier, level 4 functions, which include formatting of messages
based on the applications, are allotted to UPs. Each UP provides the functions for
using the MTP for a particular user type. Some of the UPs as currently specified
by the CCITT are :
- telephone user part (TUP)
- integrated services digital network user part (ISDN-UP)
- the signalling connection control part (SCCP)
- the transaction capabilities application part (TCAP)
For Intelligent Network (IN) application, Intelligent Application Part (INAP) and
TCAP are used. SCCP forms the interface between these UPs and MTP.
Fig.9 shows the users of the MTP as well as their relationship to one another and
to the MTP. CCS7 can be adapted to all requirements due to the modular
structure. Expansion for future applications is also possible. Each CCS7 user can
specify its own UP, for example, the mobile user part (MUP) is Siemen's own
specification for the mobile telephone network C450.

Fig. 9 Message Transfer Part Users

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98

7.

MTP users
7.1.

Telephone User Part (TUP)

Use of CCS7 for telephone call control signalling requires (i) application of TUP
functions, in combination with (ii) application of an appropriate set of MTP
functions. The TUP is one of level 4 users in CCS7. It is specified with the aim of
providing the same features for telephone signalling as other telephone signalling
systems. It exchanges signalling messages through MTP. Signalling messages
contain information relating to call set up and conditions of speech path. The TUP
message consists of SIF and a SIO. These signalling information are generated by
the TUP of the originating exchange. The label is 40 bits long, comprises DPC,
OPC and CIC. CIC indicates one of the speech circuit connecting the destination
and originating points. Level 3 identifies the user to which a message belongs by
SIO, which comprises a SI and SSF. For TUP SI value is 4. The SSF distinguishes
the signalling message is for national or international network.
7.2 Integrated Services Digital Network User Part
The ISDN-UP covers the signalling functions for the control of calls, for the
processing of services and facilities and for the administration of circuits in ISDN.
The ISDN-UP has interface to the MTP and the SCCP for the transport of MSUs.
The ISDN-UP can use SCCP functions for end-to-end signalling.
CCITT SIGNALLING SYSTEM NO. 7 :
INTEGRATED SERVICES DIGITAL NETWORK USER PART
7.2.1

Overview of the ISDN User Part

The integrated services digital network user part (ISUP) is the protocol which
provides the signalling functions required by CCITT No. 7 signalling to support
basic bearer services and supplementary services for voice and non-voice
applications in an Integrated Services Digital Network (ISDN).
The ISUP is suited for application in dedicated telephone and circuit-switched
data networks and in analogue and moved analogue/digital networks. In particular,
the ISUP meets the requirements defined by the CCITT for world-wide
International semiautomatic and automatic telephone and circuit-switched data
traffic.
The ISUP can be used for national and international applications. The signalling
procedures, information elements and message type specified are for both
applications. Coding space has been reserved to allow national administrations
and recognized private operating agencies to introduce network specific signalling
messages and elements of information within the protocol structure.
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99

The ISUP makes use of the services provided by the messages transfer part (MTP)
(1) and, in some cases, by the signalling connection control part (SCCP0 of
CCITT No.7 signalling for the transfer of information between ISDN user parts.
7.2.2

Services Supported by the ISDN User Part

The ISUP protocol supports the basic bearer service; that is the establishment,
supervision and release of 64 kbit/s circuit-switched network connections between
customer line exchange terminations.
In addition to the basic bearer service the ISUP is expected to support (in the 1988
Recommendations) the following supplementary services :
Calling line identification (presentation and restriction).
Call forwarding,
Closed user group,
Direct dialling-in, and
User-to-user signalling.
Signaling Connection Control Part
Introduction: The SCCP function is covered in ITU-T recommendations Q.711 to
Q.714 and Q.716. The signalling connection control part provides additional functions to
message transfer part for transfer of circuit related and non-circuit related signalling
information and other type of information between exchanges and other specialized
centrals in telecommunications network via SS#7 networks. The overall objective of
SSCP is to provide means for:
A transfer capability for signalling data units with or without the use of logical signalling
connections.
A logical signalling connection between two SCCP users with the SS#7 network.
Enhanced addressing capabilities.
The following figure illustrates the SCCP position in the SS#7 hierarchy:

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1001

The functions of SCCP are used for handling transactions required by TCAP and
also for transfer of circuit related and call related signalling information for ISDN
UP with or without set up of end-to-end logical signalling connections.
The SCCP relies on the MTP to route the signalling information from one node to
another node. For this, it interacts with the user parts and with the MTP.
Primitives are used to convey information between the levels. Primitives are
nothing but set of commands and their respective responses associated with the
services requested of the SCCP.
SCCP and OSI model
The SCCP enhances the services of MTP to provide the functional equivalent of
Network layer (i.e. layer #3 of OSI model). The MTP and the SCCP together is
also referred to as Network Service Part (NSP).
SCCP Addressing
The addressing capability of MTP is limited to delivering the message to a node
(identified by Network indicator and DPC) and to distribute it to a user using four
bit service indicator (octet SIO ). SCCP supplements this capability by providing
an addressing capability that uses DPC + SSN .The SSN is a local addressing
information used by SCCP to identify each of the SCCP users at a node.
SCCP provides enhanced addressing capability to MTP to enable it to address
messages with Global Title (GT). A Global Title is an address that does not
explicitly contain information usable for routing by MTP.

SC
CP
Addressin
g

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1011

SCCP provides enhanced addressing capability to MTP to enable it to address

messages with Global Title (GT). A Global Title is an address that does not
explicitly contain information usable for routing by MTP.
Global Title
The SS#7 signalling method identifies the destination and origination using
signalling point codes. Since a signalling point code has only fourteen bits. It is too
small to be uniquely addressed on a global scale. For this reason, signalling point
codes are always combined with a Network indicator- which means that a code is
only valid in one particular network.
To facilitate unambiguous global addressing, a unique international address or
sender information is necessary. The global address is known as The Global title
and is sent in the SCCP message in SS#7 messages. Since, SPCs are only ever
valid in individual networks, so called Global title translation must be performed at
each relevant network gateway. The SCCP performs the Global title translation,
whereby an internationally unique address (Global Title) is translated to an SPC
and Network indicator in order to be transferred to the network border or to a
destination if it is located in the same network.
Global title translation is always used if no SPC for the destination is available at
all. For example, the HLR is to be identified on the basis of the IMSI.
SCCP Functional Units
The services supported by
SCCP are divided into two groups viz. Connection-oriented services and
Connectionless services. The protocols used for providing these services are
divided in four classes; two for connectionless services and two for connectionoriented services. Each protocol class defines which level of services SCCP to
provide. The four protocol classes are described below:
Connectionless services
Class 0 Basic connectionless
Class 1 Sequenced connectionless
Connection oriented services
Class 2 Basic connection oriented
Class 3 Flow control connection oriented
The first two classes 0 and 1, support the connectionless environment, for example,
for use by TCAP. These are particularly suitable for frequent transmission of short
messages. As an example, to check validity of the credit card, an interrogation
message can sent to a data centre and reply received on the same route. The
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1021

connectionless services are all that is used in todays networks. Classes 2 and 3 are
used for connection-oriented services, for example by ISDNUP and, even though
well-defined, are not used in todays network.
The SCCP is divided into four functional units:
SCCP routing control (SCRC).
SCCP connectionless control (SCLC).
SCCP connection-oriented control.
SCCP management control (SCMG).

SCCP Functional Units

Transaction Capabilities Application Part (TCAP)

The TCAP recommendations are covered in ITU-T Q.771 to Q.775. The TCAP
portion of the CCS#7 protocol is used to transfer non-circuit related information
between two signalling points in the network. It is used to communicate between
the SSP, SCP or other SSPs through an exchange of TCAP messages. There is no
setup of speech/data channel connections.
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Non-circuit related information would be such things as data queries for

services (1600) where there is not a physical end-to-end connection between the
signalling points
TCAP supports the exchange of non-circuit related data between applications
across the SS7 network using the SCCP connectionless service. Queries and
responses sent between SSPs and SCPs are carried in TCAP messages. For
example, an SSP sends a TCAP query to determine the routing number associated
with a dialed 1600 number and to check the personal identification number (PIN)
of a calling card user. In mobile networks (IS-41 and GSM), TCAP carries Mobile
Application Part (MAP) messages sent between mobile switches and databases to
support user authentication, equipment identification, and roaming.
Applications for the TCAP
In mobile networks to report the location of a mobile network subscriber to
the home exchanges.
In credit card service to check the validity and to execute account
transactions.
Functions of TCAP
TCAP supports real-time remote operations and is structured in two sublayers:
i) Component sub-layer, dealing with individual actions called components.
ii)Transaction sub-layer,dealing with the exchange of messages containing
components
The component sub-layer is above the transaction sub-layer.
The TCAP layer interfaces directly with SCCP layer.
A component consists of a request to invoke an operation.
An invocation of the operation is identified by a Component ID.
Components are passed individually between TCAP users.
The originating TC user may send several components to the component
sub-layer before they are transmitted in a single message to the remote end. At the
remote end each one is delivered individually to the destinating TC-user.
Successive component exchanged between TC-users in order to perform an
application constitute a dialogue. The component sub-layer allows several
dialogues to run concurrently between two TC-users each being identified by a
particular ID.

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1041

TCAP serves all ( application specific0 ASEs in a node.To send a message an

ASE passes a series of TC requests to TCAP and TCAP passes the message to
SCCP. When a TCAP receives a message from its SCCP it passes the contents to
the destination ASE in its node.

ASE1
TC-primitives
TCAP-A

Component

TCAP messages

ASE2
TC primitives
TCAP-B

N-primitives

N-primitive
SCCP-A

SCCP-B

MTP-primitives

MTP-primitives

MTP-A

MTP-B
MSUs

Messages and message paths

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CHAPTER - 6

FIBRE-OPTIC COMMUNICATION TECHNOLOGY

1.0

A Brief History of Fiber-Optic Communications

Optical communication systems date back to the 1790s, to the optical semaphore

telegraph invented by French inventor Claude Chappe. In 1880, Alexander Graham Bell
patented an optical telephone system, which he called the Photophone. However, his
earlier invention, the telephone, was more practical and took tangible shape.
By 1964, a critical and theoretical specification was identified by Dr. Charles K.
Kao for long-range communication devices, the 10 or 20 dB of light loss per kilometer
standard. Dr. Kao also illustrated the need for a purer form of glass to help reduce light
loss. By 1970 Corning Glass invented fiber-optic wire or "optical waveguide fibers"
which was capable of carrying 65,000 times more information than copper wire, through
which information carried by a pattern of light waves could be decoded at a destination
even a thousand miles away. Corning Glass developed an SMF with loss of 17 dB/km at
633 nm by doping titanium into the fiber core. By June of 1972, multimode germaniumdoped fiber had developed with a loss of 4 dB per kilometer and much greater strength
than titanium-doped fiber. Prof. Kao was awarded half of the 2009 Nobel Prize in Physics
for "groundbreaking achievements concerning the transmission of light in fibers for
optical communication". In April 1977, General Telephone and Electronics tested and
deployed the world's first live telephone traffic through a fiber-optic system running at 6
Mbps, in Long Beach, California. They were soon followed by Bell in May 1977, with an
optical telephone communication system installed in the downtown Chicago area,
covering a distance of 1.5 miles (2.4 kilometers). Each optical-fiber pair carried the
equivalent of 672 voice channels and was equivalent to a DS3 circuit. Today more than
80 percent of the world's long-distance voice and data traffic is carried over optical-fiber
cables.

2.0

Fiber-Optic Applications
FIBRE OPTICS: The use and demand for optical fiber has grown tremendously

and optical-fiber applications are numerous. Telecommunication applications are

widespread, ranging from global networks to desktop computers. These involve the
transmission of voice, data, or video over distances of less than a meter to hundreds of
kilometers, using one of a few standard fiber designs in one of several cable designs.
Carriers use optical fiber to carry plain old telephone service (POTS) across their
nationwide networks. Local exchange carriers (LECs) use fiber to carry this same service

between central office switches at local levels, and sometimes as far as the neighborhood
or individual home (fiber to the home [FTTH]).
Optical fiber is also used extensively for transmission of data. Multinational firms
need secure, reliable systems to transfer data and financial information between buildings
to the desktop terminals or computers and to transfer data around the world. Cable
television companies also use fiber for delivery of digital video and data services. The
high bandwidth provided by fiber makes it the perfect choice for transmitting broadband
signals, such as high-definition television (HDTV) telecasts. Intelligent transportation
systems, such as smart highways with intelligent traffic lights, automated tollbooths, and
changeable message signs, also use fiber-optic-based telemetry systems.
Another important application for optical fiber is the biomedical industry. Fiber-optic
systems are used in most modern telemedicine devices for transmission of digital
diagnostic images. Other applications for optical fiber include space, military,
automotive, and the industrial sector.

3.0

ADVANTAGES OF FIBRE OPTICS :

Fibre Optics has the following advantages :

SPEED: Fiber optic networks operate at high speeds - up into the
gigabits
BANDWIDTH: large carrying capacity
DISTANCE: Signals can be transmitted further without needing to be "refreshed"
or strengthened.
RESISTANCE: Greater resistance to electromagnetic noise such as radios, motors
or other nearby cables.
MAINTENANCE: Fiber optic cables costs much less to
maintain.

4.0

Fiber Optic System :

Optical Fibre is new medium, in which information (voice, Data or Video) is

transmitted through a glass or plastic fibre, in the form of light, following the
transmission sequence give below :
(1)

Information is Encoded into Electrical Signals.

(2)

Electrical Signals are Coverted into light

Signals. (3)
(4)

A Detector Changes the Light Signals into Electrical

Signals. (5)
-

Light Travels Down the Fiber.

Electrical Signals are Decoded into Information.

Inexpensive light sources available.

-

Repeater spacing increases along with operating speeds because low loss
fibres are used at high data rates.

Fig. 1
5.0
Principle of Operation - Theory
Total Internal Reflection - The Reflection that Occurs when a Ligh Ray Travelling in One
Material Hits a Different Material and Reflects Back into the Original Material without
any Loss of Light.
Fig. 2

Speed of light is actually the velocity of electromagnetic energy in vacuum such as space.
Light travels at slower velocities in other materials such as glass. Light travelling from one
material to another changes speed, which results in light changing its direction of
travel. This deflection of light is called Refraction.
The amount that a ray of light passing from a lower refractive index to a higher one is bent
towards the normal. But light going from a higher index to a lower one refracting
away from the normal, as shown in the figures.

Angle of incidence

2
Light is bent
away from
normal

n1

n1

n2

n2

Angle of
reflectio
n

n1
2

n2

Light does not

enter second
material

Fig. 3
As the angle of incidence increases, the angle of refraction approaches 90o to the normal.
The angle of incidence that yields an angle of refraction of 90o is the critical angle. If the
angle of incidence increases amore than the critical angle, the light is totally reflected back
into the first material so that it does not enter the second material. The angle of
incidence and reflection are equal and it is called Total Internal Reflection.
6.0

PROPAGATION OF LIGHT THROUGH FIBRE

The optical fibre has two concentric layers called the core and the cladding. The inner
core is the light carrying part. The surrounding cladding provides the difference refractive
index that allows total internal reflection of light through the core. The index of the
cladding is less than 1%, lower than that of the core. Typical values for example are a core
refractive index of 1.47 and a cladding index of 1.46. Fibre manufacturers control this
difference to obtain desired optical fibre characteristics. Most fibres have an additional
coating around the cladding. This buffer coating is a shock absorber and has no optical
properties affecting the propagation of light within the fibre. Figure shows the idea of
light travelling through a fibre. Light injected into the fibre and striking core to cladding
interface at grater than the critical angle, reflects back into core, since the angle of
incidence and reflection are equal, the reflected light will again be reflected. The light
will continue zigzagging down the length of the fibre. Light striking the interface at less
than the critical angle passes into the cladding, where it is lost over distance. The cladding
is usually inefficient as a light carrier, and light in the cladding becomes attenuated fairly.
Propagation of light through fibre is governed by the indices of the core and cladding by
Snell's law.
Such total internal reflection forms the basis of light propagation through a optical
fibre. This analysis consider only meridional rays- those that pass through the fibre

axis each time, they are reflected. Other rays called Skew rays travel down the fibre
without passing

through the axis. The path of a skew ray is typically helical wrapping around and
around the central axis. Fortunately skew rays are ignored in most fibre optics analysis.
The specific characteristics of light propagation through a fibre depends on many factors,
including
-

The size of the fibre.

The composition of the fibre.

The light injected into the fibre.

Jacket

Jacket
Cladding
Core

Cladding (n
2)

Cladding

Core (n2)

Jacket
Light at less than
critical angle is
absorbed in
jacket

Angle of Angle of
incidence reflection
Light is propagated by
total internal reflection

Fig. 4 Propagation of light through fiber

7.0

Geometry of Fiber
A hair-thin fiber consist of two concentric layers of high-purity silica glass the

core and the cladding, which are enclosed by a protective sheath as shown in Fig. 5. Light
rays modulated into digital pulses with a laser or a light-emitting diode moves along the
core without penetrating the cladding.

Fig. 5 Geometry of fiber

The light stays confined to the core because the cladding has a lower refractive index
a measure of its ability to bend light. Refinements in optical fibers, along with
the

1101
101

development of new lasers and diodes, may one day allow commercial fiber-optic
networks to carry trillions of bits of data per second.
The diameters of the core and cladding are as follows.

125 8

Core ( m)

Cladding ( m)

125

50

125

62.5

125

100

140

125 62.5

125 50

125 100

Core
Cladding
Typical Core and Cladding Diameters

Fibre sizes are usually expressed by first giving the core size followed by the cladding
size. Thus 50/125 means a core diameter of 50 m and a cladding diameter of 125 m.
8.0
FIBRE TYPES
The refractive Index profile describes the relation between the indices of the core
and cladding. Two main relationship exists :
(I)

Step Index

(II)

Graded Index

The step index fibre has a core with uniform index throughout. The profile shows a sharp
step at the junction of the core and cladding. In contrast, the graded index has a
non- uniform core. The Index is highest at the center and gradually decreases until it
matches with that of the cladding. There is no sharp break in indices between the core and
the cladding.
By this classification there are three types of fibres :
(I)
fibre)

Multimode Step Index fibre (Step Index

(II)

Multimode graded Index fibre (Graded Index

fibre) (III)
Fibre)

Single- Mode Step Index fibre (Single Mode

8.1

STEP-INDEX MULTIMODE FIBER has a large core, up to 100 microns in

diameter. As a result, some of the light rays that make up the digital pulse may travel a
direct route, whereas others zigzag as they bounce off the cladding. These alternative
pathways cause the different groupings of light rays, referred to as modes, to arrive
separately at a receiving point. The pulse, an aggregate of different modes, begins to
spread out, losing its well-defined shape. The need to leave spacing between pulses to
prevent overlapping limits bandwidth that is, the amount of information that can be sent.
Consequently, this type of fiber is best suited for transmission over short distances, in an
endoscope, for instance.

Fig. 6 STEP-INDEX MULTIMODE FIBER

8.2

GRADED-INDEX MULTIMODE FIBER contains a core in which the

refractive index diminishes gradually from the center axis out toward the cladding. The
higher refractive index at the center makes the light rays moving down the axis advance
more slowly than those near the cladding.

Fig.7 GRADED-INDEX MULTIMODE FIBER

Also, rather than zigzagging off the cladding, light in the core curves helically
because of the graded index, reducing its travel distance. The shortened path and the
higher speed allow light at the periphery to arrive at a receiver at about the same time as
the slow but straight rays in the core axis. The result: a digital pulse suffers less
dispersion.
8.3

SINGLE-MODE FIBER has a narrow core (eight microns or less), and the
index of refraction between the core and the cladding changes less than it does
for multimode fibers. Light thus travels parallel to the axis, creating little pulse
dispersion. Telephone and cable television networks install millions of kilometers
of this fiber every year.

Fig. 8 SINGLE-MODE FIBER

9.0

OPTICAL FIBRE PARAMETERS

Optical fiber systems have the following parameters.

(I)

Wavelength.

(II)

Frequency.

(III)

Window.

(IV)

Attenuation.

(V)

Dispersion.

(VI)

Bandwidth.

9.1

WAVELENGTH

It is a characterstic of light that is emitted from the light source and is measures in
nanometers (nm). In the visible spectrum, wavelength can be described as the colour of
the light.
For example, Red Light has longer wavelength than Blue Light, Typical wavelength for
fibre use are 850nm, 1300nm and 1550nm all of which are invisible.
9.2

FREQUENCY

It is number of pulse per second emitted from a light source. Frequency is measured in
units of hertz (Hz). In terms of optical pulse 1Hz = 1 pulse/ sec.
9.3

WINDOW
A narrow window is defined as the range of wavelengths at which a fibre best
operates. Typical windows are given below :

9.4

Window

Operational Wavelength

800nm - 900nm

850nm

1250nm - 1350nm

1300nm

1500nm - 1600nm

1550nm

ATTENUATION

Attenuation is defined as the loss of optical power over a set distance, a fibre with lower
attenuation will allow more power to reach a receiver than fibre with higher attenuation.
Attenuation may be categorized as intrinsic or extrinsic.

9.4.1

INTRINSIC ATTENUATION

It is loss due to inherent or within the fibre. Intrinsic attenuation may occur as
Absorption - Natural Impurities in the glass absorb light energy.

Light
Ray

Fig. 9 Absorption of Light

(1)

Scattering - Light Rays Travelling in the Core Reflect from small Imperfections

into a New Pathway that may be Lost through the cladding.

Ligh t is lost

L ig ht
R ay

Fig. 10 Scattering
9.4.2

EXTRINSIC ATTENUATION

It is loss due to external sources. Extrinsic attenuation may occur as

(I)

Macrobending - The fibre is sharply bent so that the light travelling down
the fibre cannot make the turn & is lost in the cladding.

Fig. 11 Micro and Macro bending

(II)
Microbending - Microbending or small bends in the fibre caused by
crushing contraction etc. These bends may not be visible with the naked eye.
Attenuation is measured in decibels (dB). A dB represents the comparison between the
transmitted and received power in a system.
9.5

BANDWIDTH

It is defined as the amount of information that a system can carry such that each pulse of
light is distinguishable by the receiver.
System bandwidth is measured in MHz or GHz. In general, when we say that a system
has bandwidth of 20 MHz, means that 20 million pulses of light per second will travel
down the fibre and each will be distinguishable by the receiver.
9.6

NUMBERICAL APERTURE

Numerical aperture (NA) is the "light - gathering ability" of a fibre. Light injected into
the fibre at angles greater than the critical angle will be propagated. The material NA
relates to the refractive indices of the core and cladding.
NA = n12 - n22
where n1 and n2 are refractive indices of core and cladding respectively.
NA is unitless dimension. We can also define as the angles at which rays will be
propagated by the fibre. These angles form a cone called the acceptance cone, which
gives the maximum angle of light acceptance. The acceptance cone is related to the NA
NA
where

arc sing (NA) or

sin
is the half angle of acceptance

The NA of a fibre is important because it gives an indication of how the fibre accepts and
propagates light. A fibre with a large NA accepts light well, a fibre with a low NA
requires highly directional light.
In general, fibres with a high bandwidth have a lower NA. They thus allow fewer modes
means less dispersion and hence greater bandwidth. A large NA promotes more modal
dispersion, since more paths for the rays are provided NA, although it can be defined for
a single mode fibre, is essentially meaningless as a practical characteristic. NA in a

multimode fibre is important to system performance and to calculate anticipated

performance.
Fig. 12 Numerical Aperture of fiber
* Light Ray A : Did not Enter Acceptance Cone - Lost
* Light Ray B : Entered Acceptance Cone - Transmitted through the Core by Total
Internal Reflection.
9.7

DISPERSION

Dispersion is the spreading of light pulse as its travels down the length of an optical fibre
as shown in figure 13. Dispersion limits the bandwidth or information carrying capacity
of a fibre. The bit-rates must be low enough to ensure that pulses are farther apart and
therefore the greater dispersion can be tolerated.
There are three main types of dispersion in a fibre (I)

Modal Dispersion

(II)

Material dispersion

(III)

Waveguide dispersion

Fig. 13 Dispersion
9.8

BANDWIDTH AND DISPERSION :

A bandwidth of 400 MHz -km means that a 400 MHz-signal can be transmitted for 1 km.
It means that the product of frequency and the length must be 400 or less. We can send a
lower frequency for a longer distance, i.e. 200 MHz for 2 km or 100 MHz for 4 km.
Multimode fibres are specified by the bandwidth-length product or simply bandwidth.

Single mode fibres on the other hand are specified by dispersion, expressed in ps/km/nm.
In other words for any given single mode fibre dispersion is most affected by the source's
spectral width. The wider the source spectral width, the greater the dispersion.
Conversion of dispersion to bandwidth can be approximated roughly by the following
equation.
0.187
BW

-------------------------(Disp) (SW) (L)

Disp

Dispersion at the operating wavelength in seconds/ nm/ km.

SW

Spectral width of the source in nm.

Fibre length in km.

So the spectral width of the source has a significant effect on the performance of a single
mode fibre.
9.9

OPTICAL WINDOWS :

Attenuation of fibre for optical power varies with the wavelengths of light. Windows are
low-loss regions, where fiber carry light with little attenuation. The first generation of
optical fibre operated in the first window around 820 to 850 nm. The second window is
the zero-dispersion region of 1300 nm and the third window is the 1550 nm region as
shown in figure 14.

Fig. 14 Optical Windows

10.0 CABLE CONSTRUCTION

There are two basic cable designs are:
1. Tight Buffer Tube Cable

2. Loose Buffer Tube Cable

Loose-tube cable, used in the majority of outside-plant installations and tight-buffered
cable, primarily used inside buildings.

10.1 Tight-Buffered Cable

With tight-buffered cable designs, the buffering material is in direct contact with
the fiber. This design is suited for "jumper cables" which connect outside plant cables to
terminal equipment, and also for linking various devices in a premises network. Singlefiber tight-buffered cables are used as pigtails, patch cords and jumpers to terminate
loose-tube cables directly into opto-electronic transmitters, receivers and other active and
passive components.
Multi-fiber tight-buffered cables also are available and are used primarily for
alternative routing and handling flexibility and ease within buildings.The tight-buffered
design provides a rugged cable structure to protect individual fibers during handling,
routing and connectorization. Yarn strength members keep the tensile load away from the
fiber.

Fig. 15 Tight Buffer Tube Cable

10.2 Loose-Tube Cable

The modular design of loose-tube cables typically holds 6, 12, 24, 48, 96 or even
more than 400 fibers per cable. Loose-tube cables can be all-dielectric or optionally
armored. The loose-tube design also helps in the identification and administration of
fibers in the system.
In a loose-tube cable design, color-coded plastic buffer tubes house and protect
optical fibers. A gel filling compound impedes water penetration. Excess fiber length
(relative to buffer tube length) insulates fibers from stresses of installation and

environmental loading. Buffer tubes are stranded around a dielectric or steel central
member, which serves as an anti-buckling element.
The cable core, typically uses aramid yarn, as the primary tensile strength
member. The outer polyethylene jacket is extruded over the core. If armoring is required,
a corrugated steel tape is formed around a single jacketed cable with an additional jacket
extruded over the armor.
Loose-tube cables typically are used for outside-plant installation in aerial, duct
and direct-buried applications.
Here are some common fiber cables types are given below:
10.2. 1 Distribution Cable
Distribution Cable (compact building cable) packages individual 900 m buffered
fiber reducing size and cost. The connectors may be installed directly on the 900 m
buffered fiber at the breakout box location.

Fig. 16 Distribution Cable

10.2.2 Loose Tube Cable
Loose tube cable is designed to endure outside temperatures and high moisture
conditions. The fibers are loosely packaged in gel filled buffer tubes to repel water.
Recommended for use between buildings that are unprotected from outside elements.
Loose tube cable is restricted from inside building use.

Fig.17 Loose Tube Cable

10.2.3 Aerial Cable/Self-Supporting
Aerial cable provides ease of installation and reduces time and cost. Figure 8
cable can easily be separated between the fiber and the messenger. Temperature range
(-55C to +85C)

Fig. 18 Aerial Cable/Self-Supporting

10.2.4 Hybrid & Composite Cable
Hybrid cables offer the same great benefits as our standard indoor/outdoor cables,
with the convenience of installing multimode and single mode fibers all in one pull. Our
composite cables offer optical fiber along with solid 14 gauge wires suitable for a variety
of uses including power, grounding and other electronic controls

Fig. 19 Hybrid & Composite Cable

10.2.5 Armored Cable
Armored cable can be used for rodent protection in direct burial if required. This
cable is non-gel filled and can also be used in aerial applications. The armor can be
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removed leaving the inner cable suitable for any indoor/outdoor use. (Temperature rating
-40C to +85C)

Fig. 20 Armored Cable

Fibre Optic Cables (Loose Buffer Tube) have the following parts in common ;
(I)

Optical Fibre

(II)

Buffer

(III)

Strength member

(IV)

Jacket

Table-1 Cable Components

Component

Function

Material

Buffer

Protect fibre From Outside

Nylon, Mylar, Plastic

Facilitate Stranding
Central Member

Temperature Stability

Steel, Fibreglass

Anti-Buckling
Primary Strength
Member
Cable Jacket

Tensile Strength
Contain and Protect
Cable Core

Aramid Yarn, Steel

PE, PUR, PVC, Teflon

Abrasion Resistance

Cable Filling

Prevent Moisture

Water Blocking

Compound

intrusion and Migration

Compound

Armoring
11.0

Rodent Protection
Crush Resistance

Steel Tape

CABLE DRUM LENGTH :

Cables come reeled in various length, typically 1 to 2 km, although lengths of 5 or 6 kms
are available for single mode fibres. Long lengths are desirables for long distance
applications, since cable must be spliced end to end over the run. Each splice introduce
additional loss into the system. Long cable lengths mean fewer splices and less loss.

12.0 OFC Splicing

Splices are permanent connection between two fibres. The splicing involves
cutting of the edges of the two fibres to be spliced.
Splicing Methods
The following three types are widely used :

12.1

1.

Adhesive bonding or Glue splicing.

2.

Mechanical splicing.

3.

Fusion splicing.

Adhesive Bonding or Glue Splicing

This is the oldest splicing technique used in fibre splicing. After fibre end

preparation, it is axially aligned in a precision Vgroove. Cylindrical rods or another kind

of reference surfaces are used for alignment. During the alignment of fibre end, a small
amount of adhesive or glue of same refractive index as the core material is set between
and around the fibre ends. A two component epoxy or an UV curable adhesive is used as
the bonding agent. The splice loss of this type of joint is same or less than fusion splices.
But fusion splicing technique is more reliable, so at present this technique is very rarely
used.
12.2

Mechanical Splicing
This technique is mainly used for temporary splicing in case of emergency

repairing. This method is also convenient to connect measuring instruments to bare fibres
for taking various measurements.
The mechanical splices consist of 4 basic components :
(i)

An alignment surface for mating fibre ends.

(ii)

A retainer

(iii)

An index matching material.

(iv)

A protective housing

A very good mechanical splice for M.M. fibres can have an optical performance
as good as fusion spliced fibre or glue spliced. But in case of single mode fibre, this type
of splice cannot have stability of loss.
12.3

Fusion Splicing
The fusion splicing technique is the most popular technique used for achieving

very low splice losses. The fusion can be achieved either through electrical arc or through
gas flame.
The process involves cutting of the fibres and fixing them in micropositioners on
the fusion splicing machine. The fibres are then aligned either manually or automatically
core aligning (in case of S.M. fibre) process. Afterwards the operation that takes place
involve withdrawal of the fibres to a specified distance, preheating of the fibre ends
through electric arc and bringing together of the fibre ends in a position and splicing
through high temperature fusion.
If proper care taken and splicing is done strictly as per schedule, then the splicing
loss can be minimized as low as 0.01 dB/joint. After fusion splicing, the splicing joint
should be provided with a proper protector to have following protections:
(a)

Mechanical protection

(b)

Protection from moisture.

Sometimes the two types of protection are combined. Coating with Epoxy resins
protects against moisture and also provides mechanical strength at the joint.
Nowadays, the heat shrinkable tubes are most widely used, which are fixed on
the joints by the fusion tools.
The fusion splicing technique is the most popular technique used for achieving
very low splice losses. The introduction of single mode optical fibre for use in long haul
network brought with it fibre construction and cable design different from those of
multimode fibres.
The splicing machines imported by BSNL begins to the core profile alignment
system, the main functions of which are :
(1)

Auto active alignment of the core.

(2)

Auto arc fusion.

(3)

Video display of the entire process.

(4)

Indication of the estimated splice loss.

The two fibres ends to be spliced are cleaved and then clamped in accurately
machined veegrooves. When the optimum alignment is achieved, the fibres are fused
under the microprocessor contorl, the machine then measures the radial and angular off
sets of the fibres and uses these figures to calculate a splice loss. The operation of the
machine observes the alignment and fusion processes on a video screens showing
horizontal and vertical projection of the fibres and then decides the quality of the splice.
The splice loss indicated by the splicing machine should not be taken as a final
value as it is only an estimated loss and so after every splicing is over, the splice loss
measurement is to be taken by an OTDR (Optical Time Domain Reflectometer). The
manual part of the splicing is cleaning and cleaving the fibres. For cleaning the fibres,
Dichlorine Methyl or Acetone or Alcohol is used to remove primary coating.
With the special fibre cleaver or cutter, the cleaned fibre is cut. The cut has to be
so precise that it produces an end angle of less than 0.5 degree on a prepared fibre. If the
cut is bad, the splicing loss will increase or machine will not accept for splicing. The
shape of the cut can be monitored on the video screen, some of the defect noted while
cleaving are listed below :
(i)

Broken ends.

(ii)

Ripped ends.

(iii)

Slanting cuts.

(iv)

Unclean ends.

It is also desirable to limit the average splice loss to be less than 0.1 dB.

CHAPTER - 7

OF TRANSMISSION SYSTEMS & THEIR FEATURES

1.0

INTRODUCTION

With the introduction of PCM technology in the 1960s, communications networks were
gradually converted to digital technology over the next few years. To cope with the
demand for ever higher bit rates, a multiplex hierarchy called the plesiochronous digital
hierarchy (PDH) evolved. The bit rates start with the basic multiplex rate of 2 Mbit/s with
further stages of 8, 34 and 140 Mbit/s. In North America and Japan, the primary rate is
1.5 Mbit/s. Hierarchy stages of 6 and 44 Mbit/s developed from this. Because of these
very different developments, gateways between one network and another were very
difficult and expensive to realize. PCM allows multiple use of a single line by means of
digital time-domain multiplexing. The analog telephone signal is sampled at a bandwidth
of 3.1 kHz, quantized and encoded and then transmitted at a bit rate of 64 kbit/s.

transmission rate of 2048 kbit/s results when 30 such coded channels are collected
together into a frame along with the necessary signaling information. This so-called
primary rate is used throughout the world. Only the USA, Canada and Japan use a
primary rate of 1544 kbit/s, formed by combining 24 channels instead of 30. The growing
demand for more bandwidth meant that more stages of multiplexing were needed
throughout the world. A practically synchronous (or, to give it its proper name:
plesiochronous) digital hierarchy is the result. Slight differences in timing signals mean
that justification or stuffing is necessary when forming the multiplexed signals. Inserting
or dropping an individual 64 kbit/s channel to or from a higher digital hierarchy requires
a considerable amount of complex multiplexer equipment.
Traditionally, digital transmission systems and hierarchies have been based on
multiplexing signals which are plesiochronous (running at almost the same speed). Also,
various parts of the world use different hierarchies which lead to problems of
international interworking; for example, between those countries using 1.544 Mbit/s
systems (U.S.A. and Japan) and those using the 2.048 Mbit/s system. To recover a 64
kbit/s channel from a 140 Mbit/s PDH signal, its necessary to demultiplex the signal all
the way down to the 2 Mbit/s level before the location of the 64 kbit/s channel can be
identified. PDH requires steps (140-34, 34-8, 8-2 demultiplex; 2-8, 8-34, 34-140
multiplex) to drop out or add an individual speech or data channel (see Figure 1).

Fig. 1 Plesiochronous Digital Hierarchies (PDH)

The main problems of PDH systems are:
1. Homogeneity of equipment
2. Problem of Channel segregation
3. The problem cross connection of channels
4. Inability to identify individual channels in a higher-order bit stream.
5. Insufficient capacity for network management;
6. Most PDH network management is proprietary.
7. Theres no standardized definition of PDH bit rates greater than 140 Mb/s.
8. There are different hierarchies in use around the world. Specialized interface
equipment is required to interwork the two hierarchies.
1988 SDH standard introduced with three major goals:
Avoid the problems of PDH
Achieve higher bit rates (Gbit/s)
Better means for Operation, Administration, and Maintenance (OA&M)

SDH is an ITU-T standard for a high capacity telecom network. SDH is a synchronous
digital transport system, aim to provide a simple, economical and flexible telecom
infrastructure. The basis of Synchronous Digital Hierarchy (SDH) is synchronous
multiplexing - data from multiple tributary sources is byte interleaved.
SDH brings the following advantages to network providers:
1.1

High transmission rates

Transmission rates of up to 40 Gbit/s can be achieved in modern SDH systems. SDH is

therefore the most suitable technology for backbones, which can be considered as being
the super highways in today's telecommunications networks.
1.2

Simplified add & drop function

Compared with the older PDH system, it is much easier to extract and insert low-bit rate
channels from or into the high-speed bit streams in SDH. It is no longer necessary to
demultiplex and then remultiplex the plesiochronous structure.
1.3

High availability and capacity matching

With SDH, network providers can react quickly and easily to the requirements of their
customers. For example, leased lines can be switched in a matter of minutes. The network
provider can use standardized network elements that can be controlled and monitored
from a central location by means of a telecommunications network management (TMN)
system.
1.4

Reliability

Modern SDH networks include various automatic back-up and repair mechanisms to
cope with system faults. Failure of a link or a network element does not lead to failure of
the entire network which could be a financial disaster for the network provider. These
back-up circuits are also monitored by a management system.
1.5
Future-proof platform for new services
Right now, SDH is the ideal platform for services ranging from POTS, ISDN and mobile
radio through to data communications (LAN, WAN, etc.), and it is able to handle the
very latest services, such as video on demand and digital video broadcasting via ATM
that are gradually becoming established.
1.6

Interconnection

SDH makes it much easier to set up gateways between different network providers and to
SONET systems. The SDH interfaces are globally standardized, making it possible to
combine network elements from different manufacturers into a network. The result is a
reduction in equipment costs as compared with PDH.
2.0
Network Elements of SDH
Figure 2 is a schematic diagram of a SDH ring structure with various tributaries. The
mixture of different applications is typical of the data transported by SDH. Synchronous

networks must be able to transmit plesiochronous signals and at the same time be capable
of handling future services such as ATM.
Current SDH networks are basically made up from four different types of network
element. The topology (i.e. ring or mesh structure) is governed by the requirements of the
network provider.
2.1

Regenerators

Regenerators as the name implies, have the job of regenerating the clock and amplitude
relationships of the incoming data signals that have been attenuated and distorted by
dispersion. They derive their clock signals from the incoming data stream. Messages are
received by extracting various 64 kbit/s channels (e.g. service channels E1, F1) in the
RSOH (regenerator section overhead). Messages can also be output using these channels.
2.2

Terminal Multiplexer

Terminal multiplexers Terminal multiplexers are used to combine plesiochronous and

synchronous input signals into higher bit rate STM-N signals.

Fig. 2 Schematic diagram of hybrid communications networks

2.3

Add/drop Multiplexers(ADM)

Add/drop multiplexers (ADM) Plesiochronous and lower bit rate synchronous signals can
be extracted from or inserted into high speed SDH bit streams by means of ADMs. This
feature makes it possible to set up ring structures, which have the advantage that
automatic back-up path switching is possible using elements in the ring in the event of a
fault.
2.4

Digital Cross-connect
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Digital cross-connects (DXC) This network element has the widest range of functions. It
allows mapping of PDH tributary signals into virtual containers as well as switching of
various containers up to and including VC-4.
2.5

Network Element Manager

Network element management The telecommunications management network (TMN) is

considered as a further element in the synchronous network. All the SDH network
elements mentioned so far are software-controlled. This means that they can be
monitored and remotely controlled, one of the most important features of SDH. Network
management is described in more detail in the section TMN in the SDH network

3.0

SDH Rates

SDH is a transport hierarchy based on multiples of 155.52 Mbit/s. The basic unit of SDH
is STM-1. Different SDH rates are given below:
STM-1 = 155.52 Mbit/s
STM-4 = 622.08 Mbit/s
STM-16 = 2588.32 Mbit/s
STM-64 = 9953.28 Mbit/s
Each rate is an exact multiple of the lower rate therefore the hierarchy is synchronous.
4.0
Back-up network switching- Automatic protection switching (APS)
Modern society is virtually completely dependent on communications technology. Trying
to imagine a modern office without any connection to telephone or data networks is like
trying to work out how a laundry can operate without water. Network failures, whether
due to human error or faulty technology, can be very expensive for users and network
providers alike. As a result, the subject of so-called fall-back mechanisms is currently one
of the most talked about in the SDH world. A wide range of standardized mechanisms is
incorporated into synchronous networks in order to compensate for failures in network
elements.
Two basic types of protection architecture are distinguished in APS. One is the linear
protection mechanism used for point-to-point connections. The other basic form is the socalled ring protection mechanism which can take on many different forms. Both
mechanisms use spare circuits or components to provide the back-up path. Switching is
controlled by the overhead bytes K1 and K2.

4.1

Linear protection

The simplest form of back-up is the so-called 1 + 1 APS. Here, each working line is
protected by one protection line. If a defect occurs, the protection agent in the network
elements at both ends switch the circuit over to the protection line. The switchover is
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triggered by a defect such as LOS. Switching at the far end is initiated by the return of an
acknowledgment in the backward channel. 1+1 architecture includes 100% redundancy,
as there is a spare line for each working line. Economic considerations have led to the
preferential use of 1:N architecture, particularly for long-distance paths. In this case,
several working lines are protected by a single back-up line. If switching is necessary, the
two ends of the affected path are switched over to the back-up line. The 1+1 and 1:N
protection mechanisms are standardized in ITU-T Recommendation G.783. The reserve
circuits can be used for lower-priority traffic, which is simply interrupted if the circuit is
needed to replace a failed working line.

Fig 3 Linear protection

4.2

Ring protection

The greater the communications bandwidth carried by optical fibers, the greater the cost
advantages of ring structures as compared with linear structures. A ring is the simplest
and most cost-effective way of linking a number of network elements. Various protection
mechanisms are available for this type of network architecture, only some of which have
been standardized in ITU-T Recommendation G.841. A basic distinction must be made
between ring structures with unidirectional and bi-directional connections.
4.2.1 Unidirectional rings
Figure 4 shows the basic principle of APS for unidirectional rings. Let us assume that
there is an interruption in the circuit between the network elements A and B. Direction y
is unaffected by this fault. An alternative path must, however, be found for direction x.

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1301

Figure 4: Two fiber unidirectional path switched ring

The connection is therefore switched to the alternative path in network elements A and B.
The other network elements (C and D) switch through the back-up path. This switching
process is referred to as line switched. A simpler method is to use the so-called path
switched ring (see figure 4). Traffic is transmitted simultaneously over both the working
line and the protection line. If there is an interruption, the receiver (in this case A)
switches to the protection line and immediately takes up the connection.
4.2.2

Bi-directional rings

In this network structure, connections between network elements are bi-directional. This
is indicated in figure 5 by the absence of arrows when compared with figure 5. The
overall capacity of the network can be split up for several paths each with one bidirectional working line, while for unidirectional rings, an entire virtual ring is required
for each path. If a fault occurs between neighboring elements A and B, network element
B triggers protection switching and controls network element A by means of the K1 and
K2 bytes in the SOH.

Even greater protection is provided by bi-directional rings with 4 fibers. Each pair of
fibers transports working and protection channels. This results in 1:1 protection, i.e. 100
% redundancy. This improved protection is coupled with relatively high costs.

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Figure 5: Two fiber bi-directional line-switched ring (BLSR)

132

CHAPTER - 8

SDH
INTRODUCTION
Synchronous digital hierarchy (SDH) is an ITU-T universal standard that defines
acommon and reliable architecture for transporting telecommunications services ona
worldwide scale. Synchronous optical network (SONET) is today a subset ofSDH,
promoted by American National Standards Institute (ANSI) and used in theU.S., Canada,
Taiwan, and Korea.From now on we will use the acronym SDH to refer to the generic
ITU-T standardthat includes also SONET.
TH E EM ERGEN CE OF S DH/S ONE T NE TWOR
KS
During the 1980s, progress in optical technologies and microprocessors
offered new challenges to telecommunications in terms of bandwidth and data processing
.At that time, plesiochronous hierarchies (T-carrier and PDH) dominated transport
systems, but a series of limitations and the necessity to introduce new transmission
technologies moved to develop a new architecture .Antitrust legislation was the final
factor that hastened the development of SONET.It was applied to the telecommunications
business and forced the giant, Bell,to be split up into small companies, the regional Bell
operating companies(RBOCs). SONET, developed at Bellcore labs in 1984, grew out of
the need to interconnect RBOCs using standardized optical interfaces. Telecom
liberalization was confirmed around the world during the 90s, and this has inevitably led
to global competition and interoperation. In 1988, the Comit Consultatif
InternationalTlgraphique Et Tlphonique CCITT (now ITU) proposed creating
broadband-ISDN (B-ISDN) to simultaneously transport data, voice, video, and
multimedia over common transmission infrastructures. Asynchronous transfer mode
(ATM) was selected for the switching layer, and SDH for transport at the physical layer.
Li mitation s of Pl e sioc hronou s N etw
orks
Plesiochronous networks have the following limitations:
Their management, supervision, and maintenance capabilities are limited, as there are
no overhead bytes to support these functions. One example of this is that if a resource
fails, there is no standard function whereby the network can be reconfigured.
Access to 64-Kbps digital channels from higher PDH hierarchical signals requires full
demultiplexing, because the use of bit-oriented procedures removes any trace of the
channels.
In PDH it was not possible to create higher bit rates directly; one could do so only after
following all the steps and hierarchies (see Figure 1).
Plesiochronous ANSI and European Telecommunications Standard Institute (ETSI)
hierarchies were not compatible.
There were no standards defined for rates over 45 Mbps in T-carrier, and over 140
Mbps in PDH.
Different manufacturers of plesiochronous equipment could not always be
interconnected, because they implemented additional management channels or
proprietary bit rates.
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Figure 1.1 SDH and SONET allow for direct multiplexing and demultiplexing.
These limitations meant that it was necessary to design a new transmission architecture to
increase the flexibility, functionality, reliability, and interoperability of networks.
The SDH /S ONE T C hall
enge
What had to be decided first was how to provide smooth migration from legacy
installations. Hence a basic frame period of 125 s was selected, the same of E1 and 1
frames, in order to guarantee compatibility with existing services such as plain old
telephone service (POTS), integrated services digital network (ISDN), frame relay (FRL)
or any n x 64 Kbps . Note that a byte constantly carried on a 125-s frame period defines
a 64-Kbps channel.
Some of the remarkable features of SDH compared with its predecessors are:
Synchronous versus plesiochronous Plesiochronous means almost synchronous. This in
its turn means that nodes try to do work in the same frequency, but in fact they do not,
because each PDH island use its own clock. In synchronous networks, all digital
transitions should occur simultaneously, and all the nodes must be fed with the same
master clock . There may, however, be a phase difference between the transitions of the
two signals but this must lie within standardized limits.
Bytes versus bits
In SDH and SONET, such basic operations as multiplexing, mapping, or alignment are
byte oriented, to keep transported elements identified throughout the wholetransmission
path
Direct access
The main difference between SDH and its predecessors is in synchronization and
byte-oriented operations. Synchronization enables us to insert and extract tributaries
directly at any point and at any bit rate, without delay or extra hardware. For this reason,
PDH/T-carrier must completely demultiplex signals of various megabits per second, to
access any embedded channel of n x 64 Kbps.
Full management
In SDH and SONET, payload and overheads are always accessible, and there is no need
to demultiplex the signal. This drastically improves operation, administration, and
maintenance (OA&M) functions, which are essential to enable centralized management
independently of the bit rate. SDH and SONET also provide embedded mechanisms to
protect the network against link or node failures, to monitor network performance, and to
manage network events.

Providing circuits for public networks

The basic function of SDH, just like any transmission network, is that of providing
metropolitan or long-haul transport to networks such as POTS, ISDN, FRL, Gigabit
Ethernet (10GbE), Universal Mobile Telecommunications System (UMTS) or Internet.
Signaling, switching, routing, and billing do not depend on SDH, as it is only in charge of
providing bandwidth between two points.
Universal standard
SDH and SONET standards enable transmission over multiple media, including fiber
optics, radio, satellite, and electrical interfaces. They allow internetworking between
equipment from different manufacturers by means of a set of generic standards and open
interfaces. Scalability is also an important point, as transmission rates of up to 40 Gbps
have been defined, making SDH a suitable technology for high-speed trunk networks.
C OM P ARIS ON OF SDH AN D S
ONE T
SDH and SONET are compatible but not identical. SDH is used worldwide except in the
U.S., Canada, Japan, and partially in South Korea, and Taiwan. Both define a similar set
of structures and functions; however, there are differences in usage.

Table 1.1 Terminology comparison.

The 51.84-Mbps STS-1 is the basic building block of SONET (OC-1 if this signal is
transmitted over fiber optics). STS-1 was enough to transport all T-carrier tribu- taries but
not to carry the 140Mbps PDH tributary. Then SDH chose a basic transport frame three

times faster at 155.53 Mbps, in order to allocate the full European hierarchy. SDH and
SONET terminology have some differences, in referring to the same concepts, bytes, and
structures. Nevertheless, beyond the names, the functionality is equivalent
Internetworking is always possible because the evolution of both technologies has been
the same with the new hierarchies up to 40 Gbps and to the last standards like link
capacity adjustment scheme (LCAS). The objective is to guarantee universal
connectivity.

SDH/SONET Layers
In plesiochronous networks, interactions are simple and direct. In synchronous
networks they are more sophisticated, so responsibilities have been divided among
several layers that communicate with their counterparts by making use of specific
overheads, formats, and protocols. This architecture is equivalent to the layered open
system interconnection (OSI) model to define and design communication networks

Figure 1.2 SDH standards define a layered client/server model that can be divided into
up to four layers in order to manage transmission services.

Path layers
Path layers are the route to transport clients information across the synchronous
network from its source to its destination, where the multiplexers interface with client
equipment At this layer clients information is mapped/ demapped into a frame and path
overhead is added. There are two specialized path layers
1.
Lower-order path (LP), or virtual tributary path (VT Path) in SONET, to transport
lower-rate services. Associated overhead is lower-order path overhead (LO-POH) or
virtual tributary path overhead (VT-POH) in SONET.
2.
Higher-order path (HP), synchronous transport signal path (STS Path) in
SONET, to transport higher-rate services or a combination of lower-rate services.
Associated overhead is higher-order path overhead HO-POH or synchronous transport
signal path overhead (STS-POH) in SONET.
Some of the path layer functions are routing, performance monitoring, anomalies
and defect management, security and protection, as well as specific path OAM functions
support.

Figure1.3 SDH and SONET line topology and network layers.

M ultiple x se ction or line la ye r
Multiplexer section (MS), or line section in SONET, is a route between two
adjacent multiplexers. This layer has several capabilities such as bit error detection, and
circuit protection when an intermediate link or node collapses. It also carries
synchronization reference information and OAM information between nodes. Associated
overhead is multiplex section overhead (MSOH) or line overhead (LOH) in SONET.
Regener ati on se cti on or secti on l a ye r
The regeneration section (RS), (the section layer in SONET) is the link between
two successive NEs. It reads and writes specific overheads and management functions for
each type of transmission media. Its most typical functions are framing, bit error
detection, and regenerator OAM functions support. Associated overhead is regeneration
section overhead (RSOH) or section overhead(SOH) in SONET
Ph ysi cal l a yer s
Fiber optics and metallic cable, together with terrestrial radio and satellite links
can be used as the physical layer (PL). Fiber optics is the most common medium because
of its capacity and reliability. Radio is a cost-effective medium when distance,
geographical difficulties, or low-density areas make the optical alternative less practical.
Nevertheless, radio has some important weaknesses; for example, noise and frequency
allocation, that limit bit rates to 622 Mbps. Electrical cables are also used in some legacy
installations

SDH/SONET FORMATS AND PROCEDURES

SDH defines a set of structures to transport adapted payloads over physical
transmission networks (ITU-T Rec. G.707). Five basic procedures are involved here (see
Figure 1.4):
Mapping: A procedure by which tributaries are adapted into virtual containers at the

boundary of an SDH network.

Stuffing: This is a mapping procedure to adapt the bit rate of client data streams into

standardized, fixed-size containers.

Multiplexing: A procedure by which multiple lower-order signals are adapted into a

higher-order path signal, or when the higher-order path layer signals are adapted into a
multiplex section.
Overhead addition: This procedure is to attach information bytes to a data signal for

internal routing and management.

Aligning: A procedure by which a pointer is incorporated into a tributary unit (TU) or
an administrative unit (AU). TU and AU pointers are used to find units anywhere in the
transmission network.

Figure 1.4 SDH Multiplexing map.

SDH/SONET Frame Structure

The basic transport frame in SONET is synchronous transport signal (STS-1), while in
SDH it is synchronous transmission module (STM-1) STS-1 is a 3 x 9 byte structure
transmitted at 52 Mbps, which is equivalent to STM-0.
STM-1 is a 9 x 9 byte structure transmitted at 155 Mbps, which is equivalent to optical

carrier 3 (OC-3) and electrical STS-3. Both have the same structure that is based on three
types of information blocks:
1. Overhead blocks: These blocks contain information that is used to manage quality,

anomalies, defects, data communication channels, service channels, and so on. There are
two types of overhead blocks, RSOH (managed by the regenerator section layer) and
MSOH (managed by the multiplex section layer).
2. Payload blocks or virtual containers (VCs): They contain a combination of client

signals and overhead blocks. VC does not have a fixed position in the frame, but it floats
in the frame to accommodate clock mismatches.
3. Pointers: They track the VC position, pointing to its first byte, while moving inside the

frame
Containers as transport interfaces
Containers (C-n) are used to map client bit streams. Adaptation procedures have been
defined to suit most telecom transport requirements. These include PDH, metropolitan
area network (MAN), asynchronous transfer mode (ATM), high-level data link control
(HDLC), internet protocol (IP), and Ethernet streams. Placing signals inside a container
requires a stuffing function to match the client stream with the container capacity. The
justification function is necessary for asynchronous mappings, to adapt clock differences
and fluctuations.
Virtual containers or virtual tributaries
Virtual containers (VC-n) or virtual tributaries (VTs) in SONET, support end-to-end path
layer connections; that is, between the point where the client stream is inserted into the
network and the point where it is delivered. Nobody is allowed to modify the VC
contents across the entire path. VCs consist of a C-n payload and a path overhead (POH).
Fields are organized into a block structure that repeats every 125 or 500 s. Containers
hold client data, and the POH provides information to guarantee end-to-end data
integrity.
There are two types of VCs:
The lower-order VC, such as VC-11, VC-12, VC-2, and VC-31. These consist of a

small container (C-11, C-12, C-2, and C-3), plus a 4-byte POH attached to the container
(9 bytes for VC-3).

 The higher-order VC, such as VC-3 or VC-4. These consist of either a big container (C-

3, C-4) or an assembly of tributary unit groups (TUG-2, TUG-3). In both cases, a 9-byte
POH is attached.
Tributary units and tributary unit groups A tributary unit is a structure for adaptation
between the lower-order and higher-order path layer.
A TUG is an SDH signal made up of byte-interleaved multiplexing of one or more TUs.
In other cases, lower-order TUGs are multiplexed to form a higher-order TUG (for
instance, seven multiplexed TUG-2s form one TUG-3), and in other cases, a TUG is
formed by a single TU (for instance, a single TU-3 is enough to form a TUG-3). TUGs
occupy fixed positions in higher-order VCs.
2.4.1.4 Administrative unit
An administrative unit (AU-n) provides adaptation between the higher-order path layer
and the multiplex section layer. It consists of an HO-VC payload and an AU pointer
indicating the payload offset.
Multiplexing Map

A multiplexing map is a road map that shows how to transport and multiplex a number of
services in STM/OC frames The client tributary (PDH, T-carrier, ATM, IP, Ethernet,
etc.) needs to be mapped into a C-n container, and a POH added to form a VC-n, or a VT
for SONET.
The VC/VT is aligned with a pointer to match the transport signal rate. Pointers
together with VCs form TUs or AUs.
A multiplexing process is the next step, whereby TUG-n and AUG-n groups are
created.
When it comes to TUGs, they are multiplexed again to fill up a VC, synchronous
payload envelope (SPE) in SONET, and a new alignment operation is performed.
Finally, an administrative unit group (AUG) is placed into the STM/OC transport frame.
NE TWOR K E LE M EN TS AN D TOP
OLOGY
Network Elements

SDH systems make use of a limited number of network elements (NEs) within which all
the installations are fitted
Regenerators (REGs) or section terminating equipments (STEs): Every signal sent
through any transmission medium (optical, electrical or radio-electrical) experiences
attenuation, distortion, and noise. Regenerators supervise the re ceived data and restore
the signals physical characteristics, including shape and synchronization. They also
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manage the monitoring and maintenance functions of the regenerator section (RS) or
section in

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 Line terminal multiplexers (LTMUX) or path terminal equipment (PTE): they are
common in line and access topologies. Their function is to insert and extract data in
synchronous frames .

Add and drop multiplexers (ADMs) can insert or extract data directly into or from the

traffic that is passing across them, without demultiplexing/multiplexing the frame. Direct
access to the contents of the frame is a key feature of SDH, as it enables us to turn any
point of the network into a service node, just by installing an ADM.

Digital cross-connects (DXCs) configure semipermanent connections to switch traffic

between separate networks. The switched traffic can be either SDH streams or selected
tributaries. Although it is not common, DXCs can also insert and drop tributaries in
transport frames.

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Network Topology

Synchronous multiplexers provide great flexibility for building topologies, which is why
point-to-point, linear, ring, hub, meshed, and mixed topologies are all possible:
Linear point-to-multipoint: this topology follows the basic point-to-point structure, but
now includes ADM multiplexers performing add and drop functions at intermediate
points.

Ring: this topology closes itself to cover a specific area, with ADM multiplexers
installed at any point. It is flexible and scalable, which makes it very suit able for wide
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area and metropolitan networks. Rings are frequently used to build fault-tolerant
architectures.

Hub or star: this topology concentrates traffic at a central point, to make topology
changes easier. A hub can join several networks with different topologies.

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144

CHAPTER - 9

DENSE WAVELENGTH DIVISION MULTIPLEXING

(DWDM)
1.0

INTRODUCTION

The revolution in high bandwidth applications and the explosive growth of the Internet,
however, have created capacity demands that exceed traditional TDM limits. To meet
growing demands for bandwidth, a technology called Dense Wavelength Division
Multiplexing (DWDM) has been developed that multiplies the capacity of a single fiber.
DWDM systems being deployed today can increase a single fibers capacity sixteen fold,
to a throughput of 40 Gb/s. The emergence of DWDM is one of the most recent and
important phenomena in the development of fiber optic transmission technology. Dense
wavelength-division multiplexing (DWDM) revolutionized transmission technology by
increasing the capacity signal of embedded fiber. One of the major issues in the
networking industry today is tremendous demand for more and more bandwidth. Before
the introduction of optical networks, the reduced availability of fibers became a big
problem for the network providers. However, with the development of optical networks
and the use of Dense Wavelength Division Multiplexing (DWDM) technology, a new
and probably, a very crucial milestone is being reached in network evolution. The
existing SONET/SDH network architecture is best suited for voice traffic rather than
todays high-speed data traffic. To upgrade the system to handle this kind of traffic is
very expensive and hence the need for the development of an intelligent all-optical
network. Such a network will bring intelligence and scalability to the optical domain by
combining the intelligence and functional capability of SONET/SDH, the tremendous
bandwidth of DWDM and innovative networking software to spawn a variety of optical
transport, switching and management related products.
In traditional optical fiber networks, information is transmitted through optical fiber by a
single light beam. In a wavelength division multiplexing (WDM) network, the vast
optical bandwidth of a fiber (approximately 30 THz corresponding to the low-loss region
in a single-mode optical fiber) is carved up into wavelength channels, each of which
carries a data stream individually. The multiple channels of information (each having a
different carrier wavelength) are transmitted simultaneously over a single fiber. The
reason why this can be done is that optical beams with different wavelengths propagate
without interfering with one another. When the number of wavelength channels is above
20 in a WDM system, it is generally referred to as Dense WDM or DWDM.
DWDM technology can be applied to different areas in the telecommunication networks,
which includes the backbone networks, the residential access networks, and also the
Local Area Networks (LANs). Among these three areas, developments in the DWDMbased backbone network are leading the way, followed by the DWDM-based LANs. The
development on DWDM-based residential access networks seems to be lagging behind at
the current time.

[Type text]

2.0

DEVELOPMENT OF DWDM TECHNOLOGY

Early WDM began in the late 1980s using the two widely spaced wavelengths in the
1310 nm and 1550 nm (or 850 nm and 1310 nm) regions, sometimes called wideband
WDM.
The early 1990s saw a second generation of WDM, sometimes called
narrowband WDM, in which two to eight channels were used. These channels were now
spaced at an interval of about 400 GHz in the 1550-nm window. By the mid-1990s, dense
WDM (DWDM) systems were emerging with 16 to 40 channels and spacing from 100 to
200 GHz. By the late 1990s DWDM systems had evolved to the point where they were
capable of 64 to 160 parallel channels, densely packed at 50 or even 25 GHz intervals.
As fig. 1 shows, the progression of the technology can be seen as an increase in the
number of wavelengths accompanied by a decrease in the spacing of the wavelengths.
Along with increased density of wavelengths, systems also advanced in their flexibility of
configuration, through add-drop functions, and management capabilities.

Figure 1 Evolution of DWDM

3.0

VARIETIES of WDM

Early WDM systems transported two or four wavelengths that were widely spaced.
WDM and the follow-on technologies of CWDM and DWDM have evolved well
beyond this early limitation.
3.1

WDM

Traditional, passive WDM systems are wide-spread with 2, 4, 8, 12, and 16 channel
counts being the normal deployments. This technique usually has a distance limitation of
less than 100 km.
3.2

CWDM

Today, coarse WDM (CWDM) typically uses 20-nm spacing (3000 GHz) of up to 18
channels. The CWDM Recommendation ITU-T G.694.2 provides a grid of wavelengths
for target distances up to about 50 km on single mode fibers as specified in ITU-T
[Type text]

Recommendations G.652, G.653 and G.655. The CWDM grid is made up of 18

wavelengths defined within the range 1270 nm to 1610 nm spaced by 20 nm.

3.3

DWDM

Dense WDM common spacing may be 200, 100, 50, or 25 GHz with channel count
reaching up to 128 or more channels at distances of several thousand kilometers with
amplification and regeneration along such a route.
4.0

DWDM System Function

DWDM stands for Dense Wavelength Division Multiplexing, an optical technology used
to increase Band width over existing fiber optic backbones. Dense wavelength division
multiplexing systems allow many discrete transports channels by combining and
transmitting multiple signals simultaneously at different wavelengths on the same fiber.
In effect, one fiber is transformed into multiple virtual fibers. So, if you were to multiplex
32 STM-16 signals into one fiber, you would increase the carrying capacity of that fiber
from 2.5 Gb/s to 80 Gb/s. Currently, because of DWDM, single fibers have been able to
transmit data at speeds up to 400Gb/s.
A key advantage to DWDM is that it's protocol and bit rate-independent. DWDM-based
networks can transmit data in SDH, IP, ATM and Ethernet etc. Therefore, DWDM-based
networks can carry different types of traffic at different speeds over an optical channel.
DWDM is a core technology in an optical transport network. Dense WDM common
spacing may be 200, 100, 50, or 25 GHz with channel count reaching up to 128 or more
channels at distances of several thousand kilometers with amplification and regeneration
along such a route.

Fig. 2 Block Diagram of a DWDM System

The concepts of optical fiber transmission, loss control, packet switching, network
topology and synchronization play a major role in deciding the throughput of the
network.
[Type text]

4.0

TRANSMISSION WINDOWS

Today, usually the second transmission window (around 1300 nm) and the third
and fourth transmission windows from 1530 to 1565 nm (also called conventional band)
and from 1565 to 1620 nm (also called Long Band) are used. Technological reasons limit
DWDM applications at the moment to the third and fourth window.
The losses caused by the physical effects on the signal due by the type of
materials used to produce fibres limit the usable wavelengths to between 1280 nm and
1650 nm. Within this usable range the techniques used to produce the fibres can cause
particular wavelengths to have more loss so we avoid the use of these wavelengths as
well.
5.0

DWDM SYSTEM COMPONENTS

Figure 3 shows an optical network using DWDM techniques that consists of five main
components:
1. Transmitter (transmit transponder):
- Changes electrical bits to optical pulses
- Is frequency specific
- Uses a narrowband laser to generate the optical pulse
2. Multiplexer/ demultiplexer:
- Combines/separates discrete wavelengths
3. Amplifier:
- Pre-amplifier boosts signal pulses at the receive side
- Post-amplifier boosts signal pulses at the transmit side (post amplifier) and on the
receive side (preamplifier)
- In line amplifiers (ILA) are placed at different distances from the source to provide
recovery of the signal before it is degraded by loss.
- EDFA (Eribium Doped Fiber Amplifier) is the most popular amplifier.
4. Optical fiber (media):
- Transmission media to carry optical pulses
- Many different kinds of fiber are used
5. Receiver (receive transponder)
- Changes optical pulses back to electrical bits
- Uses wideband laser to provide the optical pulse

[Type text]

Figure 3: DWDM System Components

5.1

6.0

BENEFITS of DWDM

Increases bandwidth (speed and distance)

Does not require replacement or upgrade their existing legacy systems

Provides "next generation" technologies to meet growing data needs

Less costly in the long run because increased fiber capacity is automatically
available; don't have to upgrade all the time.
Fibers Supporting DWDM

For transmitting the DWDM signal, the conventional single mode optical fibers i.e. ITU
G 652 compliant, are not completely suitable. Due to availability of Optical Amplifier
working in 1550 nm region, the operating wavelengths are chosen in the C band i.e. from
1530 to 1565 nm. The ITU G 652 fiber has very high dispersion in 1550 nm region,
which limits the distance between repeater stations severely.
ITU G 652 fiber with the high dispersion at 1550 nm, typically 18 ps/nm-km. Although,
it is possible to compensate the dispersion by using dispersion compensating fibers
(DCF), these DCF adds to additional optical loss.
Conversely, in case of ITU G 653 fibers with zero dispersion at 1550 nm, the
nonlinearities such as Four Wave Mixing (FWM) plays dominant role, rendering the fiber
unsuitable for long distance transmission. A fiber that has small but non-zero amount of
dispersion can minimize the non-linearity effects. The ITU G 655 compliant, Non Zero
Dispersion Fibers (NZDF) has dispersion which is carefully chosen to be small enough to
enable high speed transmission over long distances, but large enough to suppress FWM.
With the proper use and placement of Optical Amplifiers, it is possible to have the
repeater less link.
In summary, a future-proof fiber optic network should have combination of ITU G 652
and G 655 fibers. The number of each type of fiber in a cable is generally chosen based
on the type of network. The complete fiber optic network can be defined in broader
manner in two parts:
(a) High capacity, long haul Backbone network or Transport network and (b) High
Speed, Local Access network. The Backbone network connects the major cities of the
networks and carries high bit rate signals so the G 655 fibers should be deployed in
backbone. The Local Access network is used for carrying the data up to the customer
[Type text]

premises. Due to smaller spacing between the stations / customer premises equipment,
the standard single mode fibers (ITU G 652) can be deployed.
7.0

OPTICAL NE TYPES

Optical Multiplexer/Demultiplexer
Multiplexing and Demultiplexing of different wavelength signals.
(b)

Optical Amplifiers

Pure optical 1R regeneration (just amplification) of all transmitted signals.

(c)

Transponders

Wavelength change and 2R regeneration (reshaping and amplification) or 3 R

regeneration (reshaping retiming and amplification).
(d)

Regenerators

Real 3 R regeneration (reshaping, retiming and amplification) of the

signal. Therefore, the signals have to be demultiplexed, electrically regenerated
and multiplexed again. They are necessary if the length to be bridged is too long
to be covered only by optical amplifiers, as these only perform reshaping and
retiming.
(e)

Optical Add/Drop Multiplexer

Adding and Dropping only specific wavelengths from the joint optical signal.
This may use complete de-multiplexing or other techniques.

(f)

Optical cross-connects
To cater for the huge amount of data expected in an optical network even
the cross-connects have to work on a purely optical level.

8.0 DWDM Backbone Networks

We can divide the network structures of DWDM-based backbone networks into three
classes:
Simple point-point DWDM link,
DWDM wavelength routing with electronic TDM (time domain multiplexing) and
switching/routing backbone network, and
All-optical DWDM network.
8.1

Point-To-Point DWDM links

The simplest application of the DWDM technology in backbone networks is the point-topoint link. Figure 4 shows the architecture of the networks using 4 network
switching/routing nodes as an example. In this architecture, the electronic nodes can be
SONET/SDH switches, Internet routers, ATM switches, or any other type network nodes.
The DWDM node consists of typically a pair of wavelength multiplexer / demultiplexer
(lightwave grating devices) and a pair of optical-electrical/ electrical-optical convertors.
Each wavelength channel is used to transmit one stream of data individually. The
DWDM wavelength multiplexer combines all of the lightwave channels into one light
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beam and pumps it into one single fiber. The combined light of multiple wavelengths is
separated by the demultiplexer at the receiving end. The signals carried by each
wavelength channel are then converted back to the electrical domain through the O/E
convertors (photodetectors). In this way, one wavelength channel can be equivalent to a
traditional fiber in which one lightbeam is used to carry information. It is worth noting
that the wavelength channels in one fiber can be used for both directions or two fibers are
used with each for one direction.
The advantage of the point-to-point DWDM links is that it increases the bandwidth by
creating multiple channels with low costs. The limitation of this approach, however, is
that the bandwidth of each wavelength channel may not be fully utilized due to the speed
of the electrical devices, which is referred to as the well-known electro-optic bottleneck.
Also, the use of the wavelength channels may not be optimal due to the fact that the
meshes formed by the wavelength channel are all identical, which can be seen in Figure
4.
Electronic
TDM Node

DWDM
Node

DWDM
DWDM
Node
Node

DWDM
Node

The DWDM Nodes consist

typically wavelength
and O/E converters

Electronic
TDM Node

DWDM
Node

DWDM Channels
DWDM
DWDM
Node

Electronic
TDM Node

The Electronic TDM nodes

can be SDH Switches, TDM
Switches or Internet routers
DWDM
DWDM

DWDM

Node
Node

Node
DWDM

DWDM
Node
Electronic
TDM Node

Node

Figure 4: DWDM point-to-point link backbone network

8.2

Wavelength routing with electronic TDM

Figure 5 depicts the second type of DWDM application in backbone networks, in which
wavelength routers are used to configure or reconfigure the network topology within the
optical domain and the TDM (Time Domain Multiplexing) network nodes are used to
perform multiplexing and switching in the electrical domain. This combined optical and
electrical network architecture can be applied in SONET/SDH in which the electrical
TDM network nodes would be SONET switches, or in the Internet in which the electrical
TDM network nodes would be the Internet routers. The architecture can also be used in
an ATM network where the electrical TDM network nodes would be ATM switches.
The advantage of this combined architecture in comparison with the simple DWDM
point-to-point links is that it can optimize the use of DWDM wavelength channels by

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reconfiguring the mesh formed by the wavelength channels. The topology of

reconfiguration can be dynamic in which network topology is reset periodically
according to the traffic with the time period in the order of seconds or milliseconds. The
reconfiguration can also be static in which the mesh is set for a longer period of time. The
enabling technology for this architecture is the wavelength router in the optical domain.
Different types of wavelength routers are available commercially which range from
mechanically controlled to thermally controlled and to semiconductor wavelength
switches. The advantage of this architecture is its ability to utilize the bandwidth capacity
to the level that the electronics can handle, because of the reconfiguration of the mesh by
the wavelength routers. The problem, however, is still the electro-optic bottleneck.
Nevertheless, it has improved in comparison with the point-to-point DWDM links.
Several technical issues are required to be dealt with for this architecture, which include
the control system for the mesh reconfiguration, the traffic evaluation among the DWDM
channels and among the fibers, as well as the technology for constructing the wavelength
routers. These technical issues have been the topics of research for some time and all are
still undergoing development at the current time.

h
tg r n
t
leu
eo
av
R
W

Wavelength
Router

Optical Domain
Wavelength
Router

Wavelength
Router

Electronic
TDM Node

Wavelength
Router

Electronic
TDM Node

W
R va
oe
u
l
e
r gehtnt

Electronic
TDM Node

Electronic
TDM Node

Figure 5: Wavelength routing with electronic TDM DWDM networks

8.3

All-Optical DWDM Networks

The goal of all-optical DWDM networks is to eliminate the conversions between

electricity and light. The all-optical network is also referred to as the transparent network.
Two types of all-optical DWDM backbone networks have been proposed, which are:
Wavelength switching DWDM networks without TDM, and
DWDM with optical domain TDM.
These two types of DWDM backbone networks are discussed in the following two
subsections.

8.3.1

Wavelength switching without TDM

Circuit switching can be achieved by using wavelength switches (also called wavelength
routers). Figure 6 shows the architecture of the network, in which wavelength switches
are used to establish connections between the two communicators. This is quite the same
as the old PSTN (Public Switched Telephone Network) system where the crossbar types
of electrical switches are used to establish the circuit for the two users. The wavelength
switches for this type of networks can switch among the wavelength channels of multiple
fiber input and output ports. A wavelength router may have the additional capacity of
changing the wavelength of the signal between routers resulting in high utilization of
wavelength channels. In the case of a wavelength router without wavelength conversion,
the two users involved in the communication are connected by one signal wavelength
across all of switches in the light path. However, with the wavelength conversions, the
two sides of the communication can be connected by different wavelengths in different
fiber links between switches.
To Users

Users

h
tg r n
t
e
lu
e oa v
R
W

Optical Domain

W
R va
o eW
R
u
va
e
o
elhtnt
r ge
u
l
e
r gehtnt

Wavelength
Router

To

Wavelength
Router

Wavelength
Router

To
Users

Wavelength
Router

To Users

Figure 6: Wavelength switching without TDM

Figure 7 shows an example of a wavelength router structure with multiple fiber inputs
and outputs.

Demultiplexer
w1
w2

W1, W2, W3, W4

Multiplexer
Wavelength
Switch
Switch

w1
w2

w3
w4

Incoming
Fibers

W1, W2, W3, W4

w3
w4
Wavelength
Switch

Outgoing
Fibers
Wavelength
Switch

Wavelength
Switch

Figure 7: A wavelength router

The advantage of this type of DWDM network is its simplicity in the switching
mechanism. The major problem for this network is the under utilization of the bandwidth
capacity of the wavelength channels. This is because once the light path is established
between the two sides of the communication parties; it is up to the user to use the
available bandwidth of the wavelength channel which can be tens of GHz or even
hundreds of GHz. It is obviously inefficient for voice communications.
Therefore, this type of all-optical DWDM backbone networks will unlikely be used to
replace the SONET/SDH. However, it can be useful for Enterprise intranets where
different types of communications can share the same network with low costs of building
it. Also, because of the transparency of the network, it provides many advantages, such
low error rate and low maintenance costs.
8.3.2

Wavelength switching with optical TDM

As it has been mentioned above, the wavelength routing all-optical network has the
problem of low efficiency in utilizing the bandwidth of wavelength channels with each
having the capacity of hundreds of Gigabits per second. Although the combined
wavelength routing and electronic time domain multiplexing can increase the bandwidth
utilization to some degree, it introduces the O/E conversions that may restrict the speed
and cause packet delays. Therefore, it is natural to implement optical TDM in future
optical networks, which eliminates the O/E conversions resulting in a transparent highspeed all-optical network. Replacing the electrical TDM nodes in the DWDM with
electrical TDM architecture (Figure 5) by optical TDM nodes, we obtain a DWDM
wavelength routing with optical TDM architecture, as shown in Figure 8. As we have
seen that the electrical TDM/switching nodes in Figure 5 can be of any kind, such as
SONET/SDH switches, Internet routers, and ATM switches. This indicates that the alloptical TDM nodes in the all-optical architecture can be optical SONET/SDH switches,
or all-optical ATM switches, or all-optical Internet routers. Different types of all-optical

TDM/switch nodes can also be in one network, provided the protocol conversions are
implemented.
To Users

Optical
TDM
Node

Users

Optical
TDM
Node

h
tg r n
t
e
lu
e oa v
R
W

Optical
Domain

W
a
R vW
oe
R va
o
ru
e gel nt
u gelh
tnt
re
t
h

Wavelength
Router

To

Wavelength
Router

Wavelength
Router

To

Optical
TDM
Node

Users

Wavelength
Router

Optical
TDM
Node

To Users

Figure 8: All-optical TDM/switch with wavelength router

In fact, the optical TDM/switch node and the wavelength router in one routing site
(Figure 8) can be combined into one all-optical switching node that not only forwards
packets through time domain multiplexing but also selects the light path intelligently
according to the availability and traffic loads of the links. This architecture is shown in
Figure 9.
To Users

Optical
TDM
Switch

Optical
TDM
Switch

Users

l
ich ci
ta
tM
Optical Domain
pDT w
S
O

Optical
TDM
Switch

To Users

O
S
w
D tT p i
i cl
t c a
M
h

Optical
TDM
Switch

To

To
Users

Figure 9: All-optical TDM/switch with self wavelength selection

The only question left so far is how can we build such all-optical TDM/ switch nodes.
This is the problem that has not been solved. Although some research and proposals have
been published, such as using a hybrid mechanism where electrical signals are used to
perform the control and light signals are used to carry the data, the system is far from
mature. Also it still involves electricity which still has many problems to be dealt with.
Looking into the possibilities of building the three major types of the current electrical
switches, namely the SONET/SDH switch, the ATM switch, and the Internet router, the
SONET/SDH switch is the simplest one among the three since it is for circuit switching.
The Internet router is the most complex one among the three, since it requires a digital
optical processor and all-optical memories. In any case, digital optical logic devices are
required to build these switches. Therefore, it seems clear that we need to have optical
digital processing power before we can implement any of these TDM switches, unless
other mechanism emerges to revolutionize the existing concept of packet switching
networks.

9.0

DWDM Access Optical Networks

An overriding belief existed even in the early 1970's that optical fiber would one day
make its way into the subscriber loop and be used to connect individual homes. Research
on the fiber based residential access network architecture and protocols have since then
become one of the major areas in the telecommunication arena. The ATM (Asynchronous
Transfer Mode) based B-ISDN (Broadband Integrated Services Digital Network)
architecture had been once believed to be the leading candidate for realizing the fiber-tothe-home access network. However, with the technological development of the DWDM,
broadband residential access fiber network has taken another turn, which leads to a
DWDM-based fiber optical network to deliver both narrowband and broadband services.
This section provides an overview of the network architectures that have been developed
for the residential access networks based on the DWDM technology. DWDM-based
access optical networks can be classified into two categories, passive DWDM access
networks and active DWDM networks. The term of active DWDM network here refers as
to the DWDM network in which the TDM (time domain multiplexing) is applied in the
wavelength channels. These two types of access network architecture are discussed in the
following subsections.
9.1

DWDM Passive Optical Networks (PON)

DWDM passive optical networks (PON) use the wavelength channels to connect the
users with the central office. Each service uses one wavelength channel. The early PON
was developed for narrowband services, such as the PON architecture developed by
British Telecom. However, recent PONs are for both broadband and narrowband
services. A passive subscriber loop is attractive because it uses no active devices outside
the central office (CO), except at the customer premises. Several architectures of passive
optical networks have been proposed for WDM or DWDM, which include the single-star,
the tree, the doublestar, and the star-bus. Figure 10 shows the single-star architecture in
which each household has a dedicated fiber to the central office (CO).

Figure 10: Single-star PON architecture

The WDM channels in the fiber are used to carry all required services, such as voice and
video. This architecture is designed for easy installation and upgrading; however, the cost
of dedicated optical fiber between the customer and the CO in this network is still a major
concern. Thus, this architecture may not be suitable for widespread deployment in the
near term. Figure 11 shows the tree PON architecture, in which the DWDM channels are
split in the way of tree branches with each user having one or more wavelength channels.
This architecture reduces the fiber use in comparison with the single-star. It is a better
architecture, especially for DWDM-based system in which a large number of wavelength
channels are available. This architecture can satisfy the customer needs for both
narrowband and broadband services. One drawback of this network architecture is its
rigidity, in terms of network upgrading. The star-bus architecture can be considered as a
variation of the tree architecture, which improves the flexibility of the tree architecture.
DWDM
DWDM

CO
CO

W1...Wn
W 1...W n

W1...Wn
W1...Wn

Splitter
Splitter

W1...Wn
W 1...W n

Splitter
Splitter

CP
CP

Figure 11: Tree PON architecture

Splitter
Splitter

Figure 12 depicts the double-star PON architecture. This architecture provides more
flexibility in comparison with the star-bus architecture. It can be considered as the frontrunner among the possible architectures of PON for residential access applications.
DWDM
DWDM

CO
CO

W1...Wn
W1...Wn

W1...Wn
W1...W n

DWDM
DWDM

W1...Wn
W 1...W n

DWDM
DWDM

DWDM
DWDM

CP
CP

Figure12: The double-star PON architecture

9.2

DWDM Active Access Optical Networks

In the passive DWDM access networks, each wavelength channel is used to provide one
service at a given time regardless of the channel capacity and bandwidth requirement of
the service. With the increasing bandwidth capacity of DWDM technology, the
bandwidth of one signal channel becomes high enough to carry several or many services
even in the access environment. This leads to the thinking of applying TDM in each
individual DWDM wavelength channel, resulting in the active DWDM access optical
network in which TDM is used within each channel to provide integrated services. The
Asynchronous Transfer Mode (ATM) has been proposed as the TDM protocol in the
active DWDM access networks. With the ATM coming into the picture, the original BISDN (Broadband Integrated Services Digital Network) protocols are again surfacing in
the access network arena. But this time, only one wavelength channel replaces the whole
optical fiber in the system. The network topologies for the passive DWDM access
network discussed in the previous subsection can also be used for the active DWDM
access network. Although an active DWDM access network provides high utilization of
the wavelength channels and in return reduces the fiber costs, it adds additional costs
because of the ATM devices in the system from CO to user premises. It also increases the
complexity of system management and maintenance, which leads to high operating costs.
Another twist in this hard-to-decide matter is the birth of the very high channel-count
DWDM, in which thousands of wavelength channels are created and transmitted with one
fiber. This may make the active DWDM access network architecture lose its potential
advantages, and make the passive high channel-count DWDM PON become the leader in
the race of access network architectures.

10.

Conclusion

The DWDM point-to-point technology has already played an important role in the
backbone networks and it will continue to be installed for existing and new fiber links.
However, for the all-optical DWDM network to become viable, we may have to wait till
the optical processing power becomes available. This may create a time gap between the
DWDM point-to-point applications and the all-optical DWDM transparent networks. In
the access network case, it is still not clear whether the passive or the active is the leader.
Although the cost barrier has been weakened through replacing the fiber by DWDM
channels in the active access network architecture, the TDM devices in the system may
still be too high at the present time. On the other hand, the fiber cost (along with the costs
of the passive devices) in the passive architecture is probably still not cheap enough to
make it ahead of the active architecture. However, the very high channel-count DWDM
may change the landscape of the access network world, and it may become even cheaper
than the combined costs of twisted pair copper and coaxial cables.
Assignment
Qu.1

What is DWDM?

Qu.2

Write down the benefits of DWDM?

Qu.3

Write down the network Elements of DWDM?

Qu.4

Explain the Point to Point DWDM links?

Qu.5

Explain the working of All Optical DWDM Networks?

Qu.6

What is DWDM passive optical Network?

CHAPTER -10

MOBILE COMMUNICATION
1. Principle of Mobile Communication
Introduction
In Telecom network conventionally each user is connected to the Telephone exchange
individually. This dedicated pair starts from MDF, where it is connected to the
appropriate Equipment point and ends at the customer premises Telephone. (With
flexibility at cabinet/pillar/ distribution points DPs)

The connectivity from exchange to customer premises is called Access Network or

Local Loop, and mostly comprises of underground cable from exchange up to DPs and
insulated copper wires (Drop Wires)later on This type of Access Network does not
require separate Authentication of customer before extending services. Whenever the
cable capacity has reached the maximum additional cable is laid to augment the capacity.
Even though there are advantages in introducing wireless connectivity in Subscribers
loop, we have to tackle certain issues viz,
1. Duplexing methodology.
2. Multiple Access methods.
3. Cellular principle or reuse concept.
4. Techniques to cope with mobile environment.
Duplexing Methodology:
Duplexing is the technique by which the send and receive paths are separated over the
medium, since transmission entities (modulator, amplifiers, demodulators) are involved.
There are two types of duplexing:
Frequency Division Duplexing FDD
Time Division Duplexing TDD
Frequency Division Duplexing FDD
Different Frequencies are used for send and receive paths and hence there will be a
forward band and reverse band. Duplexer is needed if simultaneous transmission (send)
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and reception (receive) methodology is adopted .Frequency separation between forward

band and reverse band is constant
Time Division Duplexing (TDD)
TDD uses different time slots for transmission and reception paths. Single radio
frequency can be used in both the directions instead of two as in FDD. No duplexer is
required. Only a fast switching synthesizer, RF filter path and fast antenna switch are
needed. It increases the battery life of mobile phones.
GSM and CDMA systems use Frequency Division Duplexing and corDECT uses Time
Division Duplexing.
Multiple Access methodology:
The technique of dynamically sharing the finite limited radio spectrum by multiple users
is called Multiple Access Technique. By adopting multiple access techniques all users
can not get the services simultaneously and some amount of blocking is introduced by the
system. This is known as GOS (Grade of Service).
Generally there are three different types of multiple access technologies. They are
Frequency Division Multiple Access (FDMA)
Time Division Multiple Access (TDMA)
Code Division multiple Access (CDMA)
Frequency Division Multiple Access (FDMA):
FDMA is a familiar method of allocating bandwidth, where a base station is allowed to
transmit on one or more number of preassigned carrier frequencies and a mobile unit
transmits on corresponding reverse channels. No other base station within range of the
mobile will be transmitting on the same forward channel, and no other mobile within
range of the base station should be transmitting on the same reverse channel. Both the
base and the mobile usually transmit continuously during a conversation, and fully
occupy their assigned forward and reverse channels. No other conversation can take place
on these channels until the first conversation is completed.

t
FDMA Analogy
It may be easier to visualize FDMA by imagining a cocktail party where two people wish
to converse with each other. Then everyone in the room must be silent except for the
speaker. The speaker may talk as long as they wish, and when they finish someone else
may start speaking, but again only one at a time. New speakers must wait (or find another
party) for the current speaker to finish before starting. Everyone in the room can hear and
understand the speaker, unless they are too far away or the speaker's voice is too soft. If
the intended listener is close enough, the speaker may decide to whisper. Conversely, if
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the listener is too far away, the speaker may have to shout. Since no one else should be
talking, this presents no problem. If someone talks out of turn, the listener will probably
be confused and not be able to understand either speaker.
Features Of Frequency Division Multiple Access (FDMA)
No Precise coordination in time domain is necessary in FDMA System.
It is well suited for narrow band analog systems.
Guard spacing between channels causes wastage of frequency resource.
Otherwise good modulation techniques are to be employed to avoid such
guard spacing.
The transmission is simultaneous and continuous and hence duplexers are
needed. Continuous transmission leads to shortening of battery life.
Time Division Multiple Access (TDMA)
TDMA is a more efficient, but more complicated way of using FDMA channels.
In a TDMA system each channel is split up into time segments, and a transmitter is given
exclusive use of one or more channels only during a particular time period. A
conversation, then, takes place during the time slots to which each transmitter (base and
mobile) is assigned. TDMA requires a master time reference to synchronize all
transmitters and receivers.

TDMA Analogy
In TDMA, everyone in the room agrees to watch a clock on the wall, and speak
only during a particular time. Each person wishing to talk is given a set period of time,
and each person listening must know what that time period will be. For example,
everyone may agree on time slots with duration of ten seconds. Speaker number one may
talk for ten seconds starting from the top of the minute. The listener who wishes to hear
this speaker must also be made aware of the schedule, and be ready to listen at the top of
the minute. Speaker number two may speak only from ten seconds after the minute until
twenty seconds after. As with FDMA, only one person at a time may speak, but each
speaker's time is now limited and many persons may take their turn. If someone in the
room cannot see the clock, they will not be able to speak and will have great difficulty
understanding the speakers.
Features of TDMA
There can be only one carrier in the medium at any time, if a simple TDMA
scheme is followed.
Transmission is in bursts and hence is well suited for digital communication.
Since the transmission is in bursts, Battery life is extended.
Transmission rate is very high compared to analog FDMA systems.
Precise synchronization is necessary.
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Guard time between slots is also necessary

Time and Frequency Division Multiple Access
Both methods of FDMA and TDMA are combined to achieve higher capacity in
practical systems. A channel gets a certain frequency band for a certain amount of time.
The Best example for such system is GSM.

Cellular Concepts:
Even though multiple access techniques allowed multiple users to share the medium
simultaneously, due to constraints in providing resources, an amount of blocking will
exist. The amount of blocking is called Grade Of Services(GOS). Based on GOS and
resource availability (no. of carriers/no. of timeslots/both) the traffic handling capacity of
the system is calculated. If this total traffic is divided by traffic per subscriber, we get
number of subscribers supported by the system. For these purposes Erlang B table
(Blocking calls cleared) is useful particularly in FDMA-TDMA.
Why Cellular?
Assuming 30mE traffic per subscriber, sub density of 30 per sq.km, and GOS 1%
Radius

Area (KM2)

Subs

Total Traffic

RF Channels

3.14

100

3.0E

28.03

900

27E

38

10

3.14

10000 300E

360

Providing 360 RF channels for 10,000 subscribers in an area of 314 sq.km on a single
base station is not feasible and if still either the area of coverage or sub density increases,
the system cannot function at all for want of bandwidth.
Hence the solution is dividing the service area into small units, called cell, with base
stations radiating with low power, and limited number of carriers required as per traffic.
The same carriers are again reused at a different cell, which is geographically separated.
(Frequency Reuse)
In case of CDMA it appears that there is no limitation for simultaneous calls but
practically there is a limit to CDMA capacity. And it is essentially the amount of
interference a CDMA receiver can tolerate. As more and more units transmit, the amount
of noise a receiver sees goes up, since all signals not using the receiver's specific PN code
appear as noise. At some point there is so much noise that the receiver can no longer hear
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the transmitter. Boosting the transmitter power won't help overall, since it increases the
noise for all the other receivers, who would in turn tell their transmitters to boost power,
and the situation remains. In a nutshell, if a unit near a base station is transmitting with
too much power, signals from units far from the base station will be lost in the noise.
Hence cellular concept is applicable even in the case of CDMA where code used
for identification of cell/sector is reused.
Advantages of Cellular Principle
Base stations can transmit at low power compared to a single high power transmitter.
It requires less RF bandwidth to cover a given area. Frequency reuse gives good spectrum
efficiency. (FDMA-TDMA)
Disadvantage of cellular principle
Reuse introduces interference.
Established calls should be handed over to next cell to avoid dropping of calls when the
customer is in mobility.
Mobile Environment
BTS is connected to Mobile or Fixed Wireless Terminal by air Interface. This
connectivity differs from our earlier UHF/Microwave which is purely Line of Sight
(LOS) system. In mobile communication due to the mobility of the user from the BTS
LOS to BTS may exist or may not exist. The radio wave is subject to attenuation,
reflection, Doppler shift and interference from other transmitter. These effects cause loss
of signal strength and distortion which will impact the quality of voice or data. To cope
with the harsh conditions, any mobile technology makes use of an efficient and protective
signal processing. Proper cellular design must ensure that sufficient radio coverage is
provided in the area.
Types of signal strength variations
The signal strength variation for mobile is due to different types of signal strength fading.
There are two types of signal strength variations
Macroscopic Variations Due to the terrain contour between BTS and MS. The
fading effect is caused by shadowing and diffraction (bending) of radio waves.
Microscopic variations. Due to multipath, Short-term or Rayleigh fading. As the
MS moves, radio waves from many different paths will be received.
Macroscopic Variations
Macroscopic Variations can be modeled as the addition of two components that
make up the path loss between mobile and base station. The first component is the
deterministic component (L) that adds loss to the signal strength as the distance(R)
increases between base and mobile. This component can be written as
L=1/Rn
Where n = typically 4.
The other macroscopic component is a Log normal random variable which takes into
account the effects of shadow fading caused by variations in terrain and other
obstructions in the radio path.
Local mean value of path loss=deterministic component +log normal random variable
Microscopic Variations
Microscopic Variations or Rayleigh Fading occur as the mobile moves over short
distances compared to the distance between mobile and base. These short term variations
are caused by signal scattering in the vicinity of the mobile unit e.g. by hill, building or
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traffic. The result is that not one but many different paths are followed between
transmitter and receiver (Multipath Propagation). The reflected wave will be altered in
both phase and amplitude. The signal may effectively disappear if the reflected wave is
180 degree out of phase with the direct path signal. The partial out of phase relationships
among multiple received signal produce smaller reduction in received signal strength.

Special features of mobile technologies

All mobile techniques incorporate some special features to overcome the hazards
created by mobile environment. The following are a few to name:
1. Coding.
2. Diversity techniques.
3. Adaptive equalization( in case of GSM)
4. Rake Receiver (in case of CDMA)

1.Coding
Coding includes:
Speech coding,
Convolutional coding or Forward Error Correction coding
Interleaving
Speech Coding
Human speech is band limited between 300Hz to 3400Hz and undergoes Frequency
Modulation in analog systems. In digital fixed PSTN systems band limited speech is
sampled at the rate of 8 KHz and each sampled is encoded into 8 bits leading to 64Kbps
(PCM A-Law of encoding).Digital cellular radio cannot handle the high bit rate used for
PSTN systems. Smart techniques for signal analysis and processing have been developed
for reduction of the bit rate
Different mobile communication systems use different bit rates for voice encoding. The
following table gives a glimpse.
No.
Technology
Bit rate per voice
Voice coding
chl
technique
1
GSM
13Kbps
RPE-LTP
2
CDMA IS95A
9.6Kbps/14.4 Kbps
QCELP/EVRC
3
Cor-DECT
32Kbps
ADPCM
RPE-LTP: Regular Pulse Excited Long Term Prediction
QCELP: Qualcomm Code Excited Linear Prediction
EVRC:
Enhanced Variable Rate Coding
ADPCM: Adaptive Differential Pulse Code Modulation
Forward Error Correction Coding:
Sometimes this process is called Convolutional Coding or Channel Coding. The
purpose of this process is to build redundancy in the signal so that even if error occurs,
the receiver will be able to recover the lost information. Several methods are available for
this purpose and each mobile system uses its own choice.
Interleaving:
Interleaving is a simple, but powerful, method of reducing the effects of burst errors and
recovering bits when burst errors occur. The symbols (output of Forward Error
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Correction Coder) from each group are interleaved in a pattern that the receiver knows.
The interleaver is located at the BTS and in the phone.
An illustrative example is shown below.

2. Diversity Techniques:
To cope up with the mobile environment Diversity techniques are employed .This
can be Space Diversity, Polarisation Diversity, Frequency Diversity and Time Diversity.
Space and Polarisation Diversity:
It is implemented in the BTS by deploying two antennas, one for
Transmitting and receiving, the other for only receiving. Both antennas should be kept
with minimal separation (10 times wave length). Space Diversity can be combined with
Polarisation Diversity by making the Diversity antenna in an opposite polarization. In
modern times the same antenna with dual polarized elements are available so that with
single antenna, at least polarisation diversity can be achieved. Space Diversity can be
implemented only when sufficient space is available in the tower for mounting the
antennas.
Frequency Diversity:
Signal degradation can be averted by changing the present frequency to another in
case of narrow band systems. This avoids frequency selective fading. In a narrow band
system like GSM this is achieved by slowly hopping the frequency of transmission of
BTS in a predetermined manner.
In case of a wide band system like CDMA signal occupies a large bandwidth and
frequency diversity is inherently achieved.
Time Diversity:
In all the mobile communication systems by employing interleaving time
diversity is automatically achieved.
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3. Adaptive Equalisation:
The transmitter trains the receiver to adapt to the air environment by sending a
known sequence along with the data. Corrections as applied to the known sequence are
applied to the data to retrieve it error free. This is used in GSM.
4. Rake Receiver:
The rake receiver is multiple receivers in one. There is a rake receiver at both the mobile
and BTS. It turns what is a problem in other technologies into an advantage for CDMA.
Signals sent over the air can take multi-paths resulting in degradation of signal. The rake
receiver identifies the three strongest multi-path signals and combines them to produce
one very strong signal. The rake receiver therefore uses multipath to reduce the power the
transmitter must send.

Conclusion:
Wireless means convenience. However to achieve this certain precautionary
measures are taken to overcome the bandwidth scarcity, multipath problems, etc., There
are multiple access techniques to share the bandwidth amongst several users.
Cell Concepts
Frequency scarcity problem
Consider a city where one lakh mobile subscribers need to be served. Considering
that a single RF loop requires a frequency spectrum of 50KHz, total 1,00,000*50KHz =
5GHz of RF bandwidth would be needed to serve these subscribers if an individual
dedicated RF loop is needed for every subscriber. Obviously such a huge bandwidth is
not available that too for a limited number of subscribers in a city. This clearly shows that
an acute problem of frequency scarcity is going to dictate the design and implementation
of a mobile radio system.
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In order to handle this problem one of the obvious measures is not to allot a
dedicated RF channel to an individual subscriber. Rather a group of a few common RF
channels would be available to a relatively large number of subscribers so that RF
channel is allocated only when a mobile user wants to make the call.
Even if we consider the number of RF channel required to serve, for example, a
city of 10Km radius with a subscriber density of 30 subs/ Sq.Km, the RF resource
requirement for 1% grade of service with 30mE traffic per mobile subscriber depicted in
the following table.
City Radius (Km)

Area (Sq.Km)

No. of Subs

RF Chls reqd.

3.14

100

28.3

900

38

10

314

10,000

@ 360

25

960

60,000

@2000

Clearly this is not practical to allot 360 radio channels (360*50KHz = 18 MHz)
for only 10,000 subscribers of this single city alone.
hence frequency reuse is a must to cover the total service area with a limited
available rf resources
hence the need for a cellular principle
What is a Cell?

Fig 1. OMNI DIRECTIONAL CELLS

A base station (transmitter) having a number of RF channels is called a cell.
Each cell covers a limited number of mobile subscribers within the cell
boundaries (Coverage area)
Typical Cell Radius Approx = 30Km (Start up), 1Km (Mature)
Concept of a cluster of cells
In a cellular network a concept of frequency reuse is applied wherein RF carrier
frequencies used in a given geographical location are simultaneously reused at
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geographically separated locations (cells. A typical seven cell pattern is depicted

in figure 2 wherein the total available frequency resource is divided in seven parts
consisting of a number of radio channels. Such a group of cells where the
available frequency spectrum is consumed within those cells only is called a
cluster of cells.

Given Freq.
Resource

Fig 2. A CLUSTER OF CELLS

1 2 3 4 5 6 7

These seven sets of different frequency (a cluster size N of seven) can be reused after
certain distance as shown in figure 3.

Given Freq.
Resource

1 2 3 4 5 6 7
Given Freq.
Resource
1 2 3 4 5 6 7
Frequency Reuse Pattern N = 7

Fig 3
Co-Channel Interference
Co-Channel Interference in a multi-cell environment

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D
R

The cells highlighted

Figas4.inAfigMulti
Cell
4 in the
adjacent cluster use same set of RF
channels and hence are termed as co-channel cells. The distance between the cells using
the same frequency should be sufficient to keep the co-channel interference to an
acceptable level.
Co-channel Interference function

Co-channel interference is
a function of Q
Q=D/R

D
R

Higher Q
Lower Q

Reduced Co-Channel Interference

Increased Co-Channel Interference

Co-Channel Interference
Q = D / R = 3N

N= Cluster Size
R= Size (Radius of Cell)
D= Distance between two Co-Chl Cells
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1701

Q = D/R
Higher Q

Less Interference

1.73

Higher N

3.00

More Cluster Size

3.46

Less RF freq/cell

4.58

Less Traffic Handling Capacity of the system

5.20

12

6.00

Lower Q

Higher Interference
Increased System Handling Capacity

Sectorisation of cells
One way of reducing the level of interference is use directional antenna at base
stations, with each antenna illuminating a sector of the cell, and with a separate channel
set allocated to each sector. There are two commonly used methods of Sectorisation
either using three 120 sector or 60 sector, both of which reduces the number of prime
interference sources.
f3

f3
1
3

1
f1

2
f2
Omni Directional

f1
2

f2

Three Sectored Cell

Three Sectored Cell

Fig 6.

The three sector case is generally used with a seven cell pattern, giving an over all
requirement for 21 channel sets as shown in fig 7 & 8.

A three sectored cell configuration

171
Single location

FREQUENCY REUSE PATTERN

Fig 8
Frequency assignment in cells
Concept of frequency assignment
The cell layout (4-Cell, 3-Sectored) is shown in fig 9.
3A

1A
4A

1C

1B
2A

4B

4C

3C

4A

2B

3C

3B

1C

4A

3C
2A

4B

2C

3A

1B

2B

2B

4C
4A

2C

3A

4B
4C

1B

4B

4C

1A

3B
2A

1C

2C

3A

1A

3B

3C

Fig 9
Frequency Assignment
1A 1, 13, 25, 37, 49, 61, 73, 85, 97, 109, 121
2A 2, 14, 26, 38, 50, 62, 74, 86, 98, 110, 122
1B 5, 17, 29, 41, 53, 65, 77, 89, 101, 113
1C 9, 21, 33, 45, 57, 69, 81, 93

172
1721

1A
1B

3B
1C

BTS
BSC

1.5.1 Frequency planning aspects

Frequency planning aspects
A1 B1 C1 D1 A2

B2

C2

D2

A3

B3

C3

D3

10

11

12

13

14

15

16

17

18

19

20

21

22

23

24

25

26

27

28

29

30

31

32

33

34

35

36

A1
A2
A3

D1

B1

D2
D3

B2
B3

C1
C2
C3

Fig 10
Trunking Efficiency
Cells in City / Sub-urban areas.
Start Up Cells with Larger Diameter
Mature cells with Smaller Diameters
Cells in city center Smaller
Diameters
Cells in Sub-urban areas Larger
Diameters

Fig capa
11
Number of access channel and system
city

Extract from Traffic Table Erlang B Model

Number of Access
Channels

GoS

5
10
20
33
50
56
99
100

0.5%

1.0%

2%

1.13
3.96
11.1
21.5
36.0
41.2
80.0
80.9

1.36
4.46
12.0
22.9
37.9
43.3
84.1
85.0

1.66
5.08
13.2
24.6
40.3
45.9
87.0
88.0

TRUNKING EFFICIENCY
Sr.No

Number of Access Channels

Erlang Capacity

Number of Users Served

5
10
20
50
100

1.66
5.08
13.2
40.3
88.0

16
50
132
403
880

1
2
3
4
5

G
O
S
2%

TRUNKING EFFICIENCY
The Number Of Users Served In A Cell Are Directly Proportional To The Access
Channels Allocated In A Cell
More The Number Of Access Channels In A Cell Further Increase In The
System Handling Capacity

TRUNKING EFFICIENCY

10

50 Subs

10

50 Subs

20

132 Subs

It is better to have a single cell than to split into two with half the number of
access channels
Capacity Considerations
Radio frequency carrier and its allotment to traffic and control.
TDMA time slots (BP0 to BP7) in one radio frequency carrier are allotted for
traffic and control channels. Normally BP0 only for control channel.
Traffic channel (TCH) is used to carry speech and data. These are dedicated
channels.
Common channels are be used by both idle mode and dedicated mode mobiles.
Capacity Considerations.
Capacity consideration with one RF carrier per cell in a site is shown below.

1
2

1
1

With 2% GoS
2.94 E

1
8

2.94 E/25mE=120
Subs

8 Access Channels
1 Signaling
7 - Voice
Capacity consideration with more than one RF carrier per cell in a site is shown below
120 Subs/Sector 3 = 360 Subscribers
1
1

4
1

8 Access Channels
1 Signaling
7 - Voice
1203 = 360 Subs

12
4

32 Access
Channels
3 Signaling
8403 = 2520 Subs

12

12

96 Access
Channels
9 Signaling
32003 = 9600 Subs

Fixed telephones, using wired access network, are meant to be used at a particular
location only. We can have telephones at our office/business and our residence. The fixed
telephones are linked to a place but the modern day life style demands that we should
have telephone facility while on move also. Mobile communication facilitates telephonic
conversation in a fast moving vehicle. This means that phones moves along with a person
thereby moving telephone is linked to a person and not to a place. In this words our reach
becomes broader and world shrinks into a Global village.
Mobile communication objectives
The important objectives of the mobile communication are
Any time Anywhere communication
Mobility & Roaming
High capacity & subs. density
Efficient use of radio spectrum
Seamless Network Architecture
Low cost
Innovative Services
Standard Interfaces

Introduction to GSM
The Global System for Mobile Communications (GSM) is a digital cellular
commununication system. It was developed in order to create a common European
mobile telephone standard but it has been rapidly accepted world wide.
Mobile communication Generations
1G

- analog (cellular revolution)

-only mobile voice services

2G

- digital (breaking digital barrier)

- mostly for voice services & data delivery possible

3G

- Voice & data ( breaking data barrier)

- Mainly for data services where voice services

Frequency bands used in GSM

GSM 900 Mhz
DCS 1800 MHz

will also be possible.

GSM 900 Mhz

Mobile to Cell(UP-LINK)
Cell to Mobile (DOWN -LIN

- 890 to 915 MHz

-

935 to 960 MHz

GSM ARCHITCTURE
A GSM network basically consists of four main subsystems: Mobile station &
The SIM, The Base Station Subsystem, The Network and Switching Subsystem and the
Operation and Support Subsystem.
GSM Network Structure
Every telephone network needs a well-designed structure in order to route
incoming calls to the correct exchange and finally to the called subscriber. In a mobile
network, this structure is of great importance because of the mobility of all its subscribers
.In the GSM system, the network is divided into the following partitioned areas.
GSM service area
PLMN service area
MSC service area
Location area
Cells
The GSM service area is the total area served by the combination of all member
countries where a mobile can be serviced.
The next level is the PLMN service area. A Public Land Mobile Network
(PLMN) is the area served by one network operator. There can be several within a
country, based on its size. All incoming calls for a GSM/PLMN network will be routed to
a gateway MSC (Mobile switching center). A gateway MSC works as an incoming transit
exchange for the GSM/PLMN.

In a GSM/PLMN network, all mobile-terminated calls will be routed to a gateway

MSC. Call connections between PLMNs, or to fixed networks, must be routed through
certain designated MSCs called a gateway MSC. The gateway MSC contains the
interworking functions to make these connections. They also route incoming calls to the
proper MSC within the network.
The next level of division is the MSC/VLR service area. In one PLMN there can
be several MSC/VLR service areas. MSC/VLR is a role controller of calls within its
jurisdiction. In order to route a call to a mobile subscriber, the path through links to the
MSC in the MSC area where the subscriber is currently located. The mobile location can
be uniquely identified since the MS is registered in a VLR, which is generally associated
with an MSC.

The next division level is that of the LAs (Location Area) within a MSC/VLR
combination. There are several LAs within one MSc/VLR combination.

A LA is a part of the MSC/VLR service area in which a MS (Mobile Station)

may move freely without updating location information to the MSC/VLR exchange that
control the LA. Within a LA a paging message is broadcast in order to find the called
mobile subscriber. The LA can be identified by the system using the location Area
Identity (LAI). The LA is used by the GSM system to search for a subscriber in a active
state.Lastly, a LA is divided into many cells. A cell is an identity served by one BTS.
The MS distinguishes between cells using the Base Station Identification code
(BSIC) that the cell site broadcast over the air.
The architecture of GSM network is shown in the figure below and the functions
of components of the network are briefly explained.

Mobile Station (MS)A Mobile Station consists of two main elements: The mobile terminal(handset)
and the SIM (Subscriber Identity Module).
The SIM is a smart card that identifies the terminal. By inserting the SIM card
into the terminal, the user can have access to all the subscribed services. Without the SIM
card ,the terminal is not operational.
The SIM card is protected by a four digit Personal Identification Number(PIN).In
order to identify a subscriber to the system, the SIM card contains some parameters of the
user such as its International Mobile Subscriber Identity (IMSI).
The Base Station Subsystem (BSS)
The BSS connects the Mobile Station and the Network and Switching Subsystem
(NSS).It is in charge of transmission and reception. The BSS can be divided into two
parts:The Base Transceiver Station (BTS) and The Base Station Controller (BSC).
The BTS corresponds to the Transceivers and the Antennas used in each cell of
the network. A BTS is usually placed in the centre of a cell. Its transmitting power
defines the size of a cell. Each BTS has normally between one and sixteen transceivers
depending on the density of users in the cell.
The BSC controls a group of BTS and manages their radio resources. A BSC is
principally in charge of handovers, frequency hopping, exchange functions and control of
the radio frequency power levels of the BTSs.

The Network and Switching Subsystem

Its main role is to manage the communication between the mobile users and other
users such as mobile users, ISDN users, landline users etc. It also includes data bases
needed in order to store information about the subscribers and to manage their mobility.
The different components of the NSS are briefed below.
The Mobile services Switching Center (MSC)
It is the central component of the NSS. The MSC performs the switching
functions of the network. It also provides connection to other networks.
The Gateway Mobile services Switching Center (GMSC)
A Gateway is a node interconnecting different networks. The GMSC is the
interface between the mobile cellular network and the PSTN. It is in charge of routing
calls from the fixed network toward a GSM user.
Home Location Register (HLR)
The HLR is considered as a very important database that stores information of a
given set of subscribers. It also stores the current location of these subscribers and the

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services to which they have access. The location of the subscriber corresponds to the SS7
address of the Visitor Location Register (VLR) associated to the terminal.
Visitor Location Register (VLR)
The VLR is collocated with an MSC. The VLR contains information obtained
from subscribers HLR in order to provide the subscribed services to visiting users. When
a subscriber enters the covering area of a new MSC, the VLR associated to this MSC will
request information about the new subscriber to its corresponding HLR. The VLR will
then have enough information in order to assure the subscribed services without needing
to ask the HLR each time a communication is established.
The Authentication Center (AuC)
The AuC register is used for security purposes. It provides the parameters needed
for authentication and encryption functions. These parameters help to verify the users
identity.
The Equipment Identity Register (EIR)
The EIR is a register containing information about the mobile equipments
Particularly it contains a list of all valid mobile terminals. A terminal is identified by its
International Mobile Equipment Identity (IMEI). The EIR allows then to forbid calls
from stolen or unauthorized terminals.
There are three classes of ME that are stored in the database, and each group has
different characteristics.
White List White List:- contains those IMEIs that are known to have been assigned
to

valid MSs. This is the category of genuine equipment.

Black List :- contains IMEIs of mobiles that have been reported stolen.
Grey List:- contains IMEIs of mobiles that have problems (for example, faulty
software, wrong make of the equipment). This list contains all MEs with faults not
important enough for barring.
The Operation and Support Subsystem (OSS)
The OSS is connected to the different components of the NSS and to the BSC, in
order to control and monitor the GSM system. It is also in charge of controlling the traffic
load of the BSS.
The OMC provides alarm-handling functions to report and log alarms generated
by the other network entities. The maintenance personnel at the OMC can define that
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criticality of the alarm. Maintenance covers both technical and administrative actions to
maintain and correct the system operation, or to restore normal operations after a
breakdown, in the shortest possible time.
The fault management functions of the OMC allow network devices to be
manually or automatically removed from or restored to service. The status of network
devices can be checked, and tests and diagnostics on various devices can be invoked. For
example, diagnostics may be initiated remotely by the OMC.
Call Processing, SMS and VMS
In this we discuss the call processing aspect and look into specifics case of a mobile
originated (MO) call and a mobile terminated (MT) call. We also look into short message
(SMS) and voice mail service (VMS) as implemented IMPCS pilot project.
RF channel overview: - RF channel play important role in call processing case. These are
basically three types of RF control channel.
Broadcast control channel : The broadcast channels are points to multi-point channel,
which are defined only for down-link direction (BTS to mobile station). They are divided
into:
BCCH (Broad cast control channel:- BCCH acts as a beacon. It informs the mobile
about system configuration parameters (e.g. LAI, CELLIDENTY, NEIGHBOURING
cell identify). Using this information MS choose the best cell to attach to.
BCCH is always transmitted on full power and it is never frequency hopped.
FCCHC frequency correction channel. MS must tune to FCCH to listen to BCCH.
FCCH transmits a constant frequency shift of the radio carrier that is used by the MS for
frequency correction.
SCH (synchronization channel). . SCH is used to synchronize the MS in time .SCH
carries TDMA frame number and BSIC (Base Station Identity Code)

Common control channels : Common control channels are specified as point to

multi-point, which operate only in one direction either in up-link or down-link direction.

PCH (Paging Channel): - PCH is used in down-link direction for sending paging
message to MS whenever there is incoming call.
RACH (Random Access Channel ) :-RACH is used by the MS to request allocation of a
specific dedicated control channel (SDCCH) either in response to a paging message or
for call origination /registration from the MS. this is an up-link channel and operate in
point to point mode.
AGCH (Access Grant Channel ):- AGCH is a logical control channel which is used to
allocated a specific dedicated control channel (SDCCH) to MS when MS request for a
channel over RACH. AGCH is used in downlink direction.

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3.Dedicated Control Channel : Dedicated control channel are full duplex, point to
point channel. They are used for signalling between the BTS and certain MS. They are
divided into: SACCH (Slow Associated Control Channel): the SACCH is a duplex channel, which is
always allocated to TCH or SDCCH. The SACCH is used for
Radio link supervision measurements.
Power control.
Timing advance information.
In 26 frame traffic multi-frame 13th frame (frame no .12) is used for
SACCH.SACCH is used only for non-urgent procedures.
FACCH (Fast Associated Control Channel). FACCH is requested in case the
requirement of signaling is urgent and signaling requirement can not be met by SACCH.
This is the case when hand-over is required during conversation phase. During the call
FACCH data is transmitted over allocated TCH instead of traffic data. This is marked by
a flag known as stealing flag.
SDCCH (Stand Alone Dedicated Control Channel)- The SDCCH is a duplex, point to
point channel which is used for signaling in higher layer. It carries all the signaling
between BTS & MS when no TCH is allocated to MS. The SDCCH is used for service
request, location updates, subscriber authentication, ciphering. equipment validation and
assignment of a TCH.
Mobile originated (MO) call: - There are four distinct phase of a mobile originated call-Setup phase.
-Ringing phase.
-Conversation phase.
-Release phase.
Out of these phases the setup phase is the most important phase and includes
authentication of the subscriber,

Ciphering of data over radio interface, validation of

mobile equipment, validation of subscriber data at VLR for requests service and
assignment of a voice channel on A-interface by MSC. Whenever MS wants to initiate on
outgoing call or want to send an SMS it requests for a channel to BSS over RACH. On
receiving request from MS, BSS assigns a stand-alone dedicated control channel
(SDCCH) to MS over access grant channel (AGCH). Once a SDCCH has been allocated
to MS all the call set up information flow takes place over SDCCH.
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A connection management (CM) entity initiates a CM Service Request message to

the network. Network tries to establish an MM connections between the MS and the
network and upon successful establishment of MM connection a CM Service Accept
message is received by MS from the network. MS now sends a Call Set up Request to
the network which contains the dialed digits (DD) of the called party. As the call setup
message is received at the MSC/VLR certain check are performed at MSC/VLR likewhether the requested service is provisioned for the subscriber or not, whether the dialed
digits are sufficient or not, any operator determined barring (ODB) does not allow call to
proceed further etc. As these checks are performed at MSC/VLR a Call Proceeding
Message is sent from the network towards the MS. After all the checks are successfully
passed MSC sends Assignment command to the BSS which contains a free voice channel
on A-interface On getting this message BSS allocates a free TCH to the MS and informs
the MS to attach to it. MS on attaching to this TCH informs the BSS about it. On
receiving a response from the BSS, MSC switches the speech path toward the calling MS.
Thus at the end of Assignment the speech path is through from MS to MSC. It is
important to note that at this stage mobile has not connected user connection as yet. MS
at this stage does not listen anything.
After assignment MSC sends a network set-up message to the PSTN requesting
that a call be set up. Included in the message are the MS dialed digits (DD) and details
specifying which trunk should be used for the call. The PSTN may involve several
switching exchanges before finally reaching the final local exchange responsible for
applying the ringing tone to the destination phone. The local exchange will generate the
ringing tone over the trunk, or series of trunk (if several intermediate switching exchange
are involved), to the MSC. At this point in time MS will hear ringing tone. The PSTN
notifies the MSC with a network-alerting message when this event occurs. MSC informs
the MS that the destination number is being alerted. It is important to note that this is
primarily a status message to the MS. The MS hears the ringing tone from the destination
local exchange through the established voice path.
When the destination party goes off hook, PSTN informs the MSC of this event. At
this point, MS is connected to the destination party and billing is started. MSC informs
the MS that connection has been established and MS acknowledges the receipts of the
connect message.

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Under normal condition, the termination of a call is MS initiated or network

initiated. In this scenario, we have assumed that MS initiates the release of the call by
pressing end button and MS send a disconnect message to the MSC. The PSTN party is
notified of the termination of the call by a release message from the MSC. The end- toend connection is terminated. When MSC is left with no side task (e.g. charging
indication etc.) to complete a release message is sent to the MS. MS acknowledges with a
release complete message. All the resources between MSC and the MS are released
completely.
Mobile Terminated (MT) call- The different phases of a mobile terminated call are
- Routing analysis
- Paging.
- Call setup.
- Call release.
The phases of mobile terminated (MT) call are similar to a mobile originated (MO)
call except routing analysis and paging phase. Call to a mobile subscriber in a PLMN first
comes to gateway MSC (GMSC). GMSC is the MSC, which is the capable of querying
HLR for subscriber routing information. GMSC need not to be part of home PLMN,
though it is normal practice to have GMSC as part of PLMN in commercially deployed
networks.
GMSC opens a MAP (Mobile Application Part) dialogue towards HLR and Send /
Routing / Info-Request (SRI request) specific service message is sent to HLR. SRI
request contains MSISDN of the subscriber. HLR based on location information of this
subscriber in its database, opens a MAP dialogue towards VLR and sends Provide /
Roaming / Number-request (PRN request)to the VLR. VLR responds to PRN request
with PRN response message, which carries an MSRN (mobile subscriber roaming
number), which can be used for routing toward visiting MSC in the network. HLR returns
MSRN to GMSC (MSC that queried HLR) in SRI response message. On getting MSRN
the GMSC routes the call towards VMSC The purpose of this entire exercise is to locate
where the terminating mobile subscriber is.
The MSRN received at GMSC is in international format (Country Code + Area
Code + subscriber number). Normally, based on the routing info at GMSC, the call may
be routed out of the GMSC towards VMSC of the terminating subscriber, in which case
appropriate signaling protocol (MF or ISUP) depending on the nature of connecting of
GMSC with subsequent exchange along the route will apply. If at VMSC the terminating
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mobile subscriber is found to be free (idle), paging is initiated for terminating mobile
subscriber. MSC uses the LAI provided by the VLR to determine which BSSs should
page the MS. MSC transmit a message to each of these BSS requesting that a page be
performed. Included in the message is the TMSI of the MS. Each of the BSSs broadcasts
the TMSI of the mobile in a page message on paging channel (PCH).
When MS detects its TMSI broadcast on the paging channel , it responds with a
channel request message over Random Access Channel (RACH). Once BSS receives a
channel

request

message

it

allocates

stand

alone

Dedicated

Control

Channel(SDCCH) and forwards this channel assignment information to the MS over

Access Grant Channel (AGCH). It is over this SDCCH that the MS communicates with
the BSS and MSC until a traffic channel assigned to the MS. MS transmits paging
response message to the BSS over the SDCCH. Included in this message is MS TMSI
and LAI. BSS forwards this paging response message to the MSC. Now Authentication
and Ciphering phases are performed to check the authenticity of MS and encrypt data
over radio interface.
On the network side after paging is initiated, while waiting for paging response, a
defensive timer called, Early ACM timer is run at MSC to avoid network timeouts. On
successfully getting paging response, a setup message is constructed to be sent towards
terminating MS. In case paging fails due to authentication failure or when the subscriber
is out of radio-coverage, the call is cleared.
In case CLIP is not subscribed by the terminating mobile subscriber, calling number
is not included in set-up message. In case CLIP is subscribed and PI value in calling
number parameter indicates presentation allowed the number is included in the set-up
message. In case CLIP is subscribed but PI received in calling number parameter
indicates presentation restricted then number is included only if CLIRO is also
subscribed to.
MS on receiving the set-up message performs compatibility Checking before
responding to the set-up message it is possible that MS might be incompatible for
certain types of call set-ups. Assuming that MS passes compatibility checking, it
acknowledges the call setup with set-up confirm message. After getting set-up confirm
message from the MS, MSC performs assignment phase (similar to one discussed in MO
call) and a voice path is established from MSC to the MS. MS begins altering the user
after it receives the traffic channel assignment. MS send alerting message to the MSC
.MSC upon receiving the alerting indication from the MS, begins generating an audible
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ringing tone to the calling party and sends a network alerting via GMSC to the PSTN.
Prior to this the calling party heard silence.
At this point in the call, MS is alerting the called party by generating on audible
tone. One of the three events can occur-calling party hangs-up, mobile subscriber answers
the phone, or the MSC times out waiting for the mobile subscriber to the answer the call.
Since radio traffic channel is a valuable resource, GSM does not allow a MS to ring
forever.
In the present scenario we have assumed that the mobile subscriber answers the
phone. The MS in response to this action stops alerting and sends a connect message to
the MSC. MSC removes the audible tone to the PSTN and connects the PSTN trunk to
BSS trunk (terrestrial channel) and sends a connect message via GMSC to the PSTN. The
caller and the called party now have a complete talk path. This event typically marks the
beginning of the call for billing purposes. MSC sends a connect acknowledge message to
the MS.
The release triggered by the land user is done in similar way as the release
triggered by mobile user. MSC receives a release message from the network to terminate
end-to-end connection. PSTN stops billing the calling landline subscriber. MSC sends a
disconnect message towards the MS and MS responds by a Release message. MSC
release the connection to the PSTN and acknowledges by sending a Release Complete
message to PSTN. Now the voice trunk between MSC and BSS is cleared, traffic channel
(TCH) is released and the resources are completely released.
The mobile-to-mobile call scenario is a combination of phases encountered in
mobile originated (MO) and mobile terminated (MT) call.
Short Message Service (SMS)
SMS is a simple bearer service and acts as a bi-directional alphanumeric paging
service, which allows value added service provision as well as management services
provision such as advice of charge. A short message can carry at most 160 characters (it
can be less depending upon the type of characters and their coding scheme). The SMS
could be either in broadcast mode (via CBCH channel) or in a point-to-point mode (via
either SDCCH channel if mobile is in idle state, or SACCH if the mobile is in dedicated
mode).
SMS allows to provide many values added service to individual/ corporate
clients. Individuals may be interested in messaging (transmitting messages in compact
way) or leisure services (weather forecast, road traffic, restaurant booking, movies, TV
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programs etc.). Business users may be interested in corporate information (company

performance, stock value), e-commerce etc.
SMS involves specific entities in the GSM network: first is the SMS Service
Centre (SMS-SC simply SMSC) which can be connected to several networks and many
MSCs

(SMS- GMSCs or SMS-IWMSCs) within the same PLMN and which is

addressed by a mobile using a E.164 number of the numbering plan of the PLMN. SMSC
is capable of following functionalitys:
-Transmission of short message towards a mobile, retaining the responsibility of the
message until reception of acknowledgement or expiration of the validity period.
-Reception of the short messages from MS and transmission of acknowledgement to the
PLMN.
-Transferring messages received from Internet to mobile.
The second entity involved by the SMS is the SME (short message entity),
which is responsible for producing or receiving a short message. The SME can be
connected to the SMSC via a data network such as X.25 or IP.
A short message is characterized by its parameters the most significant are the validity
period, the service center time stamp which indicates the SM arrival time at the SC, etc.
In IMPCS (pilot project), the SMS architecture has been implemented by CDoT. The hardware architecture of SMSC is similar to HLR and is located on same
physical platform. It services as an inter-working and relaying function of the message
transfer between two MS. The service provided are(i) Mobile Originated short message- Enables MS to send an SMS ( up-to 140
bytes) to another MS via SMSC.
(ii)Mobile terminated short message- Enables delivery of an SMS to a particular
MS.
(iii)Operator initiated SMS- This facility enables fixed network subscriber to send
an SMS to a mobile subscriber through an operator at SMSC.
(iv)SMS Newsletter Service- A group of mobile subscriber can subscribe to SMSC
for receiving periodic news regarding sports, weather, traffic etc. The subscription
is done through on operator at SMSC. The operator feeds the news segments,
which are transferred, to the subscriber periodically.
Voice Mail System (VMS)
VMS offers function of call answering device in the system. It provides
personal voice mailbox to the subscribers. VMS redirects/forwards voice calls of a
temporarily in accessible subscriber (busy or no reply) to a personal mailbox of the
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subscriber connected to the MSC. Whenever a call is redirected to VMS, it first greets the
caller with a personalised greeting message and prompts the caller to leave the message
in the mailbox. Later on the called party (mobile subscriber) can access the VMS from
PLMN/PSTN phone by means of access code.
VMS interfaces with MSC on E1 lines using R2 MF/CCS#7 signaling
protocol. In IMPCS network the VMS consists of Pentium PC equipped with Dialogic
card loaded with Windows NT 4.0. Dialogic card provides telephony network interface,
voice recording, compression and play. The disk capacity requirement of the PC is totally
application dependent. For 10,000 subscribers, if each subscriber stores 10 minutes of
voice data then disk storage for subscriber voice information is around 20 GB.
General Packet Radio Service (GPRS)
General Packet Radio Service (GPRS) is a mobile data service available to users of GSM
mobile phon. It is often described as "2.5G", that is, a technology between the second
(2G) and third (3G) generations of mobile telephony. It provides moderate speed data
transfer, by using unused TDMA channels in the GSM network. Originally there was
some thought to extend GPRS to cover other standards, but instead those networks are
being converted to use the GSM standard, so that is the only kind of network where
GPRS is in use. GPRS is integrated into GSM standards releases starting with Release 97
and onwards. First it was standardised by ETSI but now that effort has been handed onto
the 3GPP.
GPRS is different from the older Circuit Switched Data (or CSD) connection included in
GSM standards releases before Release 97 (from 1997, the year the standard was feature
frozen). In CSD, a data connection establishes a circuit, and reserves the full bandwidth
of that circuit during the lifetime of the connection. GPRS is packet-switched which
means that multiple users share the same transmission channel, only transmitting when
they have data to send. This means that the total available bandwidth can be immediately
dedicated to those users who are actually sending at any given moment, providing higher
utilization where users only send or receive data intermittently. Web browsing, receiving
e-mails as they arrive and instant messaging are examples of uses that require intermittent
data transfers, which benefit from sharing the available bandwidth.
Usually, GPRS data are billed per kilobytes of information transceived while circuitswitched data connections are billed per second. The latter is to reflect the fact that even
during times when no data are being transferred, the bandwidth is unavailable to other
potential users.
GPRS originally supported (in theory) IP, PPP and X.25 connections. The latter has been
typically used for applications like wireless payment terminals although it has been
removed as a requirement from the standard. X.25 can still be supported over PPP, or

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even over IP, but doing this requires either a router to do encapsulation or intelligence
built into the end terminal.

Mobile Phone and data standards

PTT
MTS
IMTS
AMTS
PALM
ARP
NMT
AMPS
CDPD
GSM
D-AMPS
cdmaOne
PDC
CSD
GPRS
HSCSD
wiDEN
CDMA2000 1xRTT
EDGE
W-CDMA
CDMA2000 1xEV
HSDPA
HSUPA

0G

0.5G
1G

2G

2.5G

2.75G
3G
3.5G
3.75
4G

How does GPRS work ?

GPRS is packet based, wherein GPRS data is handled as a series of "packets" that can be
routed over several paths through the network, rather than as a continuous bit-stream over
a dedicated dial-up connection. With GPRS, the information is split into separate but
related "packets" before being transmitted and reassembled at the receiving end. The
Internet itself is an example of a packet data network, the most famous of many such
network types.
In second-generation mobile networks, calls are handled using traditional circuitswitching technology. A dedicated "circuit", or "timeslot", is allocated between two
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points for the duration of a call. No other phone can use this circuit during the call,
regardless of whether any data is being transmitted or not.
The GPRS standard is delivered in a very elegant manner - with network operators
needing only to add a couple of new infrastructure nodes and making a software upgrade
to some existing GSM network elements.

Key User Features of GPRS

The General Packet Radio Service (GPRS) is a new non-voice value added service that
allows information to be sent and received across a mobile telephone network. It
supplements today's Circuit Switched Data and Short Message Service. GPRS is NOT
related to GPS (the Global Positioning System), a similar acronym that is often used in
mobile contexts. GPRS has several unique features which can be summarized as:
SPE
ED

Theoretical maximum speeds of up to 171.2 kilobits per second (kbps) are achievable
with GPRS using all eight timeslots at the same time. This is about three times as fast as
the data transmission speeds possible over today's fixed telecommunications networks
and ten times as fast as current Circuit Switched Data services on GSM networks. By
allowing information to be transmitted more quickly, immediately and efficiently across
the mobile network, GPRS may well be a relatively less costly mobile data service
compared to SMS and Circuit Switched Data.
IMMEDIA
CY

GPRS facilitates instant connections whereby information can be sent or received

immediately as the need arises, subject to radio coverage. No dial-up modem connection
is necessary. This is why GPRS users are sometimes referred to be as being "always
connected". Immediacy is one of the advantages of GPRS (and SMS) when compared to
Circuit Switched Data. High immediacy is a very important feature for time critical
applications such as remote credit card authorization where it would be unacceptable to
keep the customer waiting for even thirty extra seconds.
NEW
APPLICATIONS,
APPLICATIONS

BETTER

GPRS facilitates several new applications that have not previously been available over
GSM networks due to the limitations in speed of Circuit Switched Data (9.6 kbps) and
message length of the Short Message Service (160 characters). GPRS will fully enable

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the Internet applications you are used to on your desktop from web browsing to chat over
the mobile network. Other new applications for GPRS, profiled later, include file transfer

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and home automation- the ability to remotely access and control in-house appliances and
machines.
SERVICE
ACCESS

To use GPRS, users specifically need:

(1) a mobile phone or terminal that supports GPRS (existing GSM phones do NOT
support GPRS) .
(2) a subscription to a mobile telephone network that supports use of GPRS must be
enabled for that user. Automatic access to the GPRS may be allowed by some
mobile network operators, others will require a specific opt-in knowledge of how
to send and/ or receive GPRS information using their specific model of mobile
phone, including software and hardware configuration (this creates a customer
service requirement)
(3) a destination to send or receive information through GPRS. Whereas with SMS
this was often another mobile phone, in the case of GPRS, it is likely to be an
Internet address, since GPRS is designed to make the Internet fully available to
mobile users for the first time. From day one, GPRS users can access any web
page or other Internet applications- providing an immediate critical mass of uses.
Having looked at the key user features of GPRS, lets look at the key features from a
network operator perspective.
GPRS networks are able to handle higher bit rates than GSM networks, but the data rates
still fall short of what is required to make existing GSM networks deliver services at a
speed comparable to that promised by third-generation networks. The delay in the
deployment of third-generation systems led to the emergence of a technology known as
EDGE. This was capable of delivering services similar to those of third-generation
networks, yet with implementation on the existing second-generation networks (e.g.
GSM).
EDGE stands for 'enhanced data rates for GSM evolution'. The enhancement from GSM
was to GPRS (i.e. voice and packet), while further enhancement of GPRS led to EDGE
networks, as shown in Figure 1. The fundamental concept remains the same, i.e. voice,
CS data and PS data being carried, and the network architecture is the same as in a GPRS
network. Enhancement of HSCSD is known as ECSD (enhanced circuit-switched data),
while enhancement of GPRS is known as EGPRS.
EGPRS implementation has a major effect on protocol structure (e.g. on layer 1 or layer
2). The modulation and coding schemes are quite different in EGPRS (this is explained
later in the chapter).
In ECSD, though user data rates do not go beyond 64 kbps, fewer time slots are required
to achieve this compared HSCSD. The architecture of ECSD is based on HSCSD
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transmission and signalling, thus having minimal impact on existing specifications.

In this chapter we will focus on EDGE network planning aspects from the EGPRS perspective.

Figure 1 EDGE evolution

THE EDGE SYSTEM

As shown in Figure 2, the EDGE system is quite similar to the GPRS system (compare
with Figure 1), but with the capability for higher data rates. The most important change
the new modulation scheme. In GSM and GPRS, the GMSK modulation scheme was
Used
In GMSK modulation, only one bit per symbol is used. In an EDGE network, octagonal
phase-shift keying (8-PSK) modulation is used which enables a threefold higher gross
data rate of 59.2 kbps per radio time slot by transmitting three bits per symbol. GMSK is
a constant-amplitude modulation while 8-PSK has variations in the amplitude. This
amplitude variation changes the radio performance characteristics, so hardware changes
in the base stations are mandatory.

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CHAPTER - 11

CDMA TECHNOLOGY
Access Network:
Access network, the network between local exchange and subscriber, in the
Telecom Network accounts for a major portion of resources both in terms of capital and
manpower. So far, the subscriber loop has remained in the domain of the copper cable
providing cost effective solution in past. Quick deployment of subscriber loop, coverage
of inaccessible and remote locations coupled with modern technology have led to the
emergence of new Access Technologies. The various technological options available are
as follows :
1.
Multi Access Radio Relay
2.
Wireless In Local Loop
3.
Fibre In the Local Loop
Wireless in Local Loop (WILL)
Fixed Wireless telephony in the subscriber access network also known as Wireless in
Local Loop (WLL) is one of the hottest emerging market segments in global
telecommunications today. WLL is generally used as the last mile solution to deliver
basic phone service expeditiously where none has existed before. Flexibility and
expediency are becoming the key driving factors behind the deployment of WILL.
WLL shall facilitate cordless telephony for residential as well as commercial complexes
where people are highly mobile. It is also used in remote areas where it is uneconomical
to lay cables and for rapid development of telephone services. The technology employed
shall depend upon various radio access techniques, like FDMA, TDMA and CDMA.
Different technologies have been developed by the different countries like CT2 from
France, PHS from Japan, DECT from Europe and DAMPS & CDMA from USA. Let us
discuss CDMA technology in WLL application as it has a potential ability to tolerate a
fair amount of interference as compared to other conventional radios. This leads to a
considerable advantage from a system point of view.

SPREAD SPECTRUM PRINCIPLE

Originally Spread spectrum radio technology was developed for military use to counter
the interference by hostile jamming. The broad spectrum of the transmitted signal gives
rise to Spread Spectrum. A Spread Spectrum signal is generated by modulating the
radio frequency (RF) signal with a code consisting of different pseudo random binary
sequences, which is inherently resistant to noisy signal environment.
A number of Spread spectrum RF signals thus generated share the same frequency
spectrum and thus the entire bandwidth available in the band is used by each of the users
using same frequency at the same time.

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Fig-1 CDMA ACCESS A CONCEPT

On the receive side only the signal energy with the selected binary sequence code is
accepted and original information content (data) is recovered. The other users signals,
whose codes do not match contribute only to the noise and are not despread back in
bandwidth (Ref Fig-1) This transmission and reception of signals differentiated by
codes using the same frequency simultaneously by a number of users is known as Code
Division Multiple Access (CDMA) Technique as opposed to conventional method of
Frequency Division Multiple Access and Time Division Multiple Access.
In the above figure, it has been tried to explain that how the base band signal of 9.6 Kbps
is spread using a Pseudo-random Noise (PN) source to occupy entire bandwidth of 1.25
Mhz. At the receiving end this signal will have interference from signals of other users of
the same cell, users of different cells and interference from other noise sources. All these
signals get combined with the desired signal but using a correct PN code the original data
can be reproduced back. CDMA channel in the trans and receive direction is a FDD
(Frequency Division Duplexing) channel. The salient features of a typical CDMA system
are as follows:
Frequency of operation:
824-849Mhz and 869-894 Mhz
Duplexing Mehtod:
Frequency Division Duplexing (FDD)
Access Channel per carrier: Maximum 61 Channels
RF Spacing:
1.25 Mhz
Coverage:
5 Km with hand held telephones and approx.
20 Km with fixed units.
The different types of codes used for identification of traffic channels and users
identification etc as follows:
Different Codes used in CDMA
Walsh Code:
In CDMA the traffic channels are separated by unique Walsh code. All such codes are
orthogonal to each other. The individual subscriber can start communication using one of
these codes. These codes are traffic channel codes and are used for orthogonal spreading
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of the information in the entire bandwidth. Orthogonality provides nearly perfect

isolation between the multiple signals transmitted by the base station.
The basic concept behind creation of the code is as follows:
(a)
Repeat the function right
(b)
Repeat the function below
Invert function (diagonally)
0 ----- 0
0 -------0
0
0
0
0
1
0
1
0
1
0
0
1
1
0
1
1
0
Long Code:
The long pseudo random noise (PN) sequence is based on 242 characteristic polynomial.
With this long code the data in the forward direction (Base to Mobile) is scrambled. The
PN codes are generated using linear shift registers. The long code is unique for the
subscribers and is known as users address mask.
Short Code:
The short pseudo random noise (PN) sequence is based on 215 characteristic polynomial.
This short code differentiates the cells & the sectors in a cell. It also consists of codes for
I & Q channel feeding the modulator.
Advantages: CDMA wireless access provides the following unique advantages.
Larger Capacity:
Let us discuss this issue with the help of Shannons Theorem. It states that the channel
capacity is related to product of available band width and S/N ratio.
C

= W log 2 (1+S/N)

Where C

= channel capacity

= Band width available

S/N

= Signal to noise ratio.

It is clear that even if we improve S/N to a great extent the advantage that we are
expected to get in terms of channel capacity will not be proportionally increased. But
instead if we increase the bandwidth (W), we can achieve more channel capacity even at
a lower S/N. That forms the basis of CDMA approach, wherein increased channel
capacity is obtained by increasing both W & S/N. The S/N can be increased by devising
proper power control methods.
Vocoder and variable data rates:
As the telephone quality speech is band limited to 4 Khz when it is digitized with PCM
its bit rate rises to 64Kb/s vocoding compress it to a lower bit rate to reduce bandwidth.
The transmitting vocoder takes voice samples and generates an encoded speech/packet
for transmission to the receiving vocoder. The receiving vocoder decodes the received
speech packet into voice samples. One of the important feature of the variable rate
vocoder is the use of adaptive threshold to determine the required data rate. Vocoders are
variable rate vocoders. By operating the vocoder at half rate on some of the frames the
capacity of the system can be enhanced without noticeable degradation in the quality of
the speech. This phenomenon helps to absorb the occasional heavy requirement of traffic
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apart from suppression of background noise. Thus the capacity advantage makes spread
spectrum an ideal choice for use in areas where the frequency spectrum is congested.
Less (Optimum) Power per cell:
Power Control Methods: As we have already seen that in CDMA the entire bandwidth of
1.25Mhz is used by all the subscribers served in that area. Hence they all will be
transmitting on the same frequency using the entire bandwidth but separated by different
codes. At the receiving end the noise contributed by all the subscribers is added up. To
minimize the level of interfering signals in CDMA, very powerful power control methods
have been devised and are listed below:
1. Reserve link open loop power control
2. Reserve link closed loop power control
3. Forward link power control
The objective of open loop power control in the reverse link (Mobile to Base) is that the
mobile station should adjust its transmit power according to the changes in its received
power from the base. Open loop power control attempts to ensure that the received signal
strength at the base station from different mobile stations, irrespective of their distances
from the base site, should be same.
In Closed loop power control in reverse link, the base station provides rapid corrections
to the mobile stations open loop estimates to maintain optimum transmit power by the
mobile stations. The base station measures the received signal strength from the mobile
connected to it and compares it with a threshold value and a decision is taken by the base
every 1.25 ms to either increase or decrease the power of the mobile.
In forward link power control (Base to Mobile) the cell (base) adjusts its power in the
forward link for each subscriber, in response to measurements provided by the mobile
station so as to provide more power to the mobile who is relatively far away from the
base or is in a location experiencing more difficult environment.
These power control methods attempt to have an environment which permits high quality
communication (good S/N) and at the same time the interference to other mobile stations
sharing the same CDMA channel is minimum. Thus more numbers of mobile station are
able to use the system without degradation in the performance. Apart from the capacity
advantage thus gained power control extends the life of the battery used in portables and
minimizes the concern of ill effects of RF radiation on the human body.
Seamless Hand-off:
CDMA provides soft hand-off feature for the mobile crossing from one cell to another
cell by combining the signals from both the cells in the transition areas. This improves
the performance of the network at the boundaries of the cells, virtually eliminating the
dropped calls.
No Frequency Planning:
A CDMA system requires no frequency planning as the adjacent cells use the same
common frequency. A typical cellular system (with a repetition rate of 7) and a CDMA

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system is shown in the following figures which clearly indicates that in a CDMA network
no frequency planning is required.

CDMA Frequency

Frequency Reuse of 7 in GSM

High Tolerance to Interference:
The primary advantage of spread spectrum is its ability to tolerate a fair amount of
interfering signals as compared to other conventional systems. This factor provides a
considerable advantage from a system point of view.
Multiple Diversity:
Diversity techniques are often employed to counter the effect of fading. The greater the
number of diversity techniques employed, the better the performance of the system in a
difficult propagation environment.
CDMA has a vastly improved performance as it employs all the three diversity
techniques in the form of the following:
A .Frequency Diversity:

A wide band RF signal of 1.25 Mhz being used.

B. Space Diversity:

Employed by way of multipath rake receiver.

C. Time Diversity:Employed by way of symbol interleaving error detection and

correction coding.

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Capacity Considerations
Let us discuss a typical CDMA wireless in local loop system consisting of a single base
station located at the telephone exchange itself, serving a single cell. In order to
increase the number of subscribers served the cell is further divided into sectors. These
sectors are served by directional antennas.
The capacity of a cellular system is claimed to be 20-40 active lines per sector per 1.25
MHz for a single CDMA Radio Channel. In WLL environment assuming an average
busy hour traffic of 0.1 Erlang, 400 subscribers can be served per sector over a single
1.25 MHz channel.
Assuming typically six sectors in a cell the total capacity of a CDMA network consisting
of 1.25 MHz duplex channels is 2400 (400x6) subscribers.
Capacity can further be increased if we use another frequency on the same base station
covering the same geographical area (overlapping cell). Thus in 10 Mhz in the bandwidth
we can utilize 5 MHz of bandwidth in the forward link and 5 Mhz in the reverse link.
Hence if we have 4 RF carriers in 5 Mhz bandwidth, the network can support 12000
(5x400x6) subscribers per cell.
Conclusion
Hence we see that use of common frequency, multipath rake receiver, power control &
variable bit rate vocoding and soft hand-off features of CDMA give us the benefits of no
frequency planning, larger capacity, flexibility alongwith high performance quality.
Introduction to CDMA 2000-1X
Network entity description
Base station subsystem (BSS) Base station subsystem is the general term for the wireless
devices and wireless channel control devices that serve one or several cells. Generally, a
BSS contains one more base station controllers (BSC) and base transmitter stations
(BTS).
Mobile switch center (MSC)
MSC is a functional entity that performs control and switching to the mobile stations
within the area that it serves, and an automatic connecting device for the subscriber
traffic between the CDMA network and other public networks or other MSCs. MSC is
the kernel of the CDMA cellular mobile communication system, and it is different from a
wired switch in that an MSC must consider the allocation of the wireless resources and
the mobility of subscribers, and at least it must implement the follows processing
activities:
1. Location Registration processing;
2. Handoff.

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Architecture of CDMA MSC Based WLL system

M SC

MS

PSTN

Um
BT S

Ai

E
Abis

B SC

M S C/SSP

VLR

BSS

Q
MC

MC

C
HLR

D
H

AUC

MSS

RGMTTC Presentation

Gateway MSC (GMSC)

When a non-CDMA subscriber calls a CDMA subscriber, the call will first be routed to
an MSC, which will inquires the corresponding HLR and further route the call to the
called partys MSC. This kind of MSC is called Gateway MSC (GMSC). It is up to the
network operator to select which MSCs as GMSCs.
Visitor location register (VLR)
VLR is responsible for the storage and updating of the subscriber data of mobile stations
that roamed to the service area of this VLR. The VLR is generally configured together
with the MSC. When the mobile station enters a new location area, the MSC will notice
the VLR, which will initiate registration processing to the HLR to update the subscriber
location information. The VLR also stores necessary information for the establishment of
calls in the database for the MSC to search. One VLR can cover one or more MSC areas.
Home location register (HLR)
The HLR provides subscriber information storage and management functions for the
mobile network, including mobile subscriber subscription and cancellation and service
authorization and cancellation. At the same time, it helps in the implementation of
subscribers call and service operations. A CDMA can contain one or more HLRs based
on the number of subscribers, equipment capacity and network organization mode, with
multi-HLR mode realized in the form of virtual HLRs. The subscriber information stored
in the HLR includes the following two types in information:

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1. Subscription information
2. Subscriber-related information stored in the HLR
Authentication center (AUC)
Authentication center is a function entity for the management of authentication
information related to the mobile station. It implement mobile subscriber authentication,
stores the mobile subscriber authentication parameters, and is able to generate and
transmit the corresponding authentication parameters based on the request from
MSC/VLR. The authentication parameters in the AUC can be stored in the encrypted
form. The authentication center is generally configured together with the HLR. The
authentication parameter stored in the AUC include:
1. Authentication key (A_KEY);
2. Share secret data (SSD);
3. Mobile identification number/international mobile subscriber identity
(MIN/IMSI);
4. Authentication algorithm (AAV);
5. Accounting (COUNT).
Short message center (MC or SC)
As an independent entity in the CDMA cellular mobile communication system, the short
message center works in coordination with other entities such as MSC, HLR to
implement the reception, storing and transfer of the short messages from CDMA cellular
mobile communication system subscribers, and store subscriber-related short message
data.
Short message entity (SME)
SME is a function entity for synthesis and analysis of short messages.
Operation and maintenance Center (OMC)
The OMC provides the network operator with network operation and maintenance
services, manages the subscriber information and implements network planning, to
enhance the overall working efficiency and service quality of the system. There two type
of operation and maintenance centers: OMC-S and OMC-R. An OMC-S is mainly used
for the maintenance work at the mobile switching subsystem (MSS) side; an OMC-R is
mainly used for the maintenance work at the base station subsystem (BSS) side.
Third Generation Standards
CDMA2000/FDD-MC CDMA2000 using Frequency Division DuplexingMulticarrier (FDD-MC) mode. Here multicarrier implies N x 1.25 MHz channels
overlaid on N existing IS-95 carriers or deployed on unoccupiedspectrum.
CDMA2000 includes:
1x using a spreading rate of 1.2288 Mcps
3x using a spreading rate of 3 x 1.2288 Mcps or 3.6864 Mcps
1xEV-DO (1x Evolution - Data Optimized)using a spreading rate of
1.2288 Mcps optimized for data
WCDMA/FDD-DS Wideband CDMA (WCDMA) Frequency Division

Duplexing-Direct Sequence spreading (FDD-DS) mode. This has a single 5 MHz

channel. WCDMA uses a single carrier per channel and employs a spreading rate of 3.84
Mcps.

UTRA TDD/ TD-SCDMA Universal Mobile Telephone Services Terrestrial

Radio Access (UTRA) and TD-SCDMA. These are Time Division Duplexed
(TDD) standards aimed primarily at asymmetric services used in unpaired (i.e., no
separate uplink and downlink) bands. TD-SCDMA is based on a synchronous
Time Division scheme for TDD and wireless local loop applications. The frame
and slot structure are the same as W-CDMA. However, in TDD mode each slot
can be individually allocated either the uplink or the downlink.
Advantages of CDMA2000:
CDMA2000 is backward compatible with IS-95. Thus a network that is converted to
CDMA2000 from IS-95 will support users with IS-95 handsets. A motivating factor for
migration to CDMA2000-enabled handsets is that it permits use of enhanced data service
and increases the voice capacity of the network. The voice capacity of a
CDMA2000 network increases as the percentage of subscribers with CDMA2000
handsets increases. IS-95 handsets do not contribute to this capacity improvement. It
reuses and builds on the full complement of existing CDMA air interface and network
standards. Both IS-95 and CDMA2000 equipped mobiles can operate on the same
frequency assignment. Existing IS-95 networks can be converted to CDMA2000 without
impact to existing IS-95
The following are the new additions in CDMA 2000 from IS95.
_ Spreading Rate 1 (1x) and Spreading Rate 3 (3x)
_ Logical Channels
_ Radio Configurations
_ Many new Physical Channels
_ Transmit Diversity Pilot Channels
_ Enhanced Access Channel
procedures
_ Reverse Link Pilot Channel
Spreading Rates
CDMA2000 supports two different spreading rates:
Spreading Rate 1 also called 1x
Both Forward and Reverse Channels use a single direct-sequence
spread carrier with a chip rate of 1.2288 Mcps.
Spreading Rate 3 also called 3x or MC (Multi-Carrier)
Forward Channels use three direct-sequence spread carriers each with a chip rate of
1.2288 Mcps.Reverse Channels use a single direct-sequence spread
carrier with a chip rate of 3.6864Mcps. As such SR3 appears to be
extinct.

Physical and Logical Channels:

In IS 95A, in the forward link Pilot, Sync, Paging and Traffic Channels exist where as in
reverse link Access and traffic channel are available. All overhead information is carried
on the Paging Channel. During conversation or in dedicated mode the signaling info is
exchanged by either fully or partially clearing the traffic. CDMA2000 technology
defines new Physical and Logical Channels for the transport of user data and signaling
information.
A Physical Channel is a communication path between the mobile and the Base Station,
described in terms of the digital coding and RF characteristics.
A Logical Channel is a communication path within the protocol layers of either the Base
Station or the mobile.
Radio Configurations: A Radio Configuration (RC) defines the following characteristics
of a Forward or Reverse Traffic Channel, Viz Rate Set, Spreading Rate Channel Coding
(Turbo or convolutional), Channel Coding Rate, Modulation (QPSK or BPSK)and
Transmit
Diversity Allowed.
IS-2000 defines Radio Configurations:
- RC1 and RC2 correspond to IS-95 A/B Rate Set 1 and Rate
Set 2 respectively - RC3 through RC9 on the Forward link
- RC3 through RC6 on the Reverse link
Variable Length Walsh Codes: Walsh Code used in IS95 is 64 chips long. CDMA20001x
can use Walsh Codes up to 128 Chips long. Higher data rate channels use shorter length
Walsh codes to maintain a constant chip rate. Using one of the shorter Walsh codes
precludes using all longer codes that contain the bit pattern of the shorter code.
New Common Channels:
CDMA2000 introduces several new Forward Link Common Channels:
Pilot Channels - If transmit diversity is supported; one or more Pilots may be used.
The auxiliary Pilot Channels may be used for smart antenna applications.
Quick Paging Channel - This channel provides for improved slotted mode
operation and improved battery life for the mobile. Walsh codes W80, W48 and W112
are reserved for Quick Paging Channels, if the Base Station supports Quick Paging
Channels.
Common Control Channel - This channel carries mobile-directed
messages for CDMA2000 compatible mobiles.
Broadcast Channel - This channel carries broadcast messages for CDMA2000
compatible mobiles, including overhead messages and broadcast Short Message
Service (SMS) messages.
Common Power Control Channel - This channel is used with Enhanced Access
Channel Procedures (Reservation Mode), to send power control bits to the mobile so that
Access Channel messages may be sent under power control.
New Dedicated Channels:
CDMA2000 introduces several new Forward Link Dedicated Channels:
Forward Fundamental Channel - This channel is used for the transmission of user and
signaling information to a specific mobile during a call. Each Forward Traffic Channel
may contain one Forward Fundamental Channel.

 Forward Dedicated Control Channel - This channel is used for transmission of user
and signaling information to a specific mobile during a call. Each Forward Traffic
Channel may contain one Forward Dedicated Control Channel.
Forward Supplemental Channel (valid for Radio Configurations 3 thro 9) This channel
is used for the transmission of user information to a specific mobile during a call. This is
typically used for high-speed data applications. Each Forward Traffic Channel may
contain up to two Supplemental Channels.
Power Control Subchannel - This subchannel is typically associated with the
Fundamental Channel, but if the F-FCH is not used for a given call, then it is associated
with the Dedicated Control Channel (F-DCCH).
All of the CDMA2000 dedicated channels can be established using
the TIA/EIA Paging (F-PCH) and Access (R-ACH) Channels.
Reverse Link Channels:
- Access Channel (R-ACH)
_ Reverse Pilot Channel (R-PICH)
_ Enhanced Access Channel (R-EACH)
_ Reverse Common Control Channel (R-CCCH)
_ Reversed Dedicated Control Channel (R-DCCH)
_ Reverse Fundamental Channel (R-FCH)
_ Reverse Supplemental Channel (R-SCH)
_ Reverse Supplemental Code Channel (R-SCCH)
The Access Channel and Reverse Supplemental Channel are retained for backward
compatibility with TIA/EIA-95A/B. For Radio Configurations 1 and 2, the channel
structure for the Reverse Fundamental Channel and Reverse Supplemental Channel is the
same as the channel structure of Rate Set 1 and Rate Set 2 used in TIA/EIA-95A/B.
EV-DO
EV-DO is a mobile technology that facilitates higher throughput on mobile platform.
The third generation of cellular standards has seen a dominance of CDMA as the
underlying access technology. UMTS (Universal Mobile Telecommunication Services) is
3G evolution for GSM world. The standardization work for UMTS is being carried-out
by 3GPP. The standardization work for CDMA 2000 and its enhancements is being
carried out under the supervision of 3GPP2.
1x Evolution-Data Optimized, abbreviated as EV-DO or 1xEV-DO, is an evolution of
CDMA 2000 1x to support higher data rates. It is defined in TIA (Telecommunication
Industry Association) standard IS 856. It is commonly referred to as DO. It is officially
termed as "CDMA2000, High Rate Packet Data Air Interface". Working on same carrier
bandwidth of 1.25 MHz as CDMA 2000 1x systems, 1xEV-DO provides significantly
higher data rates to Access Terminals (mobile devices). Downlink data rates supported
are up to 2.4576 Mb/s in Rev. 0 and up to 3.1 Mb/s in Rev. A.
Traditional wireless networks create a physical path between receiving and sending
devices, much like traditional telephone networks. EVDO instead adopts the same
approach used for the internet. IP, the Internet Protocol, breaks data into small pieces
called packets. Each packet is sent independently of all the other packets. This saves

bandwidth for use by other devices; when neither party on a phone call is speaking, the
connection consumes no bandwidth because there are no packets to send. Radio resources
are allocated only at the time of actual data transfer leading to better spectral efficiency.
EV-DO does not support voice services. In Forward link supports data rates up to 2.4576
Mbps. There is no power control in Forward Link. Peak data rate in Reverse Link is
153.6 kbps.
Generic Model of CDMA 2000 1x EVDO System:
A generic model of a CDMA 2000 1 x EV-DO System typically consists of:
a) Access Network (AN) consisting of Radio Node (RN) & Radio Network Controller
b) Packet Core Network (PCN)
a) Radio Node (RN): It is a multiple circuit transceiver which shall radiate to cover a cell
or a sector. It consists of radio modules, base band signal processor, network interface,
antenna, feeder etc. It can be co-located with RNC or remotely located. RN shall include
the functions related to channel coding/decoding, interleaving, encryption, frame
building, modulation/demodulation, RF transceiver, antenna diversity, low noise
amplification etc. as per CDMA 2000 1 x EV-DO standards.
The AN obtains the timing reference and positioning reference from the GPS system
and hence the GPS receiver shall form an integral part of the RN along with other fixtures
such as GOS antenna, cable etc. AN split mounting arrangements with tower mountable
RF components such as PAs, LNAs, Filters etc. are also acceptable.
b) Radio Network Controller (RNC): It is responsible for inter connection between the
RN and the PCN and it provides control and management for one or more RNs. It assigns
traffic channels to individual users, monitors system performance and provides interface
between the RN and the PCN. RNC performs the radio processing functions such as
management of the radio resources, radio channel management, local connection
management etc.
It also processes information required for decision on handover of calls from one RN to
another. RNC can be collocated with the PCN or remotely located. The Packet Control
Function (PCF) shall form an integral part of RNC.
Packet Core Network (PCN): The packet data core network provides packet data
services to Access Terminal (AT) and consists of PDSN, HA, AAA, AN-AAA and
FA functionalities. The functional entities AAA and ANAAA may be a single physical
entity or two separate physical entities.
Operations and Maintenance Centre (OMC): The Operations and Maintenance Centre
(OMC) allow the centralized operation of the various units in the system and the
functions needed to maintain the sub systems. The OMC provides the dynamic
monitoring and controlling of the network management functions for operation and
maintenance.
Call Processing in CDMA
Call processing refers to all the necessary functions that the system needs to carry out in
order to set up, maintain, and tear down a call between a mobile and another party.

Two types of connections are possible:

Mobile-to-Land call or Mobile-to-mobile call.
Call can be either mobile originated or mobile terminated. Since mobile station is the
common element in both cases the IS-95 standard specifies the call states from the
perspective of the CDMA.
States of mobile:
During normal operation, the mobile can occupy any one of the following states
Mobile station initialization state;
Mobile station idle state;
System access state;
Mobile station control on the traffic channel state.
After power-up, the mobile first enters the mobile station initialization state,
where the mobile selects and acquires a system. Upon exiting the initialization
state, the mobile has fully acquired the system and its timing.
Then the mobile enters the mobile station idle state where the mobile monitors
messages on the paging channel .
Any one of the following three events will cause the mobile to transition from the
idle state to the system access state
1) The mobile receives a paging channel message requiring an acknowledgment
or response,
2) The mobile originates a call, or
3) The mobile performs a registration.
In the access state, the mobile sends messages to the base station on the access
channel.
When the mobile is directed to a traffic channel, it enters the mobile station
control on the traffic channel state where the mobile communicates with the base
station using the forward and reverse traffic channels. When the call is terminated,
the mobile returns to the initialization state.
Initialization State
After power-up, the mobile enters the initialization state. This state contains four sub
states, which the mobile sequentially goes through:
1. System determination sub state;
2. Pilot channel acquisition sub state;
3. Sync channel acquisition sub state;
4. Timing change sub state
Idle State:

Paging Channel Monitoring

In the idle state, the mobile monitors the paging channel on the forward link. In order to
receive messages and receive an incoming call, the mobile needs to monitor the paging
channel for messages. The paging channel transmission is divided into slots that are 80
ms in length. There are two ways that the mobile can monitor the paging channel:
Nonslotted mode
Slotted mode.
In non-slotted mode, the mobile monitors the paging channel at all times. In slotted
mode, the mobile monitors the paging channel only during assigned paging channel slots.
Because the mobile doesnt have to monitor all the slots all the time, the mobile operating
in the slotted mode can conserve battery power.
Paging Channel Messages
There are a total of six overhead messages that are sent to the mobile on the paging
channel:
System parameters message;
Neighbor list message;
CDMA channel list message;
Extended system parameters message;
Global service redirection message;
Access parameter message.
The first five messages are of configuration parameters and the last message is of access
information.
Access State
In the access state, the mobile transmits messages to the base station using the access
channel. In addition, the mobile also receives messages from the base station on the
paging channel. There are six sub states that the mobile can occupy within the access
state.
Update overhead information sub state;
Page response sub state;
Mobile station origination attempt sub state;
Registration access sub state;
Mobile station order/message response sub state;
Mobile station message transmission sub state.
Traffic channels state
The mobile may enter the traffic channel state from 2 sub states within the access state:
the page response of state or the mobile state origination attempt sub state. In other words

after the mobile has successfully originated, the mobile may enter the traffic channel
state. In the traffic channel state, the mobile communicates with the base station using the
forward and reverse traffic channels. This state consists of 5 sub states
traffic channel initialization sub state
waiting for order sub state
waiting for mobile state answer sub state
conversation sub state
release sub state
Hand Offs in CDMA
As the phone moves through a network the system controller transfers the call from one
cell to another, this process is called handoff. Handoffs maybe done with the assistance
of the mobile or the system controller will control the process by itself. Handoffs are
necessary to continue the call as the phone travels. Handoffs may also occur in idle state
due to mobility.
Types of Handoffs in CDMA: There are primarily three types of Handoffs in CDMA.
They are
Soft
Hard and
Idle.
The type of handoff depends on the handoff situation.
To understand this we should know the cellular concept used in CDMA.
CDMA frequency- reuse planning (cellular concept):
Each BTS in a CDMA network can use all available frequencies. Adjacent cells can
transmit at the same frequency because users are separated by code channels, not
frequency channels. BTSs are separated by offsets in the short PN code This feature of
CDMA, called "frequency reuse of one," eliminates the need for frequency planning
Soft Handoff:
A soft handoff establishes a connection with the new BTS prior to breaking the
connection with the old one. This is possible because CDMA cells use the same
frequency and because the mobile uses a rake receiver. The CDMA mobile assists the
network in the handoff. The mobile detects a new pilot as it travels to the next coverage
area. The new base station then establishes a connection with the mobile. This new
communication link is established while the mobile maintains the link with the old BTS.
Soft handoffs are also called "make-before-break." Soft handoff can take place only when
the serving cell and target cell are working in the same frequency.

TUTORIAL on CDMA Principles

1) MATCH THE FOLLOWING:
COLUMN-A

COLUMN-B

1)RF Spacing in CDMA:

1)..........

A)Walsh code 0

2)RF Spacing in WCDMA

2)..........

B) 869-889MHz

3) Pilot channel

3)..........

C) Walsh code 32

4) Sync channel

4)..........

D) Walsh codes 1-7

5) Paging channel

5)..........

E) 1.25 Mhz

6) Traffic channel

6)..........

F) 824-844MHz

7) Reverse link freq

7). G) 1.25 Mhz

8) Forward link freq

8)

I) Walsh codes 8-31, 33-63.

9)Space Diversity

9)..

J)Employed by way of symbol

interleaving

10)Time Diversity

10)

K)Employed by way of multipath rake

receiver

2) SAY TRUE OR FALSE

1) CDMA is a spread spectrum technique.
2) In CDMA all subs can use the same frequency.
3) Walsh codes are orthogonal codes to each other.
4) For soft hand off freq of operation of the cells involved should be same.
5) CDMA system uses FDD technique.
6) CDMA supports two rate sets for voice encoding.
7) In CDMA power control is applicable only for forward link.
8) Hard hand off is a break before make type.
9) No of access channels will be normally equal to no. of paging channels.

10) User mask is derived from electronic serial number of the hand set/ FWT.
11) Symbol is produced after coding process.
12) Chip is produced after spreading process.
13) CDMA system power control is applied only in the reverse link.
14) All CDMA system need GPS support for their functioning.
15) Long code is used for scrambling in the forward link.
16) Short code gives the BTS ID.
3) CHOOSE THE BEST CORRECT ANSWER:
1) Softer hand off is of _
before break

type

A) break before make B) make

2) Sequence used for spreading at trans end is 1001, then receiver will use
for despreading
A) 0110 B) 1001 C) 1010 D) 1111
3) Long code is of length

A) 2^15-1 B) 2^42-1 C) 2^8-1

4) Long and short codes are: A) orthogonal codes B) pseudo random codes
C) none of the above
5) In forward link spreading is done by A) Walsh code B) Long code
6) If there are 4 shift registers in a PN code generator, the length of the code is
A) 15 bits B) 16 bits C) 8 bits
7) Forward link is

A) BTS to mobile B) Mobile to BTS

8) Reverse link is
A) BTS to mobile B) Mobile to BTS

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9) Forward link works in

10) There are

A) 824- 849MHz B) 869-894MHz

no. of Walsh codes.

11) Length of each Walsh code is

A) 20 B) 64 C) 50

A) 64 chips B) 23 chips C) 12 chips

12)Space Diversity is achieved in cdma by

A) rake receiver B) Two antenna C) Booster antenna
Questions for CDMA2000-1X
1) MATCH THE FOLLOWING:
1) MSC-VLR

1)..........

A) E interface

2) MSC-HLR

2)..........

B) 3G Standard

3) HLR-VLR

3)..........

C) 3.6864Mcps

4) MSC-MSC

4)..........

D) N interface

5) HLR- SMC

5)..........

E) peak data 2.4Mbps

6) CDMA IS 95 A network

6)..........

F) C interface

7) CDMA 2000 1X-EVDO

7)..........

G) B interface

8) CDMA 20001X network

8)..........

H) D interface

9) Spreading Rate 1

9) I)2G Standard

10)Spreading Rate 3

10).

J) 1.2288 Mcps.

2) SAY TRUE OR FALSE:

1) One BSC can control only one BTS
2) CC for India is 404
3) MCC for India is +91
4) MSIN and MIN are same.
5) ESN is a unique number assigned to the instrument.

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6) CDMA IS 95A is a 2G standard.

7) IMT 2000 is the official name for 3G.
8) CDMA 2000 is backward compatible to IS 95A.
9) CDMA 2000 1x uses single carrier.
10) CDMA 2000 3x is multi carrier system.
11) RC1 and2 of CDMA2000 correspond to rate set 1 and2 of IS95A.
12) Variable length Walsh Codes are used in IS95A
13) In RC1 and RC2 the no. of supplemental channels are seven maximum.

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CHAPTER 12

INTRODUCTION TO BROADBAND
Objectives
The main objective of this chapter is to build up the following
i)

To understand what is Broadband

ii)

To understand the need of broadband

iii)

To familiarize with Broadband Network

Overview
The confluence of two forcesthe globalization of business and the networking of
information technologyhas created the Internet economy. Advances in
telecommunications and data technology are creating new opportunities for countries,
businesses and individualsjust as the Industrial Revolution changed fortunes around
the globe. The new economy is defining how people do business, communicate, shop,
have fun, learn, and live on a global basisconnecting anyone to anything. The
evolution of Internet has come into existence & Internet service is expanding rapidly.
The demands it has placed upon the public network, especially the access network, are
great.
The rapid growth of distributed business applications; the proliferation of private
networks, e-commerce, and bandwidth-intensive applications (such as multimedia,
videoconferencing, and video on demand) generate the demand for bandwidth and access
network.
Moreover, an increasing number of consumers in this area further leads to a demand for
carrying these applications faster and reliable. Essentially, the broadband revolution is
about a huge increase in the range of services that can be offered via the Internet and
digital television. It promises a new age in entertainment and communications, as well as
a major boost for e-commerce. To meet this explosive demand for bandwidth and to
capitalize on this growing data opportunity, many data competitive local-exchange
carriers are aggressively targeting small businesses, SOHOs, and teleworkers in the
selected areas of the country in which they are operating.
However, technological advances promise big increases in access speeds, enabling public
networks to play a major role in delivering new and improved telecommunications
services and applications to consumers .The Internet and the network congestion that
followed, has led people to focus both on the first and last mile as well as on creating a

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different network infrastructure to avoid the network congestion and access problems.
The solution to this is Broadband.
As a result, many different companies have worked to develop "broadband" or
high-speed access. These broadband services will allow Internet subscribers to send or
receive video and audio content of digital quality; to download interactive graphic-rich
WebPages; and to allow Internet entrepreneurs to bring new services to market that take
advantage of speeds that will make the Internet truly interactive in real time. In short,
broadband promises to revolutionize the Internet in the same way that the introduction of
the Internet revolutionized communication.
In deploying these broadband services, service providers are developing whole
new ways to access the Internet. The Internet was never designed to handle the amount of
traffic that we are seeing today. The increasing penetration of broadband access and the
demand for multimedia applications is exacerbating this problem. The increasing
importance of the Internet and the importance of delivering Web content quickly and
reliably have put strains on the network.
Broadband indicates a means of connectivity at a high or broad bandwidth,
which is capable of delivering multiple services simultaneously. It generally refers to
transmission of data over numerous frequencies. With the evolution of computer
networking and packet switching concept a new era of integrated communication has
emerged in the telecom world.
A concept of broadband services and the means of access technologies to bridge
the need of customer and service provider has emerged through out the world.
"Broadband" refers to high-speed Internet access. Traditionally, residential subscribers
have accessed the Internet by attaching a modem to their phone line and placing a local
call to their ISP. This "dial-up" or "narrow band" service has a number of constraints on
speed. Most commercial modems can achieve a maximum speed over the phone line of
56 kilobits per second (kbps).
Meaning of Broadband services
Broadband services are defined in various terms by different organization. Few of these
are given below: Original Bell System Definition
A broadband channel is a communications channel having a Bandwidth greater than a
voice-grade channel, and therefore capable of higher-speed data transmission.
1996 Telecom Reform Act

Broadband services are capable of carrying high-quality voice, data, graphics, & video.
CCITT definition
A service requiring transmission channels capable of supporting rates greater than 1.5
Mbps or primary rate in ISDN or T1/E1 in digital terminology.
Definition of broadband
Recognizing the potential of ubiquitous Broadband service in growth of GDP and
enhancement in quality of life through societal applications including tele-education,
tele-medicine, e-governance, entertainment as well as employment generation by way of
high speed access to information and web-based communication, Government have
finalised a policy to accelerate the growth of Broadband services. In India, DoT has
issued a Broadband policy in 2004. Keeping in view the present status, Broadband
connectivity is defined at present as: An always-on data connection that is able to support interactive services
including Internet access and has the capability of the minimum download speed of 256
kilo bits per second (kbps) to an individual subscriber from the Point Of Presence (POP)
of the service provider intending to provide Broadband service where multiple such
individual Broadband connections are aggregated and the subscriber is able to access
these interactive services including the Internet through this POP. The interactive
services will exclude any services for which a separate licence is specifically required, for
example, real-time voice transmission, except to the extent that it is presently permitted
under ISP licence with Internet Telephony.
Implementation of Broadband
To Strengthen Broadband Penetration, the Government of India has formulated the
Broadband Policy whose main objectives are to:Establish a regulatory framework for the carriage and the content of information
in the scenario of convergence.
Facilitate development of national infrastructure for an information based society.
Make available broadband interactive multimedia services to users in the public
network.
Provide high speed data and multimedia capability using new technologies to all
towns with a population greater than 2 lakhs.
Make available Internet services at panchayat (village) level for access to
information to provide product consultancy and marketing advice.
Deploy state of art and proven technologies to facilitate introduction of new
services.
Strengthen research and development efforts in the telecom technologies.

Need of Broadband
The Internet, e-mail, web sites, software downloads, file transfers: they are all now part
of the fabric of doing business. But until now, it has not been possible for businesses to
fully take advantage of the benefits that technology can truly deliver. The reason for this
is a simple one - a lack of bandwidth. Even for small businesses, narrowband dial-up
access is no longer sufficient. It simply takes too long to do basic tasks, like downloading
a large file, and is increasingly being recognized as insufficient and inconvenient.
Kim Maxwell in his book-"Residential Broadband: An Insider's Guide to the Battle for
the Last Mile" has grouped potential residential broadband applications into three general
categories: "professional activities " (activities related to users' employment),
"entertainment activities " (from game playing to movie watching), and "consumer
activities " (all other non-employment and non-entertainment activities). as follows:
Professional Activities:
Telecommuting (access to corporate networks and systems to support working at
home on a regular basis)
Video conferencing (one-to-one or multi-person video telephone calls)
Home-based business (including web serving, e-commerce with customers, and
other financial functions)
Home office (access to corporate networks and e-mail to supplement work at a
primary office location)
Entertainment Activities:
Web surfing (as today, but at higher speeds with more video content)
Video-on-demand (movies and rerun or delayed television shows)
Video games (interactive multi-player games)
Consumer Activities:
Shopping (as today, but at higher speeds with more video content)
Telemedicine (including remote doctor visits and remote medical analyses by
medical specialists)
Distance learning (including live and pre-recorded educational presentations)
Public services (including voting and electronic town hall meetings)
Information gathering (using the Web for non-entertainment purposes)
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Photography (editing, distributing, and displaying of digital photographs)

Video conferencing among friends and family
These applications have different bandwidth requirements, and some of them are still out
of reach today. For example, all of the "professional" activities will likely be supported
with less than 1.0 Mbps of bandwidth. Similarly, web surfing and home shopping will be
supported with less than 1.0 Mbps of bandwidth.
Movies and video, however, demand more bandwidth. Feature length movies can
probably be delivered with 1.5 Mbps of bandwidth, but broadcast quality video will
probably require more perhaps as much as 6.0 Mbps. Moreover, if high definition
television ("HDTV") is widely accepted as a new broadcast standard, that quality of
video would require almost 20.0 Mbps of bandwidth much higher than the current
broadband technologies will support. Thus, although the technology is moving toward
flexible, high-quality video-on-demand, the necessary speed is probably still more than a
few years away from becoming a reality.
The Internet is poised to spin off thousands of specialized broadband services. The access
network needs to provide the platform for delivery of these services. Following are the
various applications or services, which are very popular in society and needs broadband
connectivity: Virtual Networks
The private virtual networks (LAN/WAN) can be used in an ample variety of multimedia
services, like bank accounts and central offices.
Education by distance
Education will not have any limits to reach from source to destination. Along with the
traditional school a concept of remote leaning center is emerged out and popular for
various courses. There is no limit of distance, area or location in such distance learning.
The student situated in the remote station can intervene directly to his class with a double
system via videoconference, whilst this happens, simultaneously, the file ex change
Telework
Organization firm workers that incorporate communication systems via satellite, can
work remotely connecting directly to their head offices Internet by a high speed
connection that permits users to work efficiently and comfortable.
Telemedicine

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Doctors situated in different clinics can stay in contact and consult themselves directly to
other regional medical centers, using videoconference and the exchange of high quality
images, giving out test results and any type of information. Also rural zone can have the
opinion of specialists situated in remote hospitals quickly and efficiently.
Electronic commerce
Electronic commerce is a system that permits users to pay goods and services by Internet.
Thanks to this service, any person connected to the network can aquire such services with
independence from the place that he is situated and during the 24 hours, simply using a
portable computer.
Services on BB-Multiplay (Triple Play i.e. Voice, Data and Video)

TVOIP
1. TVOIP (also called as IPTV) delivers television programming to households via
broadband connection using Internet protocols.
2. Internet Protocol Television (IPTV) is expected to change the way people watch
TV. As the name suggests, IPTV is television programs delivered to subscribers
through the Internet
3. It requires a subscription and IPTV set-top box (STB).
4. IPTV is typically bundled with other services like Video on Demand (VOD),
Voice Over IP (VOIP) or digital Phone, and Web access.
5. IPTV viewers will have full control over functionality such as rewind, fastforward, pause, and so on.
6. IPTV (Internet Protocol Television) is a system where a digital television service
is delivered by using Internet Protocol over a network.
7. If you've ever watched a video clip on your computer, you've used an IPTV
system in its broadest sense.
8. For residential users, IPTV is provided with Video On Demand and may be
bundled with Internet services such as Web access and VoIP.
9. Microsoft is one of the many companies developing solutions to support the
Internet Protocol TV (IPTV) market.
10. IPTV is an emerging technology and will evolve into a completely interactive
experience in the future!

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11. First things first: the Set-Top Box (STB), on its way out in the cable world, will
make resurgence in IPTV systems.
12. The box will connect to the home DSL line and is responsible for reassembling
the packets into a video stream and then decoding the contents.
13. The video stream is broken up into IP packets and dumped into the core network,
which is a massive IP network that handles all sorts of other traffic (data, voice,
etc.)

VOIP
1. The technology used to transmit voice conversations over a data network using
the Internet Protocol.
2. A category of hardware and software that enables people to use the Internet as the
transmission medium for telephone calls.
3. VoIP works through sending voice information in digital form in packets,
4. VoIP also is referred to as Internet telephony, IP telephony, or Voice over the
Internet (VOI)
Benefits of VoIP
1. Cost reduction
a. Toll by-pass
b. WAN Cost Reduction
2. Operational Improvement
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a. Common network infrastructure

b. Simplification of Routing Administration Business Tool
3. Integration
c. Voice mail, email and fax mail integration
d. Web + Call
e. Mobility using IP

The concept of socio economy has an important role in the field of

communication of data, audio, video, speech or any other kind of application. It is an era
of CAPEX and OPEX. Service providers and customers both are interested in economy
with fastest tool of communication with more throughput. Traditional circuit switching
network are not supporting the effective fast communication for new applications.
Broadband has emerged out with the evolution of packet switching network.
Communication of data for various applications is feasible to carry with different
throughput. These applications have different bandwidth requirements and most of them
need more bandwidth.
Various technologies are available to service providers by which they can extend
the Broadband services to customers. These technologies are mainly classified under two
categories i.e. Wire line and Wireless technologies. Existing infrastructure used to access
telecom services is exploited for broadband for economical aspects and faster
development. DSL on copper loops, Optical fiber, cable TV are the popular technologies
for Broadband. World has also entered in the field of wireless to provide the broadband
through GSM, CDMA, LMDS, MMDS, WiMax and Wi-Fi.
The public sector will be one of the key drivers of broadband demand. Pooling
requirements from hospitals, schools etc. could permit more cost effective procurement
and stimulate broadband rollout.
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TECHNOLOGY OPTIONS FOR BROADBAND SERVICES

Broadband Access Technology

Broadband access technology is broadly classified into two categories. They are
Wired Line & Wireless which is further classified as given below. Wireline technologies
include traditional telephone lines, coaxial cable lines, and fiber optic lines. Wireless
communications involve cellular and fixed wireless technology, high speed short range
communications and satellite transmission.

Wireless

Wireline

3G Mobile

DSL (Digital Subs Line)

Wi-Fi (Wireless Fidelity)

Cable Modem

WiMAX

Optical Fibre Technologies

LMDS & MMDS

PLC(Power Line Communication)

FSO (Free Space Optics)

Satellite
Because physical infrastructure and geography are vastly different from country to
country, technology that works well in one geographic area may not work as well in
another. Therefore, it is up to each individual locality to determine the technologies that
best meet its needs. To handle the increasing bandwidth demand, localities are
considering upgrading their current telecommunications infrastructure or installing new
infrastructure. It is essential that communities and operators consider the present and
future needs of their citizens when examining the most appropriate systems. Broadband
should be considered an accelerator of economic development .
Factors affecting Broadband Access Choices
o Population density
o Existing infrastructure (e.g., twisted pair, cable, fiber)
o Government policies
o Competitive and regulatory dynamics
o Technology evolution

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GigE Fiber

MDU

City: High-rise
multi-family units

Homes

Suburbs:
Individual
single-family
units

Farm

Rural: isolated
single family unit

Cable
Twisted pair

Satellite

The broadband services reached to customer from the three providers. Basically these are
Service Provider, Network Provider and Access Provider. The role of Network Provider
is to provide the services offered to customer through the access extended by Access
Provider. There are various types of networks which are capable of transmitting and
managing the broadband traffic to desired nodes or locations.
Wireline access technology through DSL, Fiber, Cable etc., generally adopts:
IP based Network
ATM Network
Wireless access technology through Wi-Fi, Wi-MAX, 3G mobile etc provides wireless
access to ingress point of any core network and migrates to Internet world.
Broadband technologies used in Asian countries
Broadband technologies go through two stages of development in Asian countries. In the
early stage, sharp technological divisions exist among players due to regulatory
constraints. There are various mode of access used by service providers in this field.
Following was the beginning scenario in various countries like Hong Kong, Malaysia,
Indonesia, India and Singapore: Basic Telecom service providers adopted the use of ISDN/DSL
CATV operators use cable modems
Competitive players use wireless technologies.
In the later stage of development, technological divisions are shaped by geography and
infrastructure. The broadband started establishing and due to a progressive regulatory
framework it has matured in the market. In the countries like Korea and Philippines
service providers employ several technologies for the broadband in their networks.
DSL and cable modems are used where the PSTN and CATV are in place.

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Where rainfall is light, LMDS is used to serve densely populated areas with little
infrastructure and unwired business districts.
Satellite is used to service rural areas where population densities are low
Once newer technologies are available in the market, ISDN becomes relatively less
important. Established telephone companies are calculating the economics of converting
the Last Mile of existing networks to all-digital systems. Hong Kong and Singapore
citizens already have broadband access, such as movies on demand, through their local
telecom network. Cable-TV operators, too, are venturing into high-speed Internet access
through modified networks and end-user "cable modems." Advances in wireless
communications means that people starts surfing the net with cell phones at speeds
comparable to or greater than current home access.
The service provider converged voice and data network promises to be
implemented as nodes in a neighborhood or remote switches in regional locations.
Netwo
twork Architecture of Broad
oadband
Network architecture can be broadly classified into four categories:
1. Core or Backbone Network supporting multiple services of different QoS
implemented as packet switching nodes.
2. Aggregation or Distribution Network for extending the reachability to remote
locations and able to provide a cost effective solution to access the backbone
node.
3. Last mile connecting the subscriber Access Network
4. Home Network or Subscribers network.
Summary
There are tremendous changes in the telecommunication technologies. With the evolution
of Internet telecom world has merged rapidly in computer network.
Broadband Internet connections allow users to download web pages and data many times
faster than conventional 'narrowband' Internet access. Broadband services are 'always-on'
- the computer is connected to the Internet continuously. Users pay a flat rate independent
of how long they spend on the Internet or the amount of data downloaded. Broadband
users typically spend four times as long online as narrowband customers and broadband
take-up has been faster than many comparable technologies, competitiveness. Broadband
is needed in the present scenario due to new technologies and emerging out various types
of Data communication applications.

Digital Subscriber Line Technology

DSL is the family of technology, which transforms the narrowband Copper access
network into broadband. DSL does not refer to a physical line but the equipment, which
transforms the already existing media into a digital line. DSL has various family
members, which can be broadly classified as
Symmetric DSL
Provide identical data rates upstream & downstream
Asymmetric DSL
Provide relatively lower rates upstream but higher rates downstream.
DSL exploits the copper wires which have a much greater bandwidth or range of
frequencies than that demanded for voice without disturbing the line's ability to carry
phone conversations. The wires themselves can carry frequencies up to several million
Hertz.
There are several forms of digital subscriber lines, or xDSL with the x depending on the
particular variety of DSL. All xDSL connections use the same ordinary pair of twisted
copper wires that already carry phones calls among homes and businesses. Unlike cable
modem connections, which broadcast everyones cable signals to everyone on a cable
hub, xDSL is a point-to-point connection, unshared with others using the service.
The most common form of xDSL is ADSL. The A stands for asymmetric, meaning that
data transmission rate is not the same in both directions ie.,more bandwidth, or datacarrying capacity, is devoted to data traveling downstream-from the Internet to your PCthan to upstream data traveling from your PC to the Internet. The reason for the
imbalance is that, generally upstream traffic is very limited to a few words at a time, like
for example an URL request and downstream traffic, carrying graphics, multimedia, and
shareware program downloads needs the extra capacity. Downstream data moves at about
8Mbps for the most common forms of DSL.
Transmission rates depend on the quality of the phone line, the type of equipment it uses,
the distance from the PC to a phone company switching office, and the type of xDSL
being used.
Common types of DSL:
HDSL (High Data-rate Digital Subscriber Line)
HDSL is a better way of transmitting T1/E1(Primary rate as per American standard
(1.544Mb/s) & European standard (2.048Mb/s)) over copper wires using less

bandwidth without repeaters. Can be viewed as equivalent of PCM stream. It offers

the same bandwidth, both upstream and downstream. It can work up to a distance of
3.66 to 4.57 kms depending upon the speed required. When delivering 2048 kbps On
2 phone lines, each line carries 1168 kbps and On 3 phone lines, each line carries 784
kbps. No provision exists for voice because it uses the voice band. HDSL over single
phone line requires more aggressive modulation, works only shorter distance and
requires better phone line
SDSL (Single-Line Digital Subscriber Line)
SDSL is a single line version of HDSL that transmits T1/E1 signals over a single
twisted pair line. It can work up to 3.7 kms on 0.5 mm dia cable. However, SDSL
will not reach much beyond 10,000 feet while ADSL reaches rates above 6Mbps at
the same distance. SDSL is mainly used by small businesses. It does not allow to use
the phone at the same time but the speed of downloading and uploading is the same
RADSL (Rate-Adaptive Digital Subscriber Line)
RADSL is a variation of ADSL where the modem adjusts the speed of connection
depending on the length and quality of the line.
VDSL Very-high Data-rate DSL
Originally named VADSL (A Asymmetric) but was later extended to support both
symmetric & asymmetric. Requires one phone line and supports both voice & data. It
works between 0.3-1.37 kms depending on speed. Upstream data rate can be 1.6-2.3
Mbps. Downstream data rate is 13-52 Mbps
ADSL Asymmetric Digital Subscriber Line
ADSL is one of the number of access technologies that can be used to convert the
access line into a high speed digital link and to avoid overloading the circuit switched
PSTN. You can use the same phone line for Internet service at the same time its
carrying a voice call because the two signals use widely separated areas of the
frequency spectrum. A splitter next to your xDSL modem combines the lowfrequency voice signals and the higher-frequency data signals. A splitter on the other
end of the line breaks the voice and data signals apart again, sending voice calls into
the plain old telephone system (POTS) and computer data through high-speed lines to
the Internet.
An ADSL circuit connects an ADSL modem on each end of a twisted pair telephone
line, creating three information channels
1. A high speed downstream channel

2. A medium speed duplex channel

3. A basic telephone service channel
The basic telephone service channel is split off from the digital modem by filters, thus
guaranteeing uninterrupted basic telephone service, even if ADSL fails. To create
multiple channels, ADSL modems divide the available bandwidth of a telephone line
in one of two ways: frequency-division multiplexing (FDM) or echo cancellation, as
shown in Figure. FDM assigns one band for upstream data and another band for
downstream data. The downstream path is then divided by time-division multiplexing
into one or more high-speed channels and one or more low-speed channels. The
upstream path is also multiplexed into corresponding low-speed channels. Echo
cancellation assigns the upstream band to overlap the downstream, and separates the
two by means of local echo cancellation, a technique. With either technique, ADSL
splits off a 4-kHz region for basic telephone service using a splitter.
ADSL uses a different set of frequencies for data and does not interfere with
telephone conversation. Conversely, making a phone call while accessing the Internet
does not affect the speed of the ADSL connection.

There are two different standards for ADSL:

CAP (Carrier less Amplitude/Phase) which was used on the early installations of
ADSL.
DMT (Discrete MultiTone) which is the official ANSI standard.
CAP is well-understood and relatively inexpensive because it is a single-carrier
modulation technique that uses a wide passband and is susceptible to narrowband
interference. DMT uses multiple carriers. that uses many narrowband channels.
Therefore, DMT is capable of more speed than CAP. This is one reason that the ANSI
committee T1E1.4 accorded it standards status in document T1.413. This standard calls
for 256 sub bands of 4 KHz each, thereby occupying 1.024 GHz.
CAP divides the signals of the telephone line into three bands: voice, upstream channel
and downstream channel. Voice conversations are carried in the 0 to 4 KHz band as they
are in all POTS circuits. The upstream channel that carries data from the user to the
server is between 25 and 160 KHz. The downstream channel begins at 240KHz with a
maximum of 1.5 MHz which depends on a number of conditions such as distance, line
noise and number of users. CAP by keeping the three channels widely separated,
minimises the possibility of interference both between channels on one line and signals
on different lines.

DMT also operates by dividing signals into separate channels without using two quite
broad channels for upstream and downstream. The modulation technique that has become
standard for ADSL is called the Discrete Multitone Technique, which combines QAM
and FDM. In ADSL, the available bandwidth of 1.104 MHz is divided into 256 channels.
Each channel uses a bandwidth of 4.312 KHz. Each channel is 4KHz wide with a guard
band of .312KHz.. Hence the name Discrete Multitone.
Each sub carrier can support maximum15 number of bits. Depending on signal to noise
ratio for that sub carrier, a decision is taken as to how many bits that particular sub carrier

can support. Every channel is monitored and if the quality is low, the signal is shifted to
another channel. DMT constantly shifts signals between different channels, looking for
the best channels for transmission and reception. Moreover, some of the lower frequency
channels, are used as bi-directional channels for upstream and downstream. Keeping up
with the quality of all channels, monitoring and sorting the information on the bidirectional channels, makes the implementation of DMT more complex than CAP.
However, it provides more flexibility on lines of different quality.
Voice Channel 0 is reserved for voice communication.
Idle Channels 1 to 5 are not used, to allow a gap between voice and data
communication.
Upstream data and control. Channels 6 to 30 (25 channels) are used for upstream
data transfer and control. One channel is for control and 24 channels are for data
transfer. If there are 24 channels, each using 4KHz (out of 4.312KHz available)
with QAM modulation, we have 24x4000x15, or a 1.44Mbps bandwidth, in the
upstream direction.
Downstream data and control Channels 31 to 255 (225 channels) are used for
downstream data transfer and control. One channel is for control, and 224
channels are for data. If there are 224 channels, we can achieve upto 224 channels
are for data. If there are 224 channels, we can achieve upto 224 x 4000 x 15 or
13.4 Mbps.

This is the equivalent of 250 modems connected to your PC at once

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LOW-PASS FILTER
CAP and DMT are similar in one way as they both use frequencies above 4KHz. That is
why every ADSL user needs small filters to attach to the outlets that do not provide
signal to the ADSL modem. The low pass filter blocks all signals above 4KHz. The
reason is that all voice conversations take place below 4KHz and the low-pass filter
prevents the data signals from interfering with standard telephone calls. The basic
telephone service channel is split off from the digital modem by splitter at client site

ADSL characteristics:
1. Asymmetric ? The data can flow faster in one direction than the other. More
precisely, Data transmission is faster downstream (to the user) to the subscriber
than upstream (from the user). Customers do not need a high bi-directional
transmission speed. They actually connect to the internet in a relatively passive
mode because the amount of data they download is enormously higher than the
amount of data they transmitting.
2. Digital ? No type of communication is transferred in an analog method. All data is
purely digital, and only at the end, modulated to be carried over the line.
3. Subscriber Line ? The data is carried over a single twisted pair copper loop to the
subscriber premises
ADSL Architecture
Delivery of ADSL services requires a single copper pair configuration of a standard voice
circuit with an ADSL modem at each end of the line, splits the telephone line into three
information channels: a high speed downstream channel, a medium speed upstream
channel, and a Plain Old Telephone Service (POTS) channel for voice or an ISDN
channel.
The ADSL network components are:

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1.

The ADSL modem at the customer premises, that is called an ADSL transceiver
unit-remote (ATU-R). It provides local loop termination on the customer side.

2. The modem of the central office that is called an ADSL Transmission Unitcentral office (ATU-C). It terminates the ADSL local loop at the central office
premises.
3. DSLAM - DSL access multiplexer. Many ATU-C units are inserted into the
DSLAM. This unit can connect through an ATM or an ETHERNET access
network to the internet.
4. Splitter :- An electronic low pass filter that separates the analogue voice or ISDN
signal from ADSL data frequencies when they get to the subscriber premises. For
outgoing traffic, when they are transmitted from the subscriber premises, it
combines the voice and the data frequencies onto one line. This allows a POTS
phone connection to operate at the same time as ADSL digital data is transmitted
or received on the same line. One splitter is located at the central office and
another at the subscriber premises. The splitter at the central office can be
separate device or may be incorporated into the DSLAM.

Figure 1. ADSL Loop Architechture

The ADSL modem will combine the data stream from the computer, divide it into the
available aggregate channels, interleave, attach an ECC and then finally modulate the
data on the copper wires to be received by the demodulator at the other end. The
receiving end will perform error checking, correct the errors or request for re-transmit if
not possible. Interleaved blocks will be kept in the buffers until the entire blocks can be
reconstructed from subsequent frames.
Performance

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On the outside, ADSL looks simple and transparent, but on the inside there is a miracle of
modern technology.
In ADSL, downstream data rates can be between 1.5 Mbps and 8 Mbps, while upstream
data rates are between 16 Kbps and 832 Kbps. The minimum configuration provides 1.5
or 2.0 Mbps downstream and a 16 kbps duplex channel; others provide rates of 6.1 Mbps
and 64 kbps duplex. Products with downstream rates up to 8 Mbps and duplex rates up to
640 kbps are also available.
Known Problems & Solutions
The factors that can influence the downstream data rates are the length of the line, the
gauge of the line, presence of bridged taps and crosstalk from other wires in the same
time that cause noise. Line attenuation increases with line length and frequency, and
decreases as wire diameter increases. Also, some copper loops use different gauge wires
at different points and this can cause reflections in the signal, effectively attenuating
some frequencies. Ignoring bridged taps, ADSL will perform as follows:
Data Rate

Wire gauge Wire size Distance

1.5 or 2 Mbps 24 AWG

0.5 mm

5.5 km

1.5 or 2 Mbps 26 AWG

0.4 mm

4.6 km

6.1 Mbps

24 AWG

0.5 mm

3.7 km

1.5 or 2 Mbps 26 AWG

0.4 mm

2.7

Table 1. ADSL performance over different loop conditions

Some solutions to the problems that influence the data rate in ADSL:
1. Placing an optical network unit closer to the neighborhoods which are located too
far from the central office to obtain useful ADSL connection speeds. Data is
transmitted over fiber to the optical network unit, which then distributes the
signals to ADSL modems for connection to the residences.
2. Most ADSL providers select a series of speeds that are smaller than the maximum
so that more customers can obtain higher speeds, because the speed varies with
the length and the gauge of the line.
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3. ADSL modems incorporate forward error correction that dramatically reduces

errors caused by impulse noise. Error correction on a symbol-by-symbol basis
also reduces errors caused by continuous noise coupled into a line.
Framing
The data of the ADSL transport is organized into frames and superframes. One
superframe is built of 68 ADSL frames and one sync frame. The sync frame is used to
provide superframe synchronization, identifying the start of the superframe.
Because the ADSL lines can vary and are asymmetrical, the frame sizes may vary also.
Therefore, there are no absolute frame sizes for the superframe.
Each superframe must be sent every 17 ms and each frame must be sent every
250microseconds.
Each frame contains:
1. Fast Byte - used for special superframe-related processing functions.
2. Fast Data - transmits time sensitive information and it can vary in length.
3. Forward Error Correction (FEC) - is used to ensure the accuracy of the fast
data.
4. Interleaved Data - is the user data that the ADSL interface transmits, typically
internet data, and it can be vary in length.
ADSL DMT frame rate synchronized to the 4 kHz, and multiplexed into two data buffers
fast and interleaved. FEC is applied to each of the buffers.

Figure1. ADSL superframe

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Features of ADSL
Allows simultaneous access to the line by the telephone and the computer
In case of power/ADSL failure, data transmission is lost but basic
telephone service will be operational
ADSL Provides 16-640 kbps upstream and 1.5-9 Mbps downstream. It can
work up to a distance of 3.7 to 5.5 kms depending upon the speed required
Advantages of ADSL
You can leave your Internet connection open and still use the phone line
for voice calls.
The speed is much higher than a regular modem
DSL doesn't necessarily require new wiring; it can use the phone line you
already have.
The company that offers DSL will usually provide the modem as part of
the installation.
Variants of ADSL
THE ADSL2 FAMILY
As ADSL popularity grew, it begun spreading world-wide and clearly became the
carriers, service providers and subscribers choice of broadband media. Their feedback,
along with rising demand for improvements, contributed to the completion of a new
family of standards named ADSL2. These, beside offering higher rates, are more userfriendly for subscribers on one hand, and more profitable to carriers on the other hand. In
addition, most ADSL2's modems are backward compatible and support the simple
ADSL, making it easier to step into the next level when upgrading.
How can we choose the best configuration for a connection between a certain consumer
and the CO? Well, one of ADSL2's characteristics is being almost entirely automated,
and here is no different. A new feature called "Automode" enables service providers to
give their customers the optimal level of service, by analyzing the line condition and
gathering various information during connection between the CO and the customer
premise, and then choosing (automatically) the best line configuration.

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Figure 1. Automode in action.

The CO detects on initialization that the line to the CPE is too long for conditional
ADSL2/2+. Also, the consumer doesn't require extremely high speed services such as
HD video, and so automode decide to connect via RE-ADSL2 for optimum results.
ADSL2
The ADSL2 standard G.992.3 (and G.992.4 - the splitterless ADSL2 'lite' version) was
approved in October 02 by the ITU, and is looking like a promising next-generation
ADSL technology. The basic form of ADSL2 over POTS is also referred to as ADSL2
annex A, or simply ADSL2, and was one of its earliest member of the family. We now
discuss the new characteristics and main benefits of ADSL2 over its predecessor - the
ADSL:
Rate and Reach improvement
ADSL2 provides greater download stream (12Mbps compared to 8Mbps in
ADSL) due to improved modulation efficiency, and greater upstream (1Mbps).
Higher data rates are also gained due to improved modulation technique.
ADSL2 also presents several improvements in the initialization state machine,
which also helps avoiding line interference. This has also enabled to extend the
reach of ADSL2 in regard to ADSL by approx 600 feet (~185m), and to increase
data rates. Some of these improvements are listed in the next paragraph.
In addition, ADSL2 reduce framing overhead by allowing to determine the
number of overhead bits in range 4-32 kbps (compared to ADSL frames which
had a fixed 32kbps overhead per frame).
Also, ADSL2 achieves higher coding gain due to new framing algorithms (the
reed-solomon forward error correction code).
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Improved initialization -ADSL2 improves the the line initialization process, which is more automated than
ADSL, thus reducing error rate and increasing throughput. Improvements
includes:
Reducing near-end echo and crosstalk interferences at the binder.
Enabling RFI (radio frequency interference) cancellation techniques by disabling
tones during initialization.
Receiver allocated pilot tone helps avoiding bridged taps and RFI.
Receiver and Transmitter reach optimum in signal processing functions by
controlling the length of certain initializations states.
Better equalization with spectrum shaped init signals to improve channel
discovery.
Diagnostics
ADSL2's transceivers have been enhanced with Real-time performance-monitoring
capabilities to provide measurements of line noise, loop attenuation, and signal-to-noise
ratio (SNR) at both ends of the line, even when connection can not be achieved due to
poor quality of the line. This is of great value to both service providers which can use it
to track and prevent failures, and to carriers to determine if a customer qualifies for
higher data-rate services.
Power consumption
ADSL2 introduces two power management modes to help reduce power consumption:
the first is called "L2 low power mode" which enables power saving at the transceivers in
the central office (the ATU-C) by going into low power state whenever internet traffic is
decreased. The moment traffic has increased its detected by the transceiver which goes
back into full power mode (called L0).
The other power saving technique is called "L3 low power mode", and is used on both
CO's transceivers and the remote transceivers (the ATU-R), and basically enters the
transceiver into sleep mode when no traffic is detected on the connection. Returning into
full power mode requires re-initialization, but as we see shortly it is quite fast.
Fast startup A fast start-up mode reduces initialization time from about 10 seconds (in ADSL) to
approx 3 seconds.
ADSL2 Plus
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ADSL2Plus (also known as ADSL2+), has joined the ADSL2 family In January 2003 as
standard
g992.5
after
been
approved
by
the
ITU.
ADSL2+ doubles the downstream frequency band from 1.1Mhz in basic ADSL2, up to
2.2Mhz in ADSL2+, therefore increases the downstream data rate on shorter phone lines,
reaching 20Mbps on lines of max length of 5000ft (~1.5km). ADSL2+ upstream remains
1Mbps, depending of course on loop conditions.

Figure1: ADSL2+ compared to ADSL/ADSL2 in terms of bandwidth

ADSL2+ can also be used to reduce crosstalk. It has the ability to use only tones between
1.1Mhz-2.2Mhz, it can mask all downstream frequencies below 1.1Mhz. This comes in
handy when ADSL2+ services from a remote terminal co-exist with other ADSL services
(from the CO) and are sharing the same binder. Crosstalk interferences from the remote
terminal could be performance hazard for the ADSL services coming from the CO. The
masking of frequencies below 1.1 from the remote terminal eliminates the crosstalk
interferences almost completely.

Figure2: Reducing Crosstalk with ADSL2+

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RADSL
The Rate-adaptive ADSL is a non standard version of ADSL, in which the DSL modem
have the additional capability of adjusting bandwidth to the quality of the phone line not
only at the start of the connection but also at any time during the data transmission.
RADSL increases the maximum distance supported from 3.5 to around 5.5km (18000ft)
which makes it ideal for suburban neighborhoods. However, once again we pay the cost
of reduced data rate.

CABLE MODEM
Cable companies are now competing with telephone companies for the customer who
wants high-speed access to the Internet. DSL technology provides high-data-rate
connections for residential subscribers over the local loop. This imposes an upper limit on
the data rate. Cable TV operators use the cable TV network . In this section, we briefly
discuss this technology.
Traditional cable network: Cable TV network started to distribute broadcast video signal to locations with poor or no
reception in the late 1940s .It was called Community Antenna TV (CATV) because an
antenna at the top of a hill or building received the signals from the TV stations and
distributed them, via coaxial cables, to the community. Figure 1 shows a schematic
diagram of a traditional cable TV network.

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Traditional cable TV network

Figure 1

The cable TV office called the head end, receives video signals from broadcasting
stations and feed the signals into coaxial cables. The signals become weaker and weaker,
so amplifiers were installed through the network to amplify the signals. There could be
up to 35 amplifiers between the head end and the subscriber premises. At the other end,
splitters split the cable, and drop cables makes the connections to the subscriber premises.
The traditional cable TV system used coaxial cable end to end. Due to attenuation of the
signals and the use of a large number of amplifiers, communication in the traditional
network was unidirectional (one way). Video signals were transmitted downstream, from
the head end to the subscriber premises.
HFC Network
The second generation of cable network is called a hybrid fiber-coaxial (HFC) network.
The network uses a combination of fiber-optic and coaxial cable. The transmission
medium from the cable TV office to a box, called the fiber node, is optical fiber. From
the node through the neighborhood and into the subscriber premises is still coaxial cable.
The transmission medium from the cable TV office to a box, called the fiber node, is
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optical fiber; from the fiber node through the neighborhood and into the house is still
coaxial cable, figure shows a schematic diagram of an HFC network.
The regional cable head (RCH) normally serves up to 400,000 subscribers. The RCHs
feed the distribution hubs, each of which serves up to 40,000subscribers . The distribution
hubs plays an important role in the new infrastructure. Modulation and distribution of
signals are done here; the signals are then fed to the fiber node through fiber-optic cable.
The fiber node split into the analog signals so that the same signal is sent to the each
coaxial cable. Each coaxial cable serves up to 1000subscribers .the uses of fiber optic
cable reduce the need for amplifiers down eight or less.
One reason for moving from traditional to hybrid infrastructure is to make the cable
network bi-directional (two way).

Hybrid fiber-coaxial (HFC) network

Figure 2

Even in an HFC system, the last part of the network, from the fiber node to the subscriber
premises, is still coaxial cable has a bandwidth that ranges from 5 to 750 MHz
(approximately). The cable company has divided this bandwidth into three bands,: video,
downstream data, and upstream data, as shown in figure 3.
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Video Band
The downstream only video band occupies frequencies from 54 to 550 MHz. Since each
cable TV channels occupies 6MHz, this can accommodate more than 80 channels.

Division of coaxial cable band by CATV

Figure 3

Downstream Data Band

The downstream data (from the internet to the subscriber premises) occupies the upper
band , from 550 to 750 MHz .this band is also divided into 6-MHz channels .
Modulation

Downstream data are modulated using the 64 - QAM (or possibly 256QAM) modulation technique.

Data rate

There are 6bits for each baud in 64 QAM. One bit is used for forward
error correction; this leaves 5 bits of data per baud. The standard
specifies 1Hz for each baud; this means that theoretically, downstream
data can be received at 30 Mbps (5 bits /Hz x 6 MHz). The standard
specifies only 27 Mbps. However, since the cable modem is connected

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to the computer through a 10 base T cable, this limits the data rate to 10
Mbps.
Upstream Data Band
The up stream data (from the subscriber premises to the internet) occupies the lower
band, from 5 to 42MHz. This band is also divided into 6-MHz channels.
Modulation

The upper stream data used lower frequencies that are more susceptible
to noise and interference. For this reason, the QAM technique is not
suitable for this band. a better solution is QPSK.

Data rate

There are 2 bit for each baud in QPSK. The standard specifies 1 Hz for
each baud; this means that, theoretically, up stream data can be sent at
12 Mbps (2 bit/ Hz 6 Mbps). However, the data rate is usually less
than 12 Mbps.

Sharing
Both upstream and down band are shared by the subscribers.
Upstream sharing
The up stream data bandwidth is only 37 MHz. This means that there are only six 6-MHz
channels available in the upstream direction. A subscriber needs to use channels to send
data in the upstream direction. The question is, how can six channels be shared in an area
with 1000, 2000 or even 100,000 subscribers? The solution is timesharing. The
bandwidth is divided into channels using FDM; these channels must be shared between
subscribers in the same neighborhood. The cable provider allocates one channel,
statically for a group of subscribers. If one subscriber wanted to send data, she or he
contends for the channel with other who wants access; the subscriber must wait until the
channel is available.
Downstream sharing
We have a similar situation in the down stream .the downstream band has 33 channels of
6 MHz. a cable provider probably has more than 33 subscriber; there fore, each channel
must be shared between a group of subscribers. However the situations is different for the
downstream directions; here we have a multicasting situation .if there are data for any of
the subscriber in the group, the data are sent to that channels. Each subscriber is sent the
data. But since each subscriber also has an address registered with the provider, the cable
modem for the group matches the address carried with the data to the address assigned by
the provider. If the address matches, the data are kept; otherwise, they are discarded.

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CM AND CMTS
To use a cable network for data transmission, we need two key devices; a CM and a
CMTS.
CM
The cable modem (CM) is installed on the subscriber premises. it is similar to an ADSL
modem .figure 5 shows its location
CMTS
The cable modem transmission system (CMTS) is installed inside the distribution hub by
the cable company. It receives data from the Internet and passes them to the combiner,
which sends them to the subscriber. The CMTS also receives data from the subscriber
and passes them to the Internet.

Cable modem transmission system (CMTS)

Figure 4

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Cable modem (CM)

Figure 5
Data Transmission Schemes:
During the last few decades, several schemes have been designed to create a standard for
data transmission over an HFC network. Prevalent is one devised by Multimedia Cable
Network System (MCNS), called Data Over Cable System Interface Specification
(DOCSIS). DOCSIS defines all the protocols necessary to transport data from a CMTS to
a CM.
Up stream communication
The following is a very simplified version of the protocol defined by DOCSIS for
upstream communication. It describes the steps that must be followed by a CM.
1. The CM checks the downstream channels for a specific packet periodically sent
by the CMTS. The packet asks any new CM to announce itself on a specific
upstream channels.
2. The CMTS send a packet to the CM, defining its allocated downstream and
upstream channels.
3. The CM then starts a process, called ranging, which determine the distance
between the CM and CMTS. This process is required for synchronization between

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all CM s and CMTS s for the minislots used for timesharing of the upstream
channels.
4. The CM sends a packet to the ISP, asking for the Internet address.
5. The CM and CMTS then exchange some packets to establish security parameters,
which are need for public network such as cable TV.
6. The CM sends its unique identifier to the CMTS.
7. Upstream communication can start in the allocated upstream channel; the CM can
contend for the mini slots to sent data.
Downstream communication
In the downstream direction, the communication is much simpler. There is no contention
because there is only one sender. The CMTS sends the packet with the address of the
receiving CM, using the allocated downstream channel.
DIGITAL SUBSCRIBER LINE ACCESS MULTIPLEXER(DSLAM)
Introduction
To enable DSL technology, service providers must have a DSLAM located in their
networks to interact with the customer premises equipment (CPE) at the end user
location.
DSLAM is an integrated hardware and software system that allows the user to access
Broadband services as well as originate and terminate telephone calls over the same
single pair of copper wires
A Digital Subscriber Line Access Multiplexer (DSLAM) delivers exceptionally highspeed data transmission over existing copper telephone lines
A DSLAM separates the voice-frequency signals from the high-speed data traffic and
controls and routes digital subscriber line (xDSL) traffic between the subscriber's enduser equipment (router, modem, or network interface card [NIC]) and the network service
provider's network. A DSLAM takes connections from many customers and aggregates
them onto a single, high-capacity connection to the Internet. DSLAMs are generally
flexible and able to support multiple types of DSL in a single central office, and different
varieties of protocol and modulation, both CAP and DMT, in the same type of DSL. The
DSLAM may provide additional functions including routing or dynamic IP address
assignment for the customers.
The DSLAMs is in general be collocated with existing PSTN exchanges which provide
last mile access to customers over copper wire up to average span lengths of 3 kms.

Features of DSLAM
A digital subscriber line access multiplexer (DSLAM) delivers exceptionally high-speed
data transmission over existing copper telephone lines. A multiservice DSLAM is a
broadband-access network element (NE) that combines support for multiple DSL
transmission types. When coupled with high-capacity asynchronous transfer mode
(ATM) switching, multiservice DSLAMs deliver scalability, port density, and a
redundant architecture for reliability. Multiservice DSLAMs, together with various CPE
elements, can enable the relatively efficient deployment of broadband networks for highspeed Internet access as well as voice and video applications. The basic features of
Digital Subscriber Line Access Multiplexure (DSLAM) are describes below: DSLAM aggregates the subscriber lines
A Digital Subscriber Access Multiplexer delivers exceptionally high speed data
transmission over existing copper telephone lines
DSLAM separates Voice and Data of the Subscriber i.e. it separates the voice
frequency signal from High Speed data traffic
Routes and Controls Digital Subscriber Line (xDSL) traffic between the
subscribers end-user equipment (Router, Modem, or Network Interface Card
(NIC) and the Network Service Providers network.
Voice is given to the exchange switch
Data is fed to the IP Network through the LAN Switch
DSLAMs have been categorized in to 6 types based on no. of ports (480, 240,120,
64, 48 & 24) provided and planned for deployment based on the expected demand
DSLAM provides Access from 128Kbps to 8Mbps
DSLAM supports for QOS features such as Committed Access Rate between CPE
and DSLAM, Traffic Policing per port
DSLAM works Satisfactory without any degradation in performance and without
using any repeater/regenerator over a distance for various access speeds for
0.5mm copper pair.
Distance wise downstream bit rate in DSLAM

Downstream bit rate

Distance

6 Mbps

1.5 Kms

2 Mbps

3.5 Kms

1 Mbps

4.0 Kms

Implementation Of DSLAM
Broadband connectivity is extended to these DSLAM through the core network via the
LAN switch. Commonly it is available with 480, 240, 120, 64, 48 and 24 ports. DSLAMs
are generally aggregated through a Fast Ethernet or Gigabit Ethernet Interface. DSLAMs
are available with different types of access modules and capacities. The FX or GBIC
module in DSLAM and LAN switch should be capable of driving up to 10km on a single
mode fibre. The SX or GBIC module will support
Connectivity of DSLAM
DSLAM is connected to ATM or IP based core network through the networking
elements. It aggregates the data traffic of all the users provided to it and extends to core
network. The telephone traffic of each user is separated by splitter available in it and
transmits to PSTN network. DSLAM provides user access through user access layer and
Connectivity to IP backbone is provided through IP convergence layer.

247

CHAPTER - 13
INTELLIGENT NETWORK

Overview of Intelligent Network Architecture

Over the last thirty years, one of the major changes in the implementation of
Public Switched Telephone Networks (PSTNs) has been the migration from analogue to
digital switches. Coupled with this change has been the growth of intelligence in the
switching nodes. From a customer's and network provider's point of view this has meant
that new features could be offered and used.
Since the feature handling functionality was resident in the switches, the way in
which new features were introduced into the network was by introducing changes in all
the switches. This was time consuming and fraught with risk of malfunction because of
proprietary feature handling in the individual switches.
To overcome these constraints the Intelligent Network architecture was evolved
both as a network and service architecture.
In the IN architecture, the service logic and service control functions are taken out
of the individual switches and centralized in a special purpose computer. The interface
between the switches and the central computer is standardised. The switches utilize the
services of the specialized computer whenever a call involving a service feature is to be
handled. The call is switched according to the advice received by the requesting switch
from the computer. For normal call handling, the switches do not have to communicate
with the central computer.
Objectives of the Intelligent Network
The main objectives of the IN are the introduction and modification of new
services in a manner which leads to substantial reduction in lead times and hence
development costs, and to introduce more complex network functions.
An objective of IN is also to allow the inclusion of the additional capabilities and
flexibility to facilitate the provisioning of services independent of the underlying
network's details. Service independence allows the service providers to define their own
services independent of the basic call handling implementation of the network owner.
The key needs that are driving the implementation of IN are :
Rapid Service Deployment
Most business today require faster response from their suppliers, including
telecommunication operators. By separating the service logic from the underlying switch
call processing software, IN enables operator to provide new services much more rapidly.
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Reduced Deployment Risk

Prior to IN, the risk associated with the deployment of new services was
substantial. Major investments had to be made in developing the software for the services
and then deploying them in all of the switches.
With the service creation environment available, the IN services can be
prototyped, tested and accessed by multiple switches simultaneously. The validated
services can then be rolled out to other networks as well.
Cost Reduction
Because the IN services are designed from the beginning to be reusable, many
new services can be implemented by building on or modifying an existing service.
Reusability reduces the overall cost of developing services. Also, IN is an architecture
independent concept, i.e. it allows a network operator to choose suitable development
hardware without having to redevelop a service in the event that the network
configuration changes.
Customization
Prior to IN, due to complexity of switch based feature handling software, the
considerable time frame required for service development prevented the provider from
easily going back to redefine the service after the customer started to use it. With IN, the
process of modifying the service or customization of service for a specific customer is
much less expensive and time consuming.
The customization of services is further facilitated by the integration of advanced
peripherals in the IN through standard interfaces. Facilities such as voice response
system, customized announcements and text to speech converters lead to better call
completion rate and user-friendliness of the services.
IN Architecture
Building upon the discussion in the previous section, one can envisage that an IN
would consist of the following nodes :
Specialized computer system for holding service logic, feature control, service
creation, customer data, and service management.
Switching nodes for basic call handling.
Specialized resources node.
The physical realization of the various nodes and the functions inherent in them is
flexible. This accrues form the "open" nature of IN interfaces.
Let us now look at the nodes that are actually to be found in an IN
implementation.
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The service logic is concentrated in a central node called the Service Control
Point (SCP0.
The switch with basic call handling capability and modified call processing model
for querying the SCP is referred to as the Service Switching Point (SSP).
Intelligent Peripheral (IP) is also a central node and contains specialized resources
required for IN service call handling. It connects the requested resource towards a SSP
upon the advice of the SCP.
Service Management Point (SMP0 is the management node which manages
services logic, customers data and traffic and billing data. The concept of SMP was
introduced in order to prevent possible SCP malfunction due to on-the-fly service logic or
customer data modification. These are first validated at the SMP and then updated at the
SCP during lean traffic hours. The user interface to the SCP is thus via the SMP.
All the nodes communicate via standard interfaces at which protocols have been
defined by international standardization bodies. The distributed functional architecture,
which is evident from the above discussion, and the underlying physical entities are best
described in terms of layers or planes. The following sections are dedicated to the
discussion of the physical and functional planes.
Physical Plane
Service Switching Point (SSP)
The SSP serves as an access point for IN services. All IN services calls must first
be routed through the PSTN to the "nearest" SSP. The SSP identifies the incoming call as
an IN service call by analysing the initial digits (comprising the "Service Key") dialled by
the calling subscriber and launches a Transaction Capabilities Application Part (TCAP)
query to the SCP after suspending further call processing. When a TCAP response is
obtained from the SCP containing advice for further call processing, SSP resumes call
processing.
The interface between the SCP and the SSP is G.703 digital trunk. The MTR,
SCCP, TCAP and INAP protocols of the CCS7 protocol stack are defined in this
interface.
Service Control Point (SCP)
The SCP is a fault-tolerant online computer system. It communicates with the SSPs
and the IP for providing guidelines on handling IN service calls. The physical interface to

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the SSPs is G.703 digital trunk. It communicates with the IP via the requesting SSP for
connecting specialized resources.
SCP stores large amounts of data concerning the network, service logic, and the
IN customers. For this, secondary storage and I/O devices are supported. For more details
refer to the chapter on the "SCP Architecture".
As has been commented before, the service programs and the data at the SCP are
updated from the SMP.
Service Management Point (SMP)
The SMP, which is a computer system, is the front-end to the SCP and provides
the user interface. It is sometimes referred to as the Service Management System (SMS).
It updates the SCP with new data and programs (service logic) and collects statistics from
it. The SMP also enables the service subscriber to control his own service parameters via
a remote terminal connected through dial-up connection or X.25 PSPDN. This
modification is filtered or validated by the network operator before replicating it on the
SCP.
The SMP may contain the service creation environment as well. In that case the
new services are created and validated first on the SMP before downloading to the SCP.
One SMP may be used to manage more than one SCPs.
Intelligent Peripheral (IP)
The IP provides enhanced services to all the SSPs in an IN under the control of
the SCP. It is centralized since it is more economical for several users to share the
specialized resources available in the IP which may be too expensive to replicate in all
the SSPs. The following are examples of resources that may be provided by an IP:
Voice response system
Announcements
Voice mail boxes
Speech recognition system
Text-to-speech converters
The IP is switch based or is a specialized computer. It interfaces to the SSPs via
ISDN Primary Rate Interface or G.703 interface at which ISUP, INAP, TCAP, SCCP and
MTP protocols of the CCS7 protocol stack are defined.

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The IN architecture is depicted in Fig.1

Fig. 1
IN Architecture

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Distributed Functional Plane

Functional model of IN contains nine functional entities (FE's) which are
distributed over various physical entities (PE's) described in the previous section. A
functional entity is a set of unique functions. Brief description of the FE's is given below :
CCAF
Call Control Agent Function, gives users access to the network.
CCF
Call Control Function provides the basic facility for connecting the transport (e.g.
speech). It involves the basic switching function and trigger function for handling the
criteria relating to the use of IN.
SSF
Service Switching Function is used to switch calls based on the advice of the SCF
at the SCP. This function provides a service independent interface.
SCF
It contains the service logic components and advises the SSF at SSP on further call
handling.
SDF
Service Data Function contains the user related data and data internal to the network.
SRF
Specialized Resources Function covers all types of specialized resources other
than the connection resources that are in the exchange (e.g. recorded announcements,
tones, conference bridges, etc.).
SCEF
Service Creation Environment Function specifies, develops, tests and deploys the
services on the network.
SMAF
Service Management Access Function provides an interface between service
management function and the service manager who may be an operator.
SMF
Service Management Function enables a service to be deployed and used on IN.
Fig. 2 depicts the distribution and interconnection of the various functional entities.
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SMAF
SMF

SCEF
SCF

SDF
SRF

SSF

CCAF

SSF

CCF

CCF

CCF

CCAF

Management interface
In real time interface
Signaling circuit
Fig. 2 Distributed Functional Entities
The distribution of functional entities over the physical entities and their interconnection is summarized in Table 1 and 2 below. It may be noted that all the physical
entities may not be present in all INs as the choice of functional entities to be provisioned
is entirely up to the service provider.
Table 1
Distribution of FE's over PE's
Physical Entity
SSP
SCP
SMP
IP

Possible Functional Entities

CCF, SSF, CCAF
SCF, SDF
SCEF, SMF, SMAF
SRF

Table 2
FE-FE Relationship to PE-PE Relationship
FE-FE
SSF-SCF
SCF-SDF
SCF-SRF

PE-PE
SSP-SCP
SCP-SDP
SCP-IP
SCP-SSP-IP

SRF-SSF

SSP-IP

8.5 IN Services

Protocol
INAP, TCAP, SCCP and MTP
X.25 or Proprietary
INAP, TCAP, SCCP and MTP
ISUP, INAP, TCAP, SCCP and
MTP
ISUP and MTP

The IN services proposed to be introduced in Indian network have been derived from
ITU-T recommendations. Q.1211 (April 92). This document briefly gives the description
of 25 services mentioned in Capability set no. 1 (CS1) of above mentioned ITU-T
recommendations. CS1 basically deals with single ended services (which ITU-T calls as
Type-A services). Single needed services apply to only one party in the call.
(1)
ABD Abbreviated dialing
The subscriber can register a short dialing code and use the same for access to any PSTN
Number.
(2)

ACC Account Card Calling

A special telephone instrument is required.
User dials an access code and gets acceptance tone.
Then he dials a PIN (personal identification no.) code and dials the called
no. The Exchange reads the account number from card.
The Billing is debited to an account number (Telephone no.) as defined by
the card.
In another variation of the service, the account number can be given
through DTMF telephone instrument.
The follow-on feature facilitates the subscriber to dial another number
without disconnecting the call and without need to dial PIN and account
number again.

(3)

AAB Automatic Alternative Billing

Call can be initiated by any user and any instrument.
The call charges are billed in users account and that account need not be a
calling or a called party.
The user first dials access code.
Receives an announcement to dial account code and PIN (which is given
by management).
The account code and PIN are validated to check its correctness and
expired credit limit.
On getting acceptance tone the user dials the called number.
In another variation of the service, the called party may be billed based on
his concurrence.

(4)

CD Call Distribution
This service allows subscribers to have I/C calls routed to different
destinations according to allocation law specified by management (The
Subscriber has multiple installations).

Three types of laws exist :

-

Uniform load distribution

% Load distribution

Priority list distribution

In case of congestion or fault the alternative over flow is specified.

(5)

CFU Call Forwarding Unconditional

The subscriber can forward all incoming calls to a specified destination number.
Optionally an alerting ring/reminder ring can be given to the forwarding subscriber
whenever there is an incoming call.
(6)

CRD Call Rerouting Distribution

Calls are rerouted as per conditions encountered, e.g. busy or no reply
(time specified) or overload or call limiter.
Then as per selected condition the call is rerouted to predefined choice,
e.g. paper, vocal box, announcement or queue.

(7)
Completion of calls to busy subscriber
The service cannot be fully implemented with CSI capability since the status of called
party need to be known.
The calls are completed when subscriber who is busy becomes free.
On getting busy tone user dials a code.
The user disconnects.
On called party becoming free, call is made by the exchange first to
originating then to terminating subscriber (without any call attempt by the
user).
(8)
CON Conference Calling
The service cannot be fully implemented with CSI capability. In adding or dropping the
parties concerned it is not possible to check the authenticity of the parties. This service
requires a special transmission bridge to allow conversation among multiple subscribers.
CON-Add-ON-Conference Calling
User reserves the CON resources in advance indicating date, time of
conference and duration.
Controlled by user.
In active phase of conference parties can be added, deleted, isolated again
reattached or split the group of parties.
CON-Meet-ME Conference calling meet me
User reserve the resource same as 8A.

Each participant dials a special number at specific time (specified at the

time of booking of conference) and reach the conference bridge.
(9)

CCC Credit Card Calling

The Credit Card Calling service allows subscribers to place calls from any
normal access interface to any destination number and have the cost of
these calls charged to account specified by the CCC number.
A special instrument is not required. The caller has to dial card number
and PIN using DTMF instrument.
Follow-on feature may be provided optionally.

(10) DCR Destination Call Routing

The call is routed to destination pertaining to following conditions :
Time of day, day of week
Area of call originating
Calling identity of customer
Services attributes (non payment charges against subscriber)
Priority
Charge rates applicable for destination
Proportional routing of traffic
Optionally the subscribers can be provided with traffic details
(11)

FMD Follow me Diversion

A subscriber can remotely control the call forwarding capabilities.
It can be done from any point in the network using a password.
It is required if subscriber moves from place to place in a day.
The service subscriber will pay for diverted portion of the call.

(12)

FPH Free Phone

The called subscriber is charged for active phase of a call.
For the calling user, no charging is done.
The called subscriber can have multiple destinations and have DCR
facility.

(13)

MCI Malicious Call Indication

The subscriber requests the Administration to register his number for
MCI.

Administration registers the subscriber for MCI.

The called subscriber (who has registered this service) invokes the service
during the active phase of the call if he feels that the call is malicious.
The call is logged in the network with calling and called party number and
Date and time of invoking the service.
Optionally, the network can log unanswered calls also.
Optionally, the facility to HOLD the connection may be provided.
(14)

MAS Mass Calling

It involves high volume of traffic.
Calls can be routed to one or multiple destinations depending on
geographical location or time of day.
Mainly used in Televoting.
The network operator allots a service number.
The user dials this number to register his vote.
The user is played an announcement and asked to give his choice.
At the end of the service, the network operator provides the call details
and the count on various preferences.
After the service the same number can be reallocated to another
subscriber.
Calls made to this MAS number may be charged differently.

(15)

OCS Originating Call Screening

This helps subscriber to screen outgoing call as per day and time.
The screening list may be managed by subscriber.
The restriction of screening list may be override by PIN or password.
Three call cases are possible :

(16)

Call screened and allowed

Call screened and rejected

Call passed by using override option

PRM Premium Rate

The local call is charged at a higher (premium) rate.
This service is used by service providers for value added information
services, e.g. jobs, fortune, forecast, etc.
The revenue is shared between network operator and service provider.

The network operator allots a specific number to service provider, which

can be reached from any point in the network.
The provision exists for multiple site provider, in order to achieve
minimum expenditure on actual call.
(17)

SEC Security Screening

This capability allows security screening to be performed in the network
before an end user gains access to subscribers network, systems or
application.
It detects the invalid access attempts : how many, over what time period,
by whom and from where.
It provides an added layer of security.

(18)

SCF Selected Call Forwarding (Busy/Dont answer)

This facility is used for a group of 5 to 10 subscribers.
A list of SCF is prepared by a subscriber.
The list contains the choices as per conditions and calling subscribers of
the group.
A call from outside the group is forwarded to default telephone number.
The variation in SCF list can be done as per time of the day.

(19)

SPL Split Charging

It allows service subscriber to share the call charges with calling party on
per call basis.

(20)

VOT - Televoting
It is used to survey the public opinion by different agencies.
The network operator allocates a single telephone number to surveyor.
Each time user makes a call he can get access to televoting.
An announcement asks him to input further choice digits as per
preference.
As the user presses the digits the choice counter is incremented.
After voting is ceased the service subscriber is supplied the results.

(21)

TCS Terminating Call Screening

The incoming calls are screened as per screening list.
Calls are allowed as per list and time of the day.

(22)

UAN Universal Access Number

National number is published by the subscriber.
The subscriber may specify the incoming calls to be routed to number of
different destinations based on geographical locations of caller.

(23)

UPT Universal Personal Telecommunications

A universal number is defined.
Whenever subscriber changes the destination, he inputs that number from
telephone.
When a call comes, UPT number is translated to actual number.
This number can be accessed across various multiple networks, e.g.
mobile and fixed.
It can be accessed from any user network access.

(24)

UDR User Defined Routing

The user is allowed to define the routing of outgoing calls through
different network such as private, public, virtual or mixed network.
As per time of the day, for example the call is routed to either public or
private network whichever is cheaper.
For example, outstation calls can have different routes at different times of
the day.

(25)

VPN Virtual Private Network

A private network is built using public network resources.
A virtual PABX is created using different switches.
A PNP (private numbering plan) can be incorporated on those numbers.
Facilities such as CT, CH, dialed restrictions and other supplementary
services can be provided within the network.
Each line or user is assigned a class of service and specific rights in the
network.
To access the VPN from outside by one of VPN user, he is required to dial
a password.
Screening feature can be used to put restriction on outgoing and incoming
calls.
Call charges are assigned to VPN service subscriber.
Additional Account Codes are assigned to service subscriber to analyse
the cost line wise.

444

8.6 Charging
The IN services can be broadly divided into three categories for charging purposes :
-

No charging for calling user

Charging of calling user as per local call

Charging of calling user at higher rates

No charging for calling user : FPH, VCC and VPN services fall under this category.
Level 160 is free at present and is proposed to be allotted to such services. Local
exchanges need to analyse only 160 and route the call to SSP. This level has to be
created as charge free. New services of this type can be introduced in future without any
requirement of further modification in local exchanges
Charging of calling user as per local call : UN (local) falls under this category. Level
190 is free at present and is proposed to be allotted to such services. Local exchanges
need to analyse only 190 and route the call to SSP. This level has to be created as local
charge. New services of this type can be introduced in future without any requirement of
further modification in local exchanges.
Charging of calling user at higher rates : PRM and UN (long distance) falls under this
category. Since the charging is at higher rate it is proposed that prefix 0 may be used to
have barring facility. Level 090 may be used for such purpose. Local exchange will
analyse 090 and route the call to SSP. This level has to be created as charge on
junction pulses. New services of this type can be introduced in future without any
requirement of further modification in local exchanges.
The access code of various IN services as proposed is as follows :
No charging for calling user :
FPH

1600

VCC

1601

Password change for VCC

1602

VPN

1603

Charging of calling user as per local call :

UN (local)

1901

Televoting

1902

Charging of calling user at higher rates :

PRM

0900

UN (Long distance)

0901
************

555