UNIT- I

Introduction

Telecommunication networkscarry 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.

Network with point-to-point links among all the entities are known as fully

connected networks.

This network demands not only the telephone sets and the pairs of wires, but

also switching system or switching office or the exchange.

The switching system provides various services to the subscribers.

Evolution of Telecommunications

A network using point-to-point connections is shown in Fig. 1.

Fig.1 A network with point-to-point links.

In such a network, a calling subscriber chooses the appropriate link to

establish connection with the called subscriber. In order to draw the attention

of the called subscriber before information exchange can begin, some form of

signalling is required with each link.

If the called subscriber is. engaged, a suitable modification should be given to

the calling subscriber by means of signalling.

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In Fig.1 there are five entities and 10 point-to-point links. In a general case

with n entities, there are n(n - 1)/2 links. Let us consider the n entities in some

order.

In order to connect the first entity to all other entities, we require (n - 1) links.

With this, the second entity is already connected to the rrrst. We now need (n -

2) links to connect the second entity to the others. For the third entity, we need

(n - 3) links, for the fourth (n - 4) links, and so on. The total number of links,

L, works out as follows:

L = (n - 1) + (n - 2) + ... + 1 + 0 = n(n - 1)/2

Networks with point-to-point links among all the entities are known as fully

connected networks.

The number of links required in a fully connected network becomes very large

even with moderate values of n.

With the introduction of the switching systems, the subscribers are not

connected directly to one another; instead, they are connected to the switching

system as shown in Fig. When a subscriber wants to communicate with

another,

Fig.2 Subscriber interconnection using a switching system.

a connection is established between the two at the switching system. Fig.2 shows a

connection between subscriber S2 and S n-1.

In this configuration, only one link per subscriber is required between the subscriber

and the switching system, and the total number of such links is equal to the number of

subscribers connected to the system.

Signalling is now required to draw the attention of the switching system to establish

or release a connection.

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It should also enable the switching system to detect whether a called subscriber is

busy and, if so, indicate the same to the calling subscriber.

The functions performed by a switching system in establishing and releasing

connections are known as control functions

Early switching systems were manual and operator oriented.

Limitations of operator manned switching systems were quickly recognised and auto-

matic exchanges came into existence.

In electronic switching systems, the control functions are performed by a computer or

a processor. Hence, these systems are called stored program control (SPC) systems.

New facilities can be added to a SPC system by changing the control program.

In time division switching, sampled values of speech signals are transferred at fixed

intervals.

If the coded values are transferred during the same time interval from input to output,

the technique is called space switching.

If the values are stored and transferred to the output at a later time interval, the

technique is called time switching.

A time division digital switch may also be designed by using a combination of space

and time switching techniques. Figure. summarized as the classification of switching

systems.

Classification of Switching Systems.

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Simple Telephone Communication

In the simplest form of a telephone circuit, there is a one way communication

involving two entities, one receiving (listening) and the other transmitting

(talking).

This form of one way communication shown in Fig.3 is known as simplex

communication.

Microphone Earphone

Fig.3 A simplex telephone circuit.

The microphone and the earphone are the transducer elements of the telephone

communication system.

The theory of the carbon microphone indicates that the microphone functions like an

amplitude modulator. When the sound waves impinge on the diaphragm, the

instantaneous resistance of the microphone is given by

ri = ro - r sin wt

where

ro = quiescent resistance of the microphone when there is no speech signal

r = maximum variation in resistance offered by the carbon granules, r < ro

ri = instantaneous resistance.

The negative sign in above Equation indicates that when the carbon granules are

compressed the resistance decreases and vice versa.

Basics of Switching System

The switching office performs the following basic functions irrespective of the

system whether it is a manual or electromechanical or electronic switching system. Fig.

shows the simple signal exchange diagram

Identity. The local switching centre must react to a calling signal from calling subscriber and

must be able to receive information to identify the required destination terminal seize.

Addressing. The switching system must be able to identify the called subscriber from the

input information (train of pulses or multiple frequency depends on the dialing facility)

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Finding and path setup. Once the calling subscriber destination is identified and the called

subscriber is available, an accept signal is passed to the switching system and calling

subscriber. Based on the availability, suitable path will be selected.

Busy testing. If number dialled by the calling subscriber is wrong or the called subscriber is

busy (not attending the phone) or the terminal may be free (lifting the phone) but no response

(not willing to talk or children handling), a switching system has to pass a corresponding

voice message or busy tone after waiting for some time (status).

Supervision. Once the path is setup between calling and called subscriber, it should be

supervised in order to detect answer and clear down conditions and recording billing

information.

Clear down. When the established call is completed, the path setup should be disconnected.

If the calling subscriber keeps the phone down first, the signal called clear forward is passed

to the switching system

Billing. A switching system should have a mechanism to meter to count the number of units

made during the conversation.

Requirements of Switching System

All practical switching system should satisfy the following requirements for the

economic use of the equipments of the system and to provide efficient service to the

subscribers.

High availability. The telephone system must be very reliable. System reliability can

uptime is the total time that the system is operating satisfactorily and the down time is the

total time that is not. The availability is defined as

A = Uptime______

Uptime +down time

Also

A = ___MTBF_____

MTBF +MTTR

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where, MTBF = Mean time between failure

MTTR = Mean time to repair.

The unavailability of the system is given by

U = 1 – A < MTBF____

MTBF +MTTR

High speed. The switching speed should be high enough to make use of the

switching system efficiently. The speed of switching depends on how quickly the

control signals are transmitted.

Low down time. The down time is the total time the switching system is not

operating satisfactorily. The down time is low enough to have high availability.

Good facilities. A switching system must have various facilities to serve the

subscriber.

High security . To ensure satisfied or correct operation (i.e. providing path and

supervising the entire calls to pass necessary control signals) provision should be

provided in the switching system.

Manual Switching System

If a subscriber A initiates a call to the subscriber B, A lifts the telephone

handset from the cradle. This action, closes the subscribers loop which

includes transmitter and receiver of the handset.

The closing circuit causes a dc current (from battery) to flow through line

relay and illuminates the lamp of subscriber A. By seeing the light, the human

operator, closes the speak key and ask the subscriber A ‘‘number please’’. By

knowing the called subscriber is B, The operator throws ring key B to the

ringing generator.

Limitations:

Language dependent. The operation of a human exchange is language dependent as

the subscriber needs to communicate with the operator. In multilingual areas (big

towns, cities and tourist spots). This language dependency poses severe problems.

Lack of privacy . As a human operator is involving in connecting two subscribers, he

or she may be willing to hear the conversation of two VIP’s or record the message. So

in human exchanges, privacy is not possible.

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Switching delay. Before setting a path between two subscriber, the operator has to

monitor various signalling and if the operator is not active, the delay in switching will

be high normally it takes minutes to setup a call or release a call.

Limited service. An exchange can provide service only to minimum number of

subscriber. If the subscriber rate increases, overload and thus congestion are not

unexpected.

Dial Telephones

The mechanisms that transmit the identity (number) of the called subscriber are pulse

dialing (Rotary dial) telephone and multi frequency dialing (touch tone dial)

telephone.

Rotary dial telephone.

A rotary dial telephone is used for implementing the pulse dialing. In the pulse

dialing, a train of pulses is used to represent a digit of the subscriber number. The

basic idea is to interrupt the D.C. path of the subscriber’s loop for specific number of

short periods to indicate the number dialed.

This is called loop-disconnect (or rotary) signalling and in most countries the dial

operates at about ten impulses per second with a bread of 66 2/3 msec and made of

about 33 1/3 msec. Pulse dialing of digit 3 and 2 is shown in Fig.4

Fig.4 Rotary dial telephone

Fig. Finger plate arrangement

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Fig. Parts and mechanism

This method of dialing is slow. For nationwide and international dialing, the routing

signals to the switching center is fairly slow and inconvenient.

The method, which is replacing the rotary dial telephone, is the push button

telephone, which uses the multifrequency dialing.

Strowger Switching Components

The reasons for survival of this system even in some part of the world are its (a) high

system availability (b) comprehensibility and (c) cheapness and simplicity.

There are two types of basic elements which performs most of the functions of the

strowger switching system. They are (a) Uniselectors and (b) Two motion selectors.

Uniselectors.

A uniselector is a one which has a single rotary switch with a bank of contacts.

Depending upon the number of switching contacts, uniselectors are identified as 10

outlet or 24 outlet uniselectors.

A single 10 outlet or 24 outlet uniselector can be used as a switching element for 10

or 24 subscribers. Several uniselectors can be graded together so that multiple

incoming circuits can be connected to multiple outgoing circuits. Fig.5 shows the

simple arrangement of uniselectors.

The contact arm (wiper) moves across a fixed set of switch contacts. In the case

single uniselector, each contact is connected to an outgoing channel, so a caller can

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choose to connect to any of 10 different subscribers by energize any digit from 1 to

10.

As this selector moves in just one plane, thus sort of automated selector is known as

uniselector.

An uniselector is operated by (wiper movement) is performed by a drive mechanism

of a rotary switch. This mechanism contains an armature, electromagnet, Pawl, and

Ratched wheel.

Fig. 5 10 contact uniselector, (b) graded uniselectors.

Fig.5 Uniselectors

Two motion selectors.

A two motion selector is a selector in which a set of wipers is moved in two different

planes by means of separate mechanisms.

By mounting several arcs of outlets on top of each other, the number of outlets can be

increased significantly, but the wipers are then required to move both horizontally to

select a bank and then vertically to move around that bank to the required outlet. Such

a selector is known as a two motion selector. Fig.6 shows a typical two motion

selector arrangement.

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Fig.6 Two motion selectors

Actually there are 11 vertical positions and 11 horizontal contacts. The lowest vertical

position and first horizontal position in each vertical level are home position.

Step By Step Switching

The basic principle of strowger system is the direct application of the functional

subdivision with extensive use of third wire control. There is also an element of

shared switch network but without any common control.

In general, the strowger switching system consists of subscriber’s line circuit, line

finder & alloter circuit, Group selector and final selector. Fig.7 shows the block

diagram of strowger switching which explains the process by which the switching

system connects a calling subscriber and called subscriber.

Fig.7 Block Diagram of Strowger Switching

Subscriber line circuit (SLC).

Every subscriber is connected to his local exchange by one pair of wires. This single

pair carries the voice in both directions and the ring current to ring the bell when a

call is received.

At the exchange, every subscriber line terminates into its own subscriber line circuit

(SLC).

This consists of a pair of relays dedicated to that subscriber. If there are 1000

subscriber on that exchange, then there are 1000 SLCs.

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Line Finder & Alloter.

As there are many subscribers, but only a few selectors, there has to be a method for

finding a free selector and to connect the calling subscriber to that free selector.

To find a free selector, alloter switch is used for connecting calling subscriber and

selector line, selector hunter based access or line finder based access can be used.

In selector hunter based access, when a subscriber lifts his handset, the interrupter

mechanism in his selector hunter gets activated and the wiper steps to find free first

selector.

In line finder based access approach, the size is identified by interrupt mechanism.

Through the allotter switch, free line finder is identified.

Group Selector.

Depends on the subscriber number, the group selector may comprise one or two

selectros, generally referred as first and second selectors.

For 3 digit number, only one selector is required. For a 4 digit number, two selectors

are required. Let the called subscriber number is 5345.

When the subscriber dials the first number 5, the voltage level corresponding to ‘5’ is

represented by the sequence of 5 negative pulses as shown in fig 8.

The wiper on the row 5 of first selector rotates to find the free second selector. This

second stage selector responds to the second dialed digit.

Thus, for the number chosen by us, the wiper moves to the third row of the two

motion selector. This indicates that the subscribers with first two digits of 5 and 3 are

selected.

Fig. 8 Waveform generated by dialling 5345

Final selector.

The final selector takes care of the last two digits. As the last two digits being 4 and 5.

The dialing of 4 advances the switch to row 4 and then the dialing of 5, rotates the

switch to the 5th column. If the called subscriber line is free, then, the path setup is

completed.

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Otherwise a busy signal is returned to the caller. The final selector acts as an

expander, to connect the heavily loaded trunks to the much larger number of lightly

loaded customer lines.

As the path setup between calling and called subscriber is in response to the digits

dialed, the system is called the step by step system. It is also referred as a direct

controlled switching system because each switching stage will be under direct control

of the originating telephone’s dial.

As the strowger system provides dedicated path for the subscribers during

conversation, it may be referred to as space division technology. In practice, with 4-

digit numbering scheme, this switching system provides access to fewer than 10000

subscribers.

Disadvantages. The step by step system has the advantage of being inexpensive for small

system and highly reliable due to the distributed nature of equipment. However, the system

has several draw backs. Some of them are

As this switching involves heavy mechanical displacements, regular maintenance by

the skilled technicians are necessary.

It is not feasible to select an alternate route for interoffice calls, if all the trunks are

busy as the switching is by step through various selectors.

Step by step switching is limited to dial pulses. For touchtone telephones, special

devices have to be introduced between line finder and first selector to convert the

tones into dial pulses.

If calling rate is high, heavy operation is performed by the system and the life time of

the system is less.

The last two digits of the called line numbers are specifically determined by their

location on the connector. Congestion could arise when the switching system is

heavily loaded.

The strowger system can accept only 7 to 9 pulses in 1 second. Hence if we dial fast,

the system cannot give correct performance.

100 line switching system:

A 100 line switching system can serve up to 100 subscribers. A 100 line Strowger

switching system may be configured in a variety of ways. In this section we discuss

five different design alternatives for a 100 line step-by-step switching system.

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We then compare the design based on the design parameters. Simple line diagrams

known as trunking diagrams are used to represent the configurations of switching

systems.

For computing the cost of different designs, we assume that the cost of a uniselector

is one unit and that of two-motion selector is two units.

Design 1

An elementary configuration for a 100 line Strowger switching system using 10 outlet

uniselectors is shown in Fig. The configuration has two stages.

100 line switch using uniselectors

In the first stage, there are 100 uniselectors, one for each subscriber.

The second stage has 10 or more uniselectors. The second stage outlets are folded

back to the corresponding inlets via suitable control circuitry (not shown in figure).

Usually, each subscriber line is terminated on a relay group at the exchange. The

relay group contains all the necessary circuits for the control of the switching

mechanism.

Functions like testing, switching and return of the tones are done by the relay groups.

Similarly, outlets from the first stage are terminated on relay groups at the input of the

second stage.

The four banks of the uniselectors serve to provide positive, negative, P-wire and

homing connections. The corresponding outlets of all the first stage uniselectors are

commoned or multiplied.

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As a result, two calls destined for numbers in the range 50-59 cannot be put through

simultaneously, even though other uniselectors may be free in the second stage.

This problem may be overcome by making such an arrangement by which the

uniselectors in the second stage are treated as a common resource for all the

uniselectors in the first stage. The design parameters for this design are:

S=110,SC=10, K=2, TC=0.2, EUF=0.18, C=110, CCI=9.09 In this design,

blocking may occur under two conditions:

The calls are uniformly distributed, 10 calls are in progress and the 11th one arrives.

The calls are not uniformly distributed, a call is in progress and another call arrives,

which is destined for a number in the same decade.

The blocking probability PB in the first case is dependent upon the traffic statistics. If we

assume a random distribution of calls in the second case, we can calculate PB as

Probability that there is a call in a given decade = 10/100

Probability that another call is destined to the same decade but not to the same

number = 9/98

Therefore, PB = (1/10)(9/98)

= 0.009

Design 2

An alternative scheme which does not involve any logic circuit is to employ

10 uniselectprs in the second stage for every one uniselector in the first stage.

The total number of uniselectors in the system becomes 1100;100 in the first

stage and 1000 in the second stage. There are 10000 outlets and 100 inlets.

The corresponding outlets associated with every inlet are commoned.

It may be noted that unlike the previous design, this switching system is

nonblocking. The design parameters are:

S=1100, SC=50, K=2, TC=1, EUF=0.09, C=1100, CCI=4.54, PB=0

In step-by-step switching systems, the selection of one out of many selectors

in the next subsequent stage is done by deploying a uniselector or the

horizontal rotary mechanism of a two motion selector in a self-stepping mode

using the interrupter contacts. Design 4 and 5 discussed later in this section

use such arrangements.

Design 3

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Another way of realizing a 100-line strowger switching system is to use one

two-motion selector for each subscriber.

A subscriber is assigned a number in the range 00-99, and the same number is

used to identify the two-motion selector assigned to him. The 100 outlets of

each two-motion selector are numbered as per the scheme .

The corresponding outlets in all the 100 two motion selectors are commoned

and folded back to the corresponding inlets.

The two-motion selector used to establish a call is dependent upon the

initiator of the call.

Clearly, Design 3 is superior to designs 1 and 2. Further improvements to

design 3 are possible If the switching capability is provided to meet only the

estimated peak hour.

Fig. 100 line exchange with two motion selector

Design 4:

Instead of 100 two-motion selectors as in the case of Design 3, let us consider

only 24 two motion selectors.

In this case 24 simultaneous calls can be put through the switch. This can be

shared by all the hundred users.

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Typically a 24 outlet uniselector is used as a selector hunter. Each of the 24

outlets is connected to one two motion selector. Thus a subscriber has access

to all the 24 two motion selector in the system. This scheme is shown in fig.

Fig 100 line exchange with selector finders

The call establishment in this scheme takes place in two steps.

Firstly when the subscriber lifts his receiver handset, his uniselector hunts through the

contact positions and seizes a free two motion selector.

At the next step, the two motion selector responds to the dial pulses for appropriate

connection.

The design parameters of this system are :

S= 100 uniselectors + 24 two motion selectors.

SC= 24, K=2, TC= 0.48,

EUF= 0.58, C=148,CCI = 16.2

Design 5

When the start condition is received, the line finder is caused to hunt

vertically until the wipers reach a marked level.

The vertical hunting is then stopped and the horizontal hunt commences to

find a particular marked contact in that level.

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It may me noted that in the extreme case, the line finder may have to take 20

steps (10 vertical and 10 horizontal) before a line is found. The line finders are

made to step automatically, using interrupter contact mechanism.

When the line finder locates the subscriber line, the start condition is

removed, the allotter switch steps on the next free line finder in readiness for

further calls, and the numerical selector sends out the dial tone to the

subscriber in readiness to receive dialing pulses. Thereafter the establishment

of the connection proceeds in the usual manner. The design parameters are:

S=48, SC=24, K=1, TC=0.48, EUF=1, C=96, CCI=25.

1000 line blocking exchange

It is rare that exchanges with more than a few hundred subscribers are

designed to be nonblocking. Hence we only consider a blocking design for a

1000 line exchange in this section. Here the subscribers are identified by a 3

– digit number ranging from 000-999.

As explained in 100 line exchange, the final selector is a Strowger system

responds to the last two digits dialed by the subscriber. Hence for a 1000 line

exchange, we need one more selector stage preceding the final selector stage,

which would respond to the first digit of the called subscriber and establish a

connection to a final selector.

This is a group selector stage which uses two motion selectors as switching

elements. In addition to these two stages, we need either a selector hunter or a

line finder stage as a preselector stage.

The trunking diagram for a 1000 line exchange is given in Fig. as in the case

of 100 line exchange, when a subscriber lifts his receiver, the preselector

hunts for a free group selector. When a free group selector is obtained, the

subscriber is given the dial tone.

When the subscriber dials the first digit , the group selector steps up in the

vertical direction according to the digit dialled, and hunts for a free selector in

one of its 10 outlets . If a free selector is obtained, it responds to the next two

digits and a connection is established, otherwise an engaged tone is sent out to

the subscriber.

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100 line exchange with two motion line finders

Trunking diagram of 1000 line exchange

The design must cater to about 100-200 simultaneous calls by providing as

many final selectors. The final selectors are divided into 10 separate groups,

each group containing more than one final selector, and give access to a block

of 100 subscriber numbers.

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10000 line exchange

A 10000 line exchange has four stage: a pre selector stage, two group selector

stages and a final selector stage.

In order to support 10,000 subscribers, we need a minimum of 100 final

selectors. Since there are 100 blocks of 100 numbers each (0-99, 100-199,…)

in the number range 0-9999, the final selectors are placed in 100 separate

groups.

No grouping is required as far as the first group selectors are concerned.

Consider a design that can put through 1000 simultaneous calls. We need a

minimum of 1000 first group selectors, 1000 second group selectors and 1000

final selectors.

A subscriber has access to 24 first group selectors. Two hundered and forty

subscribers are terminated on each first group selector. There are 100 second

group selectors belonging to each level of the first group selector.

There are 10 final group selectors belonging to each level of the second group

selector. Each final selector is accessed by 100 second group selectors.

The design parameters are;

S=10000 uniselectors+3000 two motion selector

SC=1000, TC=0.2, K=4, EUF-0.3, C=16000, CCI-62.5.

Touch tone dial telephone.

The touch-tone dialing scheme is shown in Fig.8 the rotary dial is replaced by

keypad. This is called a dual tone multifrequency (DTMF) dial.

The principal method uses a pair of tones to signal each digit. These two tones are

shown in Fig.9

The central exchange decodes the different combinations of tones received into

equivalent dialed digit.

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Fig. 9 Touch tone dial telephone

The telephone set uses dialler IC and associated circuits. The dialer IC’s of 16 pin

HM 91C02A, 18 pin UM 91214B, 18 pin UM91214 E, 16 pin HT 9202G, 18 pin M

2560 G, 18 pin UM 1032 CP and 22 pin HM 91501, B are popular. Fig 3.5 (b) shows

the dialer IC HM 91C02A.

Cross Bar Switching

The basic idea of crossbar switching is to provide a: matrix of n x m sets of contacts

with only n + m activators or less to select one of the n x m sets of contacts.

This form of switching is also known as coordinate switching as the switching

contacts are arranged in a xy-plane. A diagrammatic representation of a cross point

switching matrix is shown in Fig. 10 . There is an array of horizontal and vertical

wires shown by solid lines.

A set of vertical and horizontal contact points are connected to these wires. The

contact points form pairs, each pair consisting of a bank of three or four horizontal

and a corresponding bank of vertical contact points.

A contact point pair acts as a cross point switch and remains separated or open

when not in use.

The contact points are mechanically mounted (and electrically insulated) on a set of

horizontal and vertical bars shown as dotted lines. The bars, in turn, are attached to

a set of electromagnets.

When an electromagnet, say in the horizontal direction, is energized, the bar

attached to it slightly rotates in such a way that the contact points attached to the bar

move closer to its facing contact points but do not actually make any contact.

This is prevented by introducing an energising sequence for latching the cross

points. A cross point latches only if the horizontal bar is energised fIrst and then the

vertical bar. (The sequence may well be that the vertical bar is energised fIrst and

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then the horizontal bar). Hence the crosspoint BC will not latch even though the

vertical bar C is energised as the proper sequence is not maintained.

In order to establish the connection B-E, the vertical bar E needs to be energised after

the horizontal bar is energised. In this case, the crosspoint AE may latch as the

horizontal bar A has already been energised for establishing the connection A.

This should also be avoided and is done by de-energising the horizontal bar A after

the crosspoint is latched and making a suitable. arrangement such that the latch is

maintained even though the energisation in the horizontal direction is withdrawn.

The cross point remains latched as long as the vertical bar E remains energised. As

the horizontal bar A is de-energised immediately after the crosspoint AC is latched,

the crosspoint AE does not latch when the vertical bar E is energised. Thus the

procedure for establishing a connection in a crossbar switch may be summarised as

energise horizontal bar energise vertical bar

energise vertical bar or energise horizontal bar

energise horizontal bar de-energise vertical bar

Fig. 10 3 x 3 crossbar switching

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Fig.11 6 ×6 cross bar matrix

Four Wire Concepts

The term four wire implies that there are two wires carrying the signals in one

direction and two wires carrying them in opposite direction.

In normal telephone service, the local loops are two wire circuits, on which a single

telephone call can be transmitted in both directions. If the distance between the

subscribers is substantial, the amplifiers (repeaters) are necessary to compensate the

attenuation. As the amplifiers are unidirectional, for two-way communication, four-

wire transmission is necessary.

The switching equipment in the local exchange and the line from subscriber to local

office (local loops) are two wire operation.

A four-wire circuit has amplifiers in its repeaters for each direction of transmission.

Fig 13. Block diagram of two to four wire conversion

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The two directions of transmission use different frequency bands so that they do not

interfere with each other. The two directions are separated in frequency rather than

space. At the toll office, the two wires are converted into four wire for long

transmission. A hybrid coil accomplishes this conversion.

Operation Of Hybrid

A simple block diagram and the hybrid coil arrangement of the four wire circuit is

shown in Fig 17. While connecting the two wire circuit to the four wire circuit, a loop

may be created and the signal could circulate round the loop, results in continuous

oscillation known as singing. The hybrid transformer (two cross connected

transformer) and balancing network together acts as a four wire/two wire terminating

set and eliminates the singing problem.

Hybrid circuits have been traditionally implemented with specially interconnected

transformer. More recently, however, electronic hybrids have been developed.

Fig 14. Hybrid transformer

Cross-connected transformer windings results in zero current in the line balance

impedance. The power thus divides equally between the input of the send amplifier and the

output of the receive amplifier, where it has no effect. The price of avoiding singing is thus 3

dB losses in each direction of transmission together with any loss in transformers.

Echo Suppressor / Echo Cancellor

A hybrid is required to convert a 2-wire circuit into a 4-wire circuit between the

subscriber and a digital exchange.

In analog exchanges, local calls are established on 2-wire circuits. But, long distance

calls require 2-wire to 4-wire conversion at the subscriber line-trunk interfaces

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Inter exchange or intercity trunk lines carry a number of conversations on a single

bearer circuit which may be derived from a coaxial cable, microwave or satellite

system.

Due to the long distances involved the bearer circuits need amplifiers or repeaters at

appropriate intervals to boost the signals. The amplifiers are almost invariably one-

way devices and cannot handle bidirectional signals.

Since the telephone conversation calls for signal transmission both ways, long

distance trunks require separate circuits for each direction, leading to 4-wire circuits.

Hence, the need for 2-wire to 4-wire conversion in long distance connections The

conversion is done by the hybrids.

Short delay echos are controlled by using attenuators and long delay ones by echo

suppressors or cancellers. CCITT recommends the use of echo suppressors or

cancellers if the round trip delay exceeds 50m seconds .

Use of echo suppressors is mandatory in satellite circuits as the round trip delay

involved is several hundred milliseconds.

For delays up to 50 ms, simple attenuators in the transmission path limit the loudness

of echo to a tolerable level,

The fig 14 below shows that the attenuation required increases as the delay increases

Fig.15 Echo as reflected signal

Attenuation vs Echo delay

If the echo path delay is 20 ms, ll-dB attenuators must be introduced in the

transmission paths.

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It may be noted that this loss must be accounted for in the overall transmission loss

budget, i.e. in ORE.

The operation of an echo suppressor is illustrated in Fig 16.

Echo J suppressors are voice activated attenuators. Normally, the echo suppressors

remain in a deactivated state, i.e. the attenuators are bypassed. Speech in one channel

activates the echo suppressor in the return path In Fig.16

Fig 16 : Echo suppressor operation

speech activates the echo suppressor EB and B speech EA. Fig 14. depicts the situation when

B is talking and A is silent. Should A attempt to talk at this juncture, his talk is also

attenuated. turn off the echo suppressor by interrupting B loudly.

The echo suppressor is deactivated automatically a few milliseconds after the talker

stops speaking

One drawback of echo suppressors is that they may clip the beginning portion of

speech segments. If subscriber A begins talking at the tail end of B's speech, his

talkspurt is not transmitted until the echo suppressors have had time to reverse

directions.

New designs of echo suppressors attempt to minimize the time required to reverse

directions. Typically reversal times are in the range of 2-5 ms

The operation of a system with echo suppressors is clearly half duplex When

telephone circuits are used for data transmission (see Section 10.1), full duplex

operation is required.

Moreover, with several milliseconds of interruption while an echo suppressor in one

direction is turned off and the one in the other direction is turned on, it is very

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difficult to organize data transmission. Hence, echo suppressors are usually disabled

while the circuits are used for data transmission.

This is done by providing a disabler feature in the echo suppressor and triggering the

same with a special signal. Usually, a 2025-Hz or 2100-Hz tone, transmitted for at

least duration of 300 ms with a signal level not .less than -5 dBm, is used to trigger

the disabler.

Once disabled, an echo suppressor remains so, as long as signals in the frequency

range of 300 to 3000 Hz are being transmitted in either direction. A no-signal

interval of 100 ms or more switches the echo suppressor back into the circuit

Recent developments in electronics technology have paved the way for a new form

of echo control by echo cancellation.

Echo cancellers do not physically insert attenuators or bypass them, instead they

process the incoming signal to eliminate the reflected component from it.

Transmitted speech is stored for a period of time equal to the round trip delay of the

circuit.

The stored signal is attenuated by a quantity equal to the loop loss and then

subtracted from the incoming signal. An integrated circuit echo canceller

incorporating the functions just described is now available for use on satellite

circuits.

It may be noted that echo cancellers eliminate speech clipping and permit full duplex

operation.

In the discussions so far, we considered only one reflection of the signal at the

listener's end being echoed at the talker's end.

This is referred to as 'talker echo'. If a second reflection takes place at the talker's end,

'listener echo' occurs. If repeated multiple reflections occur, oscillations are produced

and a condition known as 'singing' is said to exist.

Under this condition, the circuit is said to experience instability. Singing occurs if the

loop gain at some frequency is greater than unity.

If the loop gain is close to but less than unity, damped oscillations or near singing

conditions result. Singing or near singing conditions have a disturbing effect on both

the talker and the listener.

In general, the procedures used to control echos also control singing. But in some

short connections where no control is necessary for echos, singing may become a

problem.

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Singing can occur in idle circuits whereas echos cannot. To control singing under

such circumstances, the 4-wire circuits must have a certain minimum loss. CCITT

recommends a minimum loss of 10 dB on national networks to avoid singing.

The amount by which the reflected signal is attenuated is known as return loss. This is

given by

RL= 20 LOG (Z4+Z2 ) / (Z4+Z2) dB

Where

RL = return loss

z4 = impedance of the 4-wire circuit (or) in terms of power,

RL= 20 LOG (P4) / (P4+P2) dB

Where

P4 = incoming power on the 4-wire circuit

P2 = power reaching the 2-wire circuit

P4 -P2 = power reflected on to the return path

or , in terms of signal voltages,

RL= 20 log V4 / (V4 - V2)db.

= 20 log 1 / r c

Where r c is the reflection co-efficient defined as

r c= reflected signal / incident signal

If the two networks are perfectly balanced, then Z4= Z2 .therefore we have

RL (balanced ) = 20 log 2z2 / 0 = α

The return loss is infinite, i.e. the return signal experiences an infmite attenuation and

hence there is no reflected signal

SPC Exchange

In SPC (Stored Program Control), a programme or a set of instructions are stored in

its memory and executed automatically one by one by the processor.

Carrying out the exchange control functions through programs stored in the memory

of a computer led to the name stored program control.

A computer can be programmed to test the conditions of the inputs and last states and

decide on new outputs and states. The decisions are expressed as programs which can

be rewritten to modify or extend the functions of control system.

All switching systems manufactured for use as public switching systems now use

computers and software programming to control the switching of calls.

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Using SPC, 20 mA transmitter (old transmitter need 23 mA) with 52 V battery feed

and longer subscriber loop can be achieved. Basic view of SPC telephony switch. Fig

15. shows a basic control structure of a SPC telephony exchange.

The SPC uses processors designed to meet the various requirements of the exchange.

More than one processors are used for the reliability.

Normally these processors are duplicated. Also the SPC system uses distributed

software and hardware architectures. To carry over the maintenance functions of the

switching system, a separate processor is used.

Using the above setup, the SPC performs trunk routing to other control or tandem

offices. Special features and functions are also enabled with sophisticated equipments

and in compact form.

There are two types in SPC exchanges, namely centralised SPC and distributed SPC.

In the following sections, both types are described..

Fig 15. Basic control structure of SPC

Centralised SPC

Early electronic switching systems are centralised SPC exchanges and used a single

processor to perform the exchange functions.

Presently centralised exchanges uses dual processor for high reliability. All the

control equipments are replaced by the processors.

A dual processor architecture may be configured to operate in (a) standby mode (b)

synchronous duplex mode and (c) Load sharing mode.

Standby mode.

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In this mode, any one of the processors will be active and the rest is standby. The

standby processor is brought online only when the active processor fail.

This mode of exchange uses a secondary storage common to both processors.

The active processor copies the status of the system periodically and stores in axis secondary

storage.

In this mode the processors are not connected directly. In secondary storage,

programs and instructions related to the control functions, routine programs and other

required informations are stored.

Synchronous duplex mode .

In this mode, the processors p1 and p2 are connected together to exchange

instructions and controls. Instead of a secondary storage common to P1 and P2,

separate memory M1 and M2 are used.

These processors are coupled to exchange stored data. This mode of operation also

uses a comparator in between p1 and p2. The comparator compares the result of the

processors.

During normal operation, both processors receive all the information from the

exchange and receives related data from their memories. Although only one processor

actually controls the exchange and remaining is in synchronism with first one.

If a mismatch occurs, the fault is identified by the comparator, and the faulty

processor is identified by operating both individually. After the rectification of fault,

the processor is brought into service.

Fig 16 Centralized SPC Organisation

Load sharing mode.

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In this mode, the comparator is removed and alternatively an exclusion device (ED) is

used. The processor calls for ED to share the resources, so that both the processors do

not seek the same resource at the same time.

In this mode, both the processor are active simultaneously and share the resources of

exchange and the load dynamically.

If one processor fails, with the help of ED, the other processor takes over the entire

load of the exchange. Under normal operation, each processor handles one half of the

calls on a statistical basis.

However the exchange operator can vary the processor load for maintenance purpose.

Availability

Single processor. Availability A = ___MTBF____

MTBF+MTTR

where MTBF = Mean time between failures

MTTR = Mean time to repair

Unavailability = 1 – A

U = 1 – ___MTBF____

MTBF+MTTR

U = __MTTR____

MTBF+MTTR

If MTBF >> MTTR, U = MTTR

MTBF

Dual Processor. A dual processor system is said to have failed only when both processor

fails and the total system is unavailable. The MTBF of dual processor is given by

(MTBF)D = (MTBF) 2

2MTTR

where (MTBF)D = MTBF of dual processor

MTBF = MTBF single processor

Availability AD = (MTBF)D

MTTR + (MTBF)D

Substituting (MTBF)D in the above equation, we have

AD = (MTBF) 2 /2MTTR

MTTR +(MTBF) 2

2MTTR

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AD = ___(MTBF) 2 ____

(MTBF)2 + 2(MTTR)2

Unavailability U = 1 – AD = 1 – ___(MTBF) 2 ____

(MTBF)2 + 2(MTTR)2

=___2(MTTR) 2 ___

(MTBF)2 + 2(MTTR)2

If MTBF >> MTTR, UD = 2(MTTR) 2

(MTBF)2

Distributed SPC

The introduction of distributed SPC enabled customers to be provided with a wider

range of services than those available with centralised and electromechanical

switching system.

Instead of all processing being performed by a one or two processor in centralised

switching, functions are delegated to separate small processors (referred as regional

processors).

But central processors is still required to direct the regional processors and to perform

more complex tasks.

The distributed SPC offers better availability and reliability than the centralised SPC.

Entire exchange control functions may be decomposed either horizontally or vertically for

distributed processing.

In vertical decomposition, the exchange environment is divided into several blocks

and each block is assigned to a processor that performs all control functions related to that

block of equipments.

In horizontal decomposition, each processor performs one or some of the exchange

control functions. Figure shows the distributed control where switching equipment is divided

into parts, each of which have its own processor.

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Fig 17. Distributed SPC

Switching System Software Organization

For effective processing of a call, to perform various functions of subsystems

and to interface with the other subsystems, softwares plays a vital role.

The software programs enables any digital switching system input data, to

give outputs in a fraction of seconds, concurrent processing of many calls in

real time and performs many features other than simple pathset between

subscribers for conversation.

Need for Software

Other than call processing, any exchange is to serve the subscriber various facilities and

many administrative tasks. Fig. below shows various activities of a switching system. To

carry out these activities efficiently and effectively, the use of software is unavoidable.

Fig 18 various activities of digital switching system

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To perform the above tasks, a large amount of software is required. However, the

software for basic functions are must and remaining services are optional and requires

software depends on the location of switching systems. Approximately 70% of the total

software is used to perform basic functions. Only 0.1% of the total processing time is used by

the 30% of the remaining service oriented software packages.

Software Classification and Interfacing

Classification. At various levels of hardware architecture, the softwares

are used. Thus, many digital switching systems employ some system level software. Basic

software systems are classfied as :

Maintenance software

Call processing software

Database/Administration software

Feature software.

Above software packages are divided into program modules. Each module dealing with

specific task. Several molules are grouped together to form functional units. These factors

include the requirements of the business, the location of telephone exchanges, customer

needs, internal requirements,and parameterised design.

The parameterised design includes hardware parameter and software

parameters. The hardware parameter are based on the hardwares used in the central

office or exchanges.

They are number of network control processors, number of line controllers, number

of subscribers to be serviced, number of trunks for which the exchange is engineered

etc.

Some examples of software parameters are the registers associated with number and

size of automatic message accounting (AMA) registers, number and size

of buffers for various telephony function and various features to be included for

that particular exchanges.

Thus, the parameterised design helps in designing software common to the similar

types of exchanges.

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Maintenance software

There are various activities and tests involved to maintain a switching system. Some of

them are :

Supervision of the proper functioning of the exchange equipment, trunks and

subscriber lines.

Monitoring the database of line and trunk assignments.

Efforts for the system recovery in case of failure.

Automatic line tests, which permits maintenance persons to attend several exchanges

from one control location.

Effective diagonestic programs and maintenance strategies used to reduce

the maintenance cost.

The root cause of the failure of any digital switching system is related to the software

bugs which affects the memory and program loops, hardware failures, failure to

identify the exact problem of failure and atleast but not least the human error.

Thus, the first step in software build is to select the appropriate program modules

which are suitable for the switching system. The points to be considered are types of

lines, location of switching system, signalling systems, availability of skilled person

or the level of diagonisation.

Preventive maintenance programs are activated during the normal traffic. If a fault

occurs, the OS activates the maintenance program to recover the system.

Effective preventive and maintenance programs and strategies help in proper

maintenance of digital switching system with reduced maintenance cost.

Call processing software. The call processing functions are controlled by a central

processor. Other functions carried out by the central processor are maintenance

and administration, signalling, network control. T

Thus, the call processing programs are usually responsible for call processing and to

interface with the translation data, office data, and automatic message acounting and

maintenance programs.

The translation data is the type of data generated by telephone companies

related to subscriber. T

The office data is related to a particular digital switch. The call processing programs

can be derived from state-transition diagrams in specification and description

language (SDL). An interpreter programs is written to access the lists and tables and

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to process the call by interpreting the data within them. Fig. 19 shows three levels of

call processing program. But it varies depends on the digital switching system.

Fig 19. Call processing software levels

Data base/Administration software

For administration and data base management, large amount of software required. But

these tasks are performed infrequently, it uses less than 5% of the total processing time. The

administration tasks includes

Alarm processing

Traffic recording

Change of numbers or area codes corresponding to the change in subscriber Rate and

government policy.

Changing routing and routing codes. This decisions made on the traffic intensity of a

particular exchange.

Generation of exchange management statistics.

Most digital switching system employ a data base system to :

Record office information

Billing information

Software and hardware parameters

System recovery parameters

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System diagnostics.

Switching software’s

Software’s for digital switching systems are written in high level languages.

Early electronic switching systems used assembly language programmes.

This language is known as CCITT high level language (CHILL). It has three

major features as data structure, program structure and action statements. It is

designed for the various SPC modules discussed earlier.

Software codes for digital switching systems are also written in high level programming

languages such as C, C ++, PASCAL

Enhanced Services Offered By SPC

One of the immediate benefits of stored program control is that a host of new or

improved services can be made available to the subscribers. Over a hundred new services

have already been listed by different agencies like CCITT, and the list is growing day by day.

Although there are a large number of services, they may be grouped under four broad

categories:

Services associated with the calling subscriber and designed to reduce the time spent

on dialling and the number of dialling errors

Services associated with the called subscriber and designed to increase the call

completion rate

Services involving more than two parties

Miscellaneous services.

These new services are known as supplementary services and some of the prominent

ones are as follows::

Operator answer:

Category 1:

Abbreviated dialling

Recorded number calls or no dialling calls

Call back when free.

Category 2:

Call forwarding

Category 3:

Call number record

Call waiting

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Consultation hold

Conference calls

Category 4:

Automatic alarm

STD barring

Malicious call tracing

A subscriber issues commands to an exchange to activate or deactivate a service,

record or clear data in the subscriber line data area or solicit an acknowledgement

from the exchange. The general command syntax is shown in Fig.

Fig. Syntax of user commands

The command code is usually a 2-digit number. Enhanced services are in

general, available only to subscribers with DTMF push button telephone. The

two push buttons with symbols * and # are used extensively for

communicating subscriber commands to the exchange.

For easy handling of the commands at the exchange end, the commands are

placed under four groups as shown in table which also gives the most

popularly used command formats with and without data fir each group.

To make abbreviated dialling as simple as possible, neither a symbol is used

to indicate the beginning of the command nor is there any command code is

used in this case

Other enhanced features are :

Abbreviated dialling

Recorded number calls or no dialling calls –hot line calls

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Redialling or repeat dialling

Call back when free

Call forwarding

Operator answer service and

Calling number record record feature

Space Division Switching

The fundamental operation of a switch is to setup and release connection

between subscribers. It involves direct connection between subscriber loops at

an end office or between station loopsat a PBX.

The switches are hardware and/or software devices capable of creating

temporary connections between two or more subscribers.

A cross point switch is referred to as a space division switch because it moves

a bit stream from one circuit/bus to another. For large group of outlets,

considerable savings in total cross points can be achieved if each inlet can

access only a limited number of outlets. Such situation is called limited

availability.

By overlapping the available outlet groups for various inlet groups, a

technique called ‘‘grading’’ as established.

Rectangular cross point array is an example of grading. For longer trunk

groups, large cross points were expensive and not used now-a-days. The

number of cross points required are M × N, where M is number of inlets and

N is number of outlets.

A basic requirement for constructing switching systems (or telephone

exchanges) is to design switching networks having greater number of outlets

than the switches from which they are built. This can be done by connecting a

number of switching stages in tandem.

Single Stage Network

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M

N M N

Switch symbols

Single stage network can also be constructed by multiplying banks of M

uniselectors or one level of a group of M two-motion selectors each having N

outlets.

Number of simultaneous connections made is M (if M < N) and N (if N < M).

The switch contains MN cross points. If M=N, the number of cross points is C1

= N2. Thus cost (number of cross points) increases as the square of size of

switch. Whereas efficiency (proportion of cross points) N/N2 = 1/N decreases

inversely with N.

It is uneconomic to use single stage network for large number of inlets and

outlets. E.g. a switch with 100 inlets and outlets requires 10000 cross points.

Only 1 pc of these can be in use at any time.

Switches for making connections between large numbers of trunks are therefore

constructed as networks containing several stages of switches.

Operation of cross point at co-ordinates (j, k) to connect inlet j to outlet k thus

performs the same function as operating cross point (k, j) to connect inlet k to

outlet j. So half the cross points are redundant and thus can be eliminated.

This results in the triangular cross point matrix shown in figure.

Triangular crosspoint matrix for connecting both way trunks

Number of crosspoints required is C1 = ½ N (N-1)

Triangular switches are not found in telephone switching systems because both

way trunks are not used.

Link system:

Grade of service of a link system depends on the way the system is used. We may

classify these uses as follows:

Mode 1: Connection is required to one particular free outgoing trunk

Mode 2: Connection is required to a particular outgoing route but any free trunk on that

route may be used

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Mode 3: Connection may be made to any free outgoing trunk

Concentrator operates in mode 3, Route switch operates in mode 2 and Expander

operates in mode 1.

Two Stage Networks

N incoming trunks

N outgoing trunks

Primary switches have n inlets

Secondary switches have n outlets

So number of primary switches or secondary switches (g)= number of outlets per primary or

secondary Switch g = N/n

Number of crosspoints per primary switch = number of crosspoints per secondary

switch = gn = N

Total number of crosspoints (C2) = number of switches x crosspoints per switch C2 = 2gN

= 2N2/n

Number of links = number of primary switches x number of secondary switches

= g x g

= g2

= (N/n)2

Number of crosspoints varies as 1/n ; number of links varies as 1/n2

If n is large, to make crosspoints less, then number of links will decrease.

Let number of links be equal to the number of incoming and outgoing trunks

i.e. g2 = N substitute g=N/n

then N2/n2 = N

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g2 links

g primary switches g secondary switches

N outgoing trunks

N incoming

trunks

or n = √N

Total number of crosspoints is C2 = 2N2/ √N = 2 N3/2

The 2 stage network shown has the same number of incoming trunks and outgoing trunks.

A concentrator has more incoming trunks than outgoing trunks. An expander

has more outgoing trunks than incoming trunks. Consider a concentrator with M

incoming trunks and N outgoing trunks (M>N) Each primary switch has m

inlets and each secondary switch has n outlets.

Then

Number of primary switches = M/m

Number of secondary switches = N/n

Number of crosspoints per primary switch = mN/n

Number of crosspoints per secondary switch = nM/m

Total crosspoints = M/m (mN/n) + N/n (nM/m)

= MN (1/n + 1/m)

Total number of links = number of primary switches x number of secondary switches

= M/m x N/n

= MN/mn

Since the traffic capacity is limited by the number of outgoing trunks, there is no point

in providing more links more links than N.

Therefore MN/mn = N

Or n = M/n

Total number of crosspoints C2 = MN (m/M +1/m)

To minimize C2, dC2/dm = MN (1/M – 1/m2) = 0

Solving we get m = √M

But n = M/m = M/ √M =√M =m

Therefore n =m

Thus number of crosspoints is minimum when number of inlets per primary switch is

equal to number of outlets per secondary switch.

C2 = MN (1/n + 1/m)

= MN ( 1/√m + 1/√m)

= MN 2/√m

= 2N√m

C2 = 2M1/2 N

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m and n must be integers and factors of M and N respectively. Now if n = √N, then m =

M/√N

Three Stage Networks

There are total of N/n primary and tertiary switches. There is one link from each

primary to each secondary switch and so is the case between secondary and

tertiary switches. Any inlet on a primary switch has g2 alternate paths

(secondary switches to reach tertiary switches).

If the three stage has N incoming trunks and N outgoing trunks and has primary

switches with n inlets and tertiary switches with n outlets, then:

Number of primary switches (g1) = number of tertiary switches(g3) = N/n

Therefore, secondary switches have N/n inlets and outlets.

If the no. of primary-secondary links (A links) and secondary-tertiary (B links) are each

N, then the number of secondary switches is

g2 = N ÷ (N/n) = n = no. of outlets per primary switch = no. of inlets per tertiary switch

No. of crosspoints in primary stage = n2(N/n) = nN

No. of crosspoints in secondary stage = n (N/n)2 = N2/n

No. of crosspoints per tertiary stage = n2(N/n) = nN

and the total number of crosspoints = C3 = N(2n + N/n)

By differentiating the above equation with respect to n and equating to zero, it

can be shown that the number of crosspoints is a minimum when

n= √(N/2)

and then C3 = 2√2 N 3/2 = √2 C2 = 23/2N-1/2C1

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A links B links

g2 secondary switches

g3 tertiary

switches

g1 primary switches

If a three stage concentrator has M incoming trunks and N outgoing trunks

(M>N), its primary switches each have m inlets and its tertiary switches each

have n outlets, then:

No. of primary switches = M/m

No. of tertiary switches = N/n

If there are g2 secondary switches, then:

Crosspoints per primary switch = m g2

Crosspoints per secondary switch = M/m x N/n

Crosspoints per tertiary switch = g2n

The total number of crosspoints is C3 = M/m x mg2 + g2 x MN/mn + N/n x g2n

= g2 [M + N + MN/mn]

Since M>N, let no. of A links = no. of B links = N

Therefore N = g2M/m = g2N/n

Hence g2= n and m = n M/N

Therefore C3 = (M + N)n + N2/n

Differentiating with respect to n to find a minimum gives

m = M/√(M+N), n=N/√(M+N)

C3= 2N/√ (N+M)

To obtain an expander, M is exchanged with N and m with n.

Four Stage Networks

A four stage network can be constructed by considering a complete

two stage network as a single switch and then forming a larger two stage array

from such switches.

If a four stage network is constructed with N incoming and N

outgoing trunks, with switches of size n x n, then N = n3 and the total number of

switches is 4n2. thus the total number of crosspoints is

C4 = 4n2.n2

= 4 N4/3

The number of crosspoints per incoming trunk is 4N1/3.

Multi Stage Network

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An arrangement referred to as a multiple is used in connecting circuits to

provide the various connections. If all the relays have the contact terminals on

one side multiplied (duplicated) to a set of common leads, then the arrangement

is called a ‘multiple’.

Link

Two multiples may be joined back to back by a ‘common link’.

A circuit may reach the link by means of a multiple on a set of relays and then

reach any one of the B circuits by a similar multiple on a second set of relays.

Thus link principle provides a means of positioning the amount of switching

equipment according to the demand for simultaneous connections between two

groups of terminal circuits.

Grading is obtained by the partial multiplying of the outlets of connecting networks

when each network provides only limited availability to the outgoing group of trunks.

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Relay i/ps Relays Relay o/ps

Link

Multiples

Concentration Distribution Expansion

Calling subscriber Called subscriber

Other forms of grading:

Homogenous grading

Progressive grading

O’Dell grading

Transposed grading

Grade of service = ratio of number of lost calls to number of calls attempted

= probability of finding all the circuits engaged

Grade Of Service

Two-Stage Network

For a two stage network, let occupancy of the links be a and the occupancy of

the trunks be b.

For mode 1 (i.e. connection to a particular outgoing trunk) only one link can be

used. The probability of this being busy is a and this is the probability of loss.

For mode 2(i.e. connection to an outgoing route with one trunk on each

secondary switch) any free link can be used. The probability of using a

particular link is

1 – probability that both link and trunk are free

= 1 – (1-a)(1-b)

But there are g paths available. Assuming that each being blocked is an independent

random event, the probability of simultaneous blocking for all g paths is:

B2 = [1-(1-a)(1-b)]g

= [a + (1-a)b]g

where g is the number of secondary switches.

The number of incoming trunks is then much larger than the number of

outgoing trunks, so the grade of service is given by

B3 = E1,N (A) where A is the total traffic offered.

Three stage network

Let occupancy of links be ‘a’. Let occupancy of outgoing trunks be ‘b’.

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Occupancy measures the extent to which a stage in a multistage network is

occupied or is busy.

Erlang measure indicates the average no. of servers and therefore,

Occupancy period = total traffic ÷ no. of servers of links

For mode 1, the choice of a secondary switch determines the A and B links.

Probability that both links are free = (1-a)(1-b)

Probability of blocking = 1 – (1-a)(1-b)

However there are g2 secondary switches,

Probability that all g2 independent paths is simultaneously blocked is

B1 = [1-(1-a)(1-b)]g2

= [a + (1-a)b]g2

For mode 2,

Probability of blocking for a particular trunk

= 1 – (1-B1)(1-c)

= B1 + (1-B1)c

Therefore probability of simultaneous blocking for all g3 independent paths is

B2 = [B1 + c(1-B1)]g3 where g3 is the no. of tertiary switches.

Four stage network

For a four stage network, let the occupancy of A links be a, the occupancy of B

links be b, the occupancy of C links be c and the occupancy of outgoing trunks

be d.

For a connection from a given inlet on an input frame to a particular outlet on an

output frame (i.e. mode 1), the call may use any primary switch in the output

frame.

Probability of this path being free is = (1-a)(1-b)(1-c)

Therefore probability of this path being blocked is 1- (1-a)(1-b)(1-c)

Probability that all g2 independent paths are simultaneously blocked is

B1 = [1- (1-a)(1-b)(1-c)]g2

where g2 is the no. of secondary switches in input frame = no. of primary switches in

output frame.

Application of graph theory to link systems

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A graph is a collection of points known as nodes or vertices connected by lines

called edges or arcs. Conventional representation shows switches, network

graph focuses more on links without details of switches.

An important property of the network which is displayed channel graph is its

connectivity. This may be defined as minimum number of disjoint paths joining

the non-adjacent vertices. The larger the value of connectivity, lower is the

probability of blocks.

Channel graphs show how adequate connectivity is provided by adding a 3rd stage to the

2 stage network and a 4th stage to the partially interconnected 3 stage network.

THREE STAGE NETWORK

FOUR STAGE NETWORK

Multistage Switching

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It is inefficient to build complete exchanges in single stages. Single stage can only be

used to interconnect one particular inlet outlet pair.

Also the number of cross points grows as the square of the inputs for grading, N (N–

1)/2 for a triangular array and N (N–1) for a square array.

Also the large number of cross points on each inlet and outlet line implies a large

amount of capacitive loading on the message paths. Therefore, it is usual to build

exchanges in two or three stages to reduce the number of cross points and to provide

alternative paths. T

he sharing of cross points for potential paths through the switch is accomplished by

multiple stage switching. Fig. shows the three stage switching structure to

accommodate 128 input and 128 output terminals with 16 first stage and last stage.

Three stage networks

The structure shown in Fig. provides path for N inlets and N outlets. The N

input lines are divided into N/n groups of n lines each. Each group of n inputs

is accommodated by an n-input, k output matrix.

Thus multistage structure provides alternate paths. Also the switching link is

connected to a limited number of cross points. This enables the minimized

capacitive loading. The total number of cross points NX for three stage is

where N = Number of inlets-outlets

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n = size of each inlet-outlet group

K = number of second stage.

2NK = number of cross points in 1st and 2nd stage

= number of cross points in each array of second stage

= number of cross points in second stage.

Also

The three stage switching matrix require that k > 2n – 1 to generate no blocking.

Substituting K= 2n – 1 we get

For large N,

Substitute for n in above eqn.

Thus

Number of cross points for a single stage switching matrix to connect N inlets to N

outlets is Nx (SS) = N2. Hence from (5)

Nstage N (min,)

Combinational switching

Space and Time Switching

A tandem switching centre or the route switch of a local exchange must be

able to connect any channel on one of its incoming PCM highways to any

channel of an outgoing PCM highway. The incoming and outgoing highways

are spatially separate, so the connection requires space switching.

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Space switches:

Crosspoint matrix connects incoming and outgoing PCM highways. Different

channels of an incoming PCM frame may need to be switched by different

crosspoints in order to reach different destinations.Crosspoint is a 2 input

AND gate. One input is connected to incoming PCM highway and another to

connection store that produces a pulse at required instants.

Figure below shows space switches with k incoming, m outgoing PCM

highways carrying n channels.

Fig. Space Switch

The numbers are read out cyclically in synchronism with incoming PCM

frame. In each time slot, the number stored at corresponding store address is

read out and decoding logic converts this into a pulse on a single lead to

operate relevant crosspoint.

Time switches:

Time switch connects an incoming n-channel PCM highway to an outgoing n-

channel PCM highway. Since any incoming channel can be connected to any

outgoing channel, it is equivalent to a space-division crosspoint matrix with n

incoming and outgoing trunks.

Time slot interchange is carried out by means of two stores, each having a

storage address for every channel of the PCM frame. Speech store consists of

data of each of the incoming time slots (i.e. its speech sample) at a

corresponding address.

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FIG: TIME SWTICH

To establish a connection, the number (X) of the time slot of an incoming

channel is written into the connection store at the address corresponding to the

selected outgoing channel (Y).

Time switching introduces delay. If Y>X the output sample occurs later in the

same frame as the input sample. If Y<X, the output sample occurs in the next

frame.

Time Division switching networks:

The figure below shows a Space-Time-Space (S-T-S) switching network. Each

of the m incoming PCM highways can be connected to k links by crosspoints

in the A switch, and the other ends of the links are connected to the m

outgoing PCM highways by crosspoints in the C switch.

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Each link contains a time switch. To make a connection between time slot X

of an incoming PCM highway and time slot Y of an outgoing PCM highway,

it is necessary to select a link having address X free in its speech store and

address Y in its connection store. The time switch is set to produce a shift

from X to Y.

Fig : SPACE TIME SPACE SWITCHING NETWORK

Bidirectional paths:

PCM transmission systems use four wire circuits, it is necessary to provide

separate paths for the send and receive channels.

One way of doing this would be to provide a separate switching network for

each direction of transmission.

Fig : Bidirectional transmission

Time-Space-Time (T-S-T) switching network:

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Each of the m incoming and m outgoing PCM highways are connected to a

time switch.

The connection is established by setting the incoming time switch to shift

from X to Z setting the outgoing time switch to shift from Z to Y and

operating the appropriate crosspoint at time Z in each frame.

Time space time switching network

Telecommunication Network

A telecommunication network must have some common standards in order to obtain a better

performance and are as follows:

a) A transmission plan

b) A numbering plan

c) A charging plan

d) A routing plan

e) A signaling plan

f) The grades of service

g) Switching equipment performance

h) Interconnection with other networks

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i) Managing networks

These above standards are dependent and are related to each other. A telecommunication

network must give a better service to customer at the prices that they pay. Therefore

networks must compromise between performance and cost.

Analog Networks:

The number of levels in the public switched telephone network depends on the cost of

transmission and switching. Suppose consider an area that is small and highly

populated will have short distance between its primary centres and traffic is very high

fig. UK analog network

Fig above has three levels of switching center. They are group switching centres(GSC).

District Switching Centres (DSC) and Main Switching Centres(MSC). The four wire

transmission is used for the trunk network. The circuits LE and GSC employed two-wire

switching. The traffic between trunk circuit is handled by direct circuits between GSC’s.

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The North American network is shown in fig below which differs from fig.7.1 in several

respects.

Integrated Digital Networks:

The need for channeling equipment has been eliminated by the use of both the

transmission system and switching system using digital time-division multiplexing.

Integrated Digital Network (IDN) has the compatibility of both digital transmission

and switching.

Another advantage of the equipment for a digital switching network is that less space

is occupied as compared to a space division switching network. As shown in fig

below, the multiplexers and remote connections replaced the small local

exchanges.So having few local exchanges helps in fewer and larger junctions.

fig. Existing analog network

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fig. Restructured analog network

The IDN of British telecom is shown in fig below, DCCE (Digital Cell Center Exchanges)

serves local areas. DCCE is accessible by customers via remote concentrators unit (RCU).

fig. British telecom integrated digital network

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Integrated digital

network

fig. Component networks of an IDN

As shown in fig above, the data for common channel signalling between digital

exchanges form a separate signalling network.

Intelligent Network

The more complex services can be controlled by a centralized processor called a

service control point (SCP). A telecommunication network that has been enhanced in

this way is called intelligent network (IN). The exchange that makes the required

connection is called a service switching point (SSP).

The architecture of intelligent network is shown in fig above. The SCP is a

centralized processor and its software is organized in three levels.

Node software : This software provides common facilities such as signaling, database

access, transmission and alarm.

Services logic program (SLP): These are program that control various services.

The services logic execution environment (SLEE): This program hosts various SLP’s

and network with the basic call control and switching of SSP.

Private Networks

Private networks are secured leased lines that were access protected by a unique

user name and a password , private network encrypt data transmitted from an

access point before it bits the network and decrypt all the received traffic before

it gets the user.

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This basic functionality not the medium the PN spans, is the true definition of a

PN.

Fig below. shows the overall topology of a private network.

The private network data path

A typical end to end PN data could contain:

Several machines not under control of the corporation

A security gateway

An internal segment

An external segment

Private Data Networks

Public data networks consists of two parts:

1)The data network identification code (DNIC) of 4 digits

2)The network terminal number (NTN) of 10 digits.

DNIC consists of a data country code DCC of 3 digit followed by a network digit. By this

means a single country can have upto ten different data networks. A country can have more

than one DDC. The format user for the ten digits NTN can be determined by the network

from another country.

CONTROL OF SWITCHING SYSTEMS

Switching systems were fast evolving from being manually controlled to being

controlled by relays and electronically. Also it became possible to perform a number

of functions using the same hardware by using different programs. This revolutionary

technique is known as the stored program technique.

Call processing functions:

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For any call to be made, a sequence of operations takes place in which the calling and

the called customers’ lines and the connections to them change from one state to

another. The various states for a simple call between two customers whose lines

terminate in the same exchange is described below.

Idle state: Initially, the customer’s hand-set is in the ‘on-hook’ condition. The line is said to

be idle (state 0). The exchange meanwhile is continuously monitoring the line to detect a

change in state.

Call request signal: When the customer lifts the handset, current flows into the line and a

signal is sent to the exchange. This is also known as seize signal.

Calling line identification: The exchange now detects the line in which the calling condition

originated. For this equipment number (EN) to directory number (DN) translation occurs.

Determination of originating class of service: The originating class of service (COS)

corresponding to the range of services available to the calling customer. In an SPC exchange,

the customer’s COS is stored as data.

Identification of calling party: If the originating COS indicates a multi-party line, it is

necessary to ensure that the correct party is billed for the call.

Connection to the calling line: The exchange now makes a connection to the calling line.

Proceed to send signal: The exchange now sends a signal (dial tone) to the calling party

indicating to him to send the identity of the number he wants to call. The exchange now waits

for this information (state 1).

Address signal: The calling customer now dials the number of the person he wants to

contact in his hand-set. This is the address signal to the exchange.

Selection of outgoing line termination: The exchange now determines the required outgoing

line termination from the address signal it just received. For this DN-EN translation is done.

Determining the terminating class of service: Just as in the case of the caller, the COS of

the called line must also be checked.

Testing called line termination: The exchange first tests the called line before making a

connection to it because it may be busy or out of service.

Status signal: A status signal, called a call progress signal is now sent back to inform the

caller regarding the progress of the call. If the busy tone or number unobtainable tone is sent,

then the caller replaces the hand-set and connection is released. Idle (0) state is resumed.

Connection to called line: If the called line is obtainable and free, the exchange now makes

a connection to it.

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Alerting called customer: The exchange now sends an alerting signal to the called line, i.e.

the phone at the called line end now rings indicating the called party to answer the call. At

the same time, it sends a ring-back tone to the caller. The exchange now waits for an answer

(state 2).

Answer signal: When the called customer answers by lifting the hand-set, the line is looped

and current flows. This provides an answer signal to the exchange and thus it ceases to send

the ringing tone back to the caller and the called line. If the customer does not reply, the

calling line hangs up and idle (0) state is resumed.

Completion of the connection: On receiving the answer signal from the called customer, the

exchange completes the connection between the called and the calling parties.

Conversational state: Since connection is complete, they can converse as long as they want

(state 3). The exchange supervises the connection to detect the end of the call.

Clear signal: When each customer replaces the hand-set, line current ceases and provides a

clear signal to the exchange.

Release of connection: The exchange then clears down the connection and hence idle (0)

state is resumed.

Since the calling party is billed, the connection is released after the calling party

hangs up.

Various other situations resulting in problems can occur in case one of them hangs up and the

other does not. These problems are rectified using certain time based circuits.

Signal exchanges

Signals sent in the direction away from the caller (towards the called line) are called

forward signals.

Signals sent towards the caller (and away from the called line) are called backward

signals. Also each signal should produce a response in the opposite direction, thus

verifying correct operation as follows:

The call request signal is answered by the proceed to send signal.

The address signal is answered by the call status signal.

The answer signal is a response to the alerting signal

After the originating exchange has sent the address information to the terminating

exchange, the actions from receipt of address information to alerting the called

customer take place at the terminating exchange.

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When the answer signal is got from the called customer, connection is made and after

the conversation has ended, then connections are released.

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Idle

Call request (seize)

Connect to calling terminal

Proceed to send

Address signal

Connect to called terminal

Status signal

Alert called terminal

Answer

Connection

Clear signal

Disconnect

Time (not to scale)

Event

Fig. Timing of signals exchanged for a local call

Idle

Called terminal

Calling terminal

Switching system

Alert

Answer

Forward clear

Backward clear

Forward clear

Backward clear

Call request (seize)

Proceed to send

Address

Status

Answer

Fig. Signal exchange diagram for a local call

UNIT III

INTRODUCTION

The telecommunication system has to service the voice traffic and data traffic.

The traffic is defined as the occupancy of the server.

The basic purpose of the traffic engineering is to determine the conditions

under which adequate service is provided to subscribers while making

economical use of the resources providing the service.

The function performed by the telecommunication network depends on the

applications it handles. Some major functions are switching, routing, flow

control, security, failure monitoring, traffic monitoring, accountability

internetworking and network management.

NETWORK TRAFFIC LOAD AND PARAMETERS

In the study of tele-traffic engineering, to model a system and to analyse the

change in traffic after designing, the static characteristics of an exchange

should be studied.

The incoming traffic undergoes variations in many ways. Due to peak hours,

business hours, seasons, weekends, festival, location of exchange, tourism

area etc., the traffic is unpredictable and random in nature. So, the traffic

pattern/characteristics of an exchange should be analysed for the system

design. The grade of service and the blocking probability are also important

parameters for the traffic study.

Traffic load

An understanding of the nature of telephone traffic and its distribution with

respect to time (traffic load) which is normally 24 hours is essential. It helps

in determining the amount of lines required to serve the subscriber needs.

According to the needs of telephone subscribers, the telephone traffic varies

greatly

If the behaviour of the traffic shown above is systematic for period of time or

season, good judgement about the design of switching system or lines or

trunks or any common shared equipments can be made. Thus, the combination

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of historical records, experience, location of exchanges (business area or

residential area), vacations, govt. policies on holidays etc., decides the design

of telecommunication network. Various parameters related to traffic pattern

one discussed below :

Busy hour. Traditionally, a telecommunication facility is engineered on the

intensity of traffic during the busy hour in the busy session.

The busy hour vary from exchange to exchange, month to month and day to

day and even season to season.

Peak busy hour. It is the busy hour each day varies from day to day, over a

number of days.

Time consistent busy hour. The 1 hour period starting at the same time each day for which

the average traffic volume or the number of call attempts is greatest over the days under

consideration.

In order to simplify the traffic measurement, the busy hour always commences on the

hour, half hour, or quarter hour and is the busiest of such hours. The busy hour can also be

expressed as a percentage (usually between 10 and 15%) of the traffic occurring in a 24 hour

period.

Call completion rate (CCR). Based on the status of the called subscriber or the design of

switching system the call attmepted may be successful or not. The call completion rate is

defined as the ratio of the number of successful calls to the number of call attempts. A CCR

value of 0.75 is considered excellent and 0.70 is usually expected.

Busy hour call attempts. It is an important parameter in deciding the processing capacity of

an exchange. It is defined as the number of call attempts in a busy hour.

Busy hour calling rate. It is a useful parameter in designing a local office to handle the peak

hour traffic. It is defined as the average number of calls originated by a subscriber during the

busy hour.

Day-to-day hour traffic ratio. It is defined as the ratio of busy hour calling rate to the

average calling rate for that day. It is normally 6 or 7 for rural areas and over 20 for city

exchanges.

UNITS OF TELEPHONE TRAFFIC

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Traffic intensity is measured in two ways. They are (a) Erlangs and (b) Cent

call seconds (CCS).

Erlangs.

The international unit of traffic is the Erlangs.

The maximum capacity of a single server (or channel) is 1

erlang (server is always busy). Thus the maximum capacity in erlangs of a

group of servers is merely equal to the number of servers. Thus, the traffic

intensity which is the ratio of the period for which the server is occupied to

the total period of observation is measured in erlangs.

For the present day networks which support voice, data and many other

services, erlang is better measure to represent traffic intensity.

Cent call seconds (CCS). It is also referred as hundred call seconds. CCS as a measure of

traffic intensity is valid only in telephone circuits. CCS represents a call time product.

This is used as a measure of the amount of traffic expressed in units of 100 seconds.

Sometimes call seconds (CS) and call minutes (CM) are also used as a measure of traffic

intensity. The relation between erlang and CCS is given by

1E = 36 CCS = 3600 CS = 60 CM

Traffic Parameters

The following statistical information provides answer for the requirement of

trunk circuits for a given volume of offered traffic and grade of service to

interconnect the end offices. The statistical descriptions of a traffic is

important for the analysis and design of any switching network.

Calling rate.

This is the average number of requests for connection that are made per unit

time. If the instant in time that a call request arises is a random variable, the

calling rate may be stated as the probability that a call request will occur in a

certain short interval of time.

If ‘n’ is the average number of calls to and from a terminal during a period T seconds,

the calling rate is defined as

λ = n

T

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In telecommunication system, voice traffic and data traffic are the two types of

traffic. The calling rate (λ) is also referred as average arrival rate. The average calling rate is

measured in calls per hour.

Holding time.

The average holding time or service time ‘h’ is the average duration of

occupancy of a traffic path by a call. For voice traffic, it is the average holding time per call

in hours or 100 seconds and for data traffic, average transmission per message in seconds.

The reciprocal of the average holding time referred to as service rate (µ) in calls per hour is

given as

µ = 1

h

Sometimes, the statistical distribution of holding time is needed. The

distribution leads to a convenient analytic equation. The most commonly used

distribution is the negative exponential distribution. The probability of a call

lasting atleast t seconds is given by

P(t) = exp (– t/h)

For a mean holding time of h = 100 seconds, the negative exponential

distribution function is shown in Fig.

Fig. Negative distribution function for h = 100 sec.

Figure shows that, 50% probability call lasts longer than 70 sec.

Distribution of destinations.

Number of calls receiving at a exchange may be destined to its own exchange

or remoted exchange or a foreign exchange. The destination distribution is

described as the probability of a call request being for particular destination.

User behaviour.

The statistical properties of the switching system are a function of the

behaviour of users who encounter call blocking.

Average occupancy.

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If the average number of calls to and from a terminal during a period T

seconds is ‘n’ and the average holding time is ‘h’ seconds, the average

occupancy of the terminal is given by

A =nh= λh= λµ

T

Thus, average occupancy is the ratio of average arrival rate to the average

service rate. It is measured in Erlangs. Average occupancy is also referred as

traffic flow or traffic intensity or carried traffic.

GRADE OF SERVICE (GOS)

For non-blocking service of an exchange, it is necessary to provide as many

lines as there are subscribers. But it is not economical. So, some calls have to

be rejected and retried when the lines are being used by other subscribers.

GOS = Blocked Busy Hour calls

Offered Busy Hour calls

GOS =A-A0

A

where A0 = carried traffic (equation 8.4)

A = offered traffic

A – A0 = lost traffic.

The smaller the value of grade of service, the better is the service. The recommended GOS is

0.002, i.e. 2 call per 1000 offered may lost. In a system, with equal no. of servers and

subscribers, GOS is equal to zero.

GOS is applied to a terminal to terminal connection. But usually a switching centre is

broken into following components

An internal call (subscriber to switching office)

An outgoing call to the trunk network (switching office to trunk)

The trunk network (trunk to trunk)

A terminating call (switching office to subscriber).

The GOS calculated for each component is called component GOS. The overall GOS

is in fact approximately the sum of the component grade of service.

There are two possibilities of call blocking. They are

Lost system

Waiting system.

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In lost system, a suitable GOS is a percentage of calls which are lost because no

equipment is available at the instant of call request. In waiting system, a GOS objective could

be either the percentage of calls which are delayed or the percentage which are delayed more

than a certain length of time.

BLOCKING PROBABILITY

The value of the blocking probability is one aspect of the telephone

company’s grade of service.

The blocking probability is discussed in unit 2. The basic difference between

GOS and blocking probability is that GOS is a measure from subscriber point

of view whereas the blocking probability is a measure from the network or

switching point of view.

Based on the number of rejected calls, GOS is calculated, whereas by

observing the busy servers in the switching system, blocking probability will

be calculated. The blocking probabilities can be evaluated by using various

techniques. Lee graphs and Jacobaeus methods are popular and occurate

methods (See unit 2). The blocking probability B is defined as the probability

that all the servers in a system are busy.

Congestion theory deals with the probability that the offered traffic load

exceeds some value. Thus, during congestion, no new calls can be accepted.

There are two ways of specifying congestion.

They are time congestion and call congestion. Time congestion is the

percentage of time that all servers in a group are busy. The call or demand

congestion is the proportion of calls arising that do not find a free server. In

general GOS is called call congestion or loss probability and the blocking

probability is called time congestion.

If the number of sources is equal to the number of servers, the time congestion

is finite, but the call congestion is zero. When the number of sources is large,

the probability of a new call arising is independent of the number already in

progress and therefore the call congestion is equal to time congestion.

1. Lee graphs. It was proposed by C.Y. Lee. It is a most versatile and straight forward

approaches of calculating probabilities with the use of probability graphs.

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2. Jacobaeus method. It was presented in 1950 by C. Jacobaeus. It is more accurate than Lee

graph method.

Lee graphics. C.Y. Lee’s approach of determing the blocking probabilities of various

switching system is based on the use of utilization percentage or loadings of individual links.

Let p be the probability that a link is busy. The probability that a link is idle is

denoted by q = 1 – p. When any one of n parallel links can be used to complete a connection,

the blocking probability B is the probability that all links are busy is given by

B = pn------------------------------------()

when a series of n links are all needed to complete a connection,

-------------------------------------------()

For a probability graph of three stage network, shown in Fig. 5.12, the probability of

blocking is given by

----------------------------------()

MODELING SWITCHING SYSTEM

To analyze the statistical characteristics of a switching system, traffic flow

and service time, it is necessary to have a mathematical model of the traffic

offered to telecommunication systems.

The model is a mathematical expression of physical quantity to

represents the behaviour of the quantity under consideration. Also the model

provides an analytical solutions to a teletraffic problems.

Also in data transfer, a system has to buffer message while

waiting for transmission. Here size of the buffer depends on traffic flow. As

serving the number of subscribers subject to fluctuation (due to random

generation of subscriber calls, variations in holding time, location of the

exchange, limitation in servers etc), modelling of traffic is studied using the

concepts and methods of the theory of probability. If a subscriber finds no

available server for his call attempt, he will wait in a line (queue) or leave

immediately.

BLOCKING MODELS

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The traffic generated by a subscriber is random in telecommunication

switching system. The behaviour of the network for the random request for

service is random process. In this section, some elemental ideas of probability

theory and probability distributions are discussed.

Probability.

Probability can be defined as the relative frequency of occurrence of a random

event. Each event has a probability defined as the ratio of the number of times

it occurs to the total number of trials. Thus the probability of the occurrence of

an event A for N

trials is

P(A) = lim N →∞NA

N

The probability of occurance of an event P is a positive number and that 0 ≤ P

≤ 1. If an event is not possible P = 0, while if an event is certain, P = 1.

Mutually Exclusive events.

Two possible outcomes of an experiment are defined as being mutually

exclusive if the occurrence of one outcome precludes the occurrence of the

other. If the events are A and B with probabilities P(A1) and P(B), then the

probability of occurrence of either A or B is written as

P(A or B) = P(A) + P(B) ...(8.9)

For more than two mutually exclusive outcomes, say A1, A2, ..., An.

P(A1 or A2 or ... AL) = ∑ P(Aj)

j =1, 2 ….L

For example, in tossing a coin, if head occurs, occurrence of tail cannot take place.

Conditional and Joint probability.

Suppose that we contemplate two experiments and B with outcomes A1, A2, ...

and B1, B2, ... . The probability of outcome Bk, given that Aj is known to have

occurred is called conditional probability given as

P (Bk|Aj) = P (Aj, Bk)

P (Aj)

Similarly, the probability of outcome Aj, given that Bk is known to have occurred is given as

P (Aj||Bk) = P (Aj, Bk)

P (Bk)

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where P (Aj, Bk) is called joint probability, that is the joint occurrence of A j and Bk. From the

above two equations then P (Aj, Bk) is given as

P (Aj, Bk) = P (Bk|Aj) P (Aj) = P (Aj||Bk) P(Bk)

Sub P (Aj, Bk) in above equation is rearranged as

P (Aj|Bk) = P (Aj) P (Bk|Aj)

P(Bk)

This result is known as Bayes’ theorem. If the outcome of Bk does not depend

at all on which outcome Aj accompanies it, we say that the outcomes A j and

Bk are independent. When outcomes are independent, the probability of a joint

occurrence of particular outcomes is the product of the probabilities of the

individual independent outcomes. It is given as

P (Aj, Bk) = P(Aj) P(Bk)

Random variables and Random process.

Subscribers generates calls in random manner. The call

generation by the subscribers and therefore the behaviour of the network or

the switching system is described as a random process

In telecommunication system, telephone traffic is referred as

random process and the number of simultaneous active subscribers and

simultaneous busy servers are assumed as random variables.

Definition of statistical terms :

Mean : E(x) = ∑ xi/N

i= 1,2…. N

where x = a random variable

N = number of trials

xi = the outcome of the individual trials.

Mean is also referred as the expectation or average of x and represented by µ.

Variance. The variance of x is

Var (x) = ∑ (xi-E(x))2 =∑ (xi) 2 - E(x)2

i= 1,2…. N i= 1,2…. N

N N

The variance of a process is a measure of how the individual outcomes differ from the mean.

Standard deviation. The standard deviation of a random variable x is given by

σ(x) = sqrt (Var(x))

The ratio of standard deviation to the mean of a random variable x is given by

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ρ(x) = σ(x)

E(x)

Discrete Probability Distributions

While the mean and variance tell a great deal about a random variable, they do not

tell us everything. The most complete information is given by the distribution of the random

variable. The distribution is the probability associated with each possible outcome.

Mean and variance. The values of P(x1), P(x2), ..., P(xn) for a discrete random process X =

x1 , x2 , ..., x may be plotted as shown in Fig.

The value of x corresponding to the centre of gravity of the above diagram is called the

average or mean or expectation E(x)

µ = E(x) = ∑ xj P(xj)

j= 1,2…. N

Fig. Discrete probability distribution.

P(xj) must be non-negative and must sum to 1. The mean µ is also known as the first

moment of the random variable. µk , the kth moment of the random variable is defined as

µk = ∑ (xj)k P(xj)

j =1, 2,….∞

The variance Var(X) or σ2 is a measure of the dispersion or spread of the histogram.

It is defined as the mean squared deviation of x from the mean.

Var(X) = σ2 = ∑ (xj-µ)2 P(xj)

j =1, 2,….∞

=∑ (xj)k P(xj) -µ2

j =1, 2,….∞

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The variance is generally referred as second centrol moment. The kth central moment

is defined as

Ck = ∑ (xj-µ)k P(xj)

j =1, 2,….∞

The standard deviation, σ if defined as the square root of the variance.

σx = sqrt(Var (X))

If X and Y are independent random variable, the mean and variance of sum or

difference is given by

E( X + Y) = E(X) + E(Y)

Var (X + Y) = Var (X) + Var (Y)

E(X – Y) = E(X) – E(Y)

Var (X – Y) = Var (X) + Var (Y) [not minus]

Bernouilli or binomial distribution. The distribution of x repetitions of an event (say head

of toss) with two possible outcomes is called a binomial distribution and the numbers above

are called binomial coefficients. Consider the series of trials (n) satisfies the following

conditions :

(a) Each trial can have two possible outcomes. e.g. success or failure with

probabilities p and 1 – p.

(b) The outcome of each trial is an independent random event.

(c) Statistical equilibrium (i.e. the probabilities do not change).

The number of ways of choosing an x things out of n trial is given as

C(n, x)= n!___

x! (n-x)!

C(n, x) is called binomial coefficient.

The probability of one particular combination of x success and n – x failures is px (1 – p)n–x.

Given two disjoint events with probabilities p and (1 – p), the probability of first occurring

x times and second occurring (n – x) times in n trial, the most general form of binomial

distribution is

P(n, x, p) = C(n, x) px (1 – p)n–x

or simply

P(x) = n p x (1 – p) n–x .

x

The mean is µ = n p

The variance is σ2 = n p (1 – p)

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UNIT IV

Subscriber Loop Design

The cables that connect the telephone handsets or other devices

to the local switching office or end office is referred as subscriber loop or

local loop.

One end of each subscriber loop is terminated on a Main

Distribution Frame (MDF) at the exchange. The drop wires (DW) from the

telephones are connected to the distribution point (DP) which is located near

the subscriber’s premises. The distribution points at various locations are

connected together by a distribution cables (DC) and terminated to the feeder

points (FD).

Fundamental Characteristics

The subscriber loop is the most common interface in the network. The fundamental

characteristic of this interface are.

Battery. To enable dc signalling and to provide bias current for carbon microphone, a battery

of about 48 V is connected to subscriber loop at exchange.

Overvoltage protection. Protection of equipment and personal from lightning strikes and

power line induction or shots.

Ringing. Application of a 20 Hz signal at 86 V rms for ringer excitation.

Supervision. Supervise the network by detecting the off hook/on hook and flow/noflow

dc current.

Coding. In the case of digital end office, analog to digital coding and digital to analog

decoding functions necessary.

Hybrid. For two wire to four wire conversion, hybrid in necessary.

Test. Line test toward the subscriber disconnection of the switch.

The first letter of the above characteristics are coined together which is commonly known as

BORSCHT.

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Limiting Factors of Subscriber Loop Design

There are two limiting factors we have to consider while

designing a subscriber loop. First one is the attenuation. The attenuation

refers to the energy loss in the line at a reference frequency, measured in

decibels

The second limiting factor is voltage drop. If the battery

voltage is kept constant with increase in length, the effectiveness of the

signalling and conversation will be limited.

This is due to IR drop of the line. The IR drop of the line varies

with resistances of the battery used in the system, telephone set resistance

and the allowable resistance of the subscriber loop.

The maximum allowable resistance in the subscriber loop and the loop

resistance limit is calculated as follows.

where Rm = maximum allowable resistance of subscriber loop.

VB = Battery voltage

Ic = minimum current required for proper operation of carbon microphone

The loop resistance limit is RL = Rm – (RB + RT)

where

RB = approximate resistance maintained at the battery protect against short circuit in the wire

between subscriber and local office

RT = Telephone set resistance.

Loop Length

The method of determining subscriber loop length using the signal resistance limit as

a basis is called the basic resistance design. The maximum subscriber loop length, which is

defined as the distance from the subscriber to the central office, is expressed as

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The dc loop resistance is measured in ohms per km

Cable size for the Loop

The knowledge of dc loop resistance, the diameter of copper wire can be determined. The

table shows American Wire Gauge (AWG) versus wire diameter and resistance.

AWG versus wire diameter and resistance

The cable sizes of 19, 22, 24 and 26 gauge are the most commonly used cable

for different dc resistance of various subscribers.

The higher the gauge number the smaller the wire diameter. With 26-gauge

wire a loop distance of only about 6.4372 Km (4 miles) is possible. With 19-

gauge wire the loop distance might be extended to as much as about 28.96 Km

(18 miles).

Digital Subscriber Line (DSL) Technology

XDSL is a generic abbreviation for the many variations of digital subscriber

line technology.DSL refers to the technology used between a customer

premises and the telephone company enabling more bandwidth over the

already installed copper cabling.

Thus XDSL is a technology backed by telephone companies to provide next

generation high bandwidth services to the home and business using the

existing telephone cabling infrastructure.

XDSL Comparison with Other Technologies

XDSL is compared with the existing technologies like cable modems, ISDN,

T1, voice band modem and wireless technologies based on the network

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formation and speed. Cable TV network is a broad band technology provides

services using cable modems.

In this network, each subscriber in an area receives the same signals as all

others in that area. The cable modem has potential bandwidth in the range 30

Mbps from the service provider to subscribers.

Cable networks require two paths, one for down stream and one for

upstream.XDSL is circuit oriented. Thus, each subscriber connection is

independent of other. It uses existing telephone lines and does not require

separate arrangement for upstream and downstream like cable TV network.

ISDN technology provides digital services in increments of 64 kbps channels.

ISDN requires the phone company to install services within their phone

switches to support their digitally switched connection service.

T1 (E1 is the European near equivalent) line is a 1.544 Mbps PCM system

comprised of 24 TDM channels of 64 kbps each. T1 lines have been installed

for end users who require dedicated high speed bandwidth between their home

and work (or internet).

T1/E1 uses have been used in voice and data networks thoughout the world

where highly available, high capacity networks needed to be built.

Voice modems or simply modems allows digital data to flow over the

telephone company’s traditional telephone network by performing digital to

analog conversions for transmission on to the network and vice versa on the

receiving end.

Modems are limited to telephone company’s voice bandwidth service (hence

the name voice modem). With the bandwidth of 3000 Hz, typical speed is 56

kbps. Wireless access technology provides access to large number of

subscribers in a relatively large area. Bandwidth ranges from few kbps to

many Mbps.

Various Types of XDSL

Several forms of XDSL are designed to suit specific application, achieve

specific goals and satisfy the needs of subscribers.

XDSL may best be categorised within the modulation methods used to encode

data and that uses the POTS splitter or not. A POTS splitter uses a low pass

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filter to separate the low end frequencies of the telephone audio spectrum

from the higher frequencies of the XDSL signals.

The splitter allows the traditional voice service that subscribers accustomed

to. A splitter is required at both the customer premises and at the telephone

exchange. XDSL that does not use a POTS splitter on customer premises is

termed ‘‘splitter less XDSL”.

In this case splitter function is performed at the service provider end. Brief

summary of well known XDSL’s are given below.

ADSL.

Asymmetric Digital Subscriber Line (ADSL) is the most popular form of XDSL

technology. Its upstream and downstream bandwidth is asymmetric or uneven.

The ADSL can provide upstream (user to provider) data rate from1.5 Mbps to 9

Mbps. Typical downstream (provider to user) speed range from 64 kbps to 1.5 Mbps.

In practice, the bandwidth of downstream is high and is the high speed path.

ADSL Lite or G.lite.

It is a low rate version of ADSL. It was proposed as an extension to ANS1 standard.

T1.413 by UAWG (Universal ADSL working group) led by Microsoft, Intel and

compaq. This is known as G.992.2 in the ITU standards.

It uses the same modulation scheme as ADSL (DMT) ; but eliminates POTS splitter

at the subscriber premises. It results in lower available bandwidth.

VDSL.

The very high bit rate digital subscriber line (VDSL) is similar to APSL and uses the

DMT modulation technique. It is proposed for shorter local loops perhaps from 300 to

1800 meters.

HDSL.

High Bit-rate Digital subscriber Line (HDSL) is generally used as a substitute for

T1/E1. It was designed by Bellcore (now Telcordia). T-1 line uses AMI encoding,

which is very susceptible to alternation at higher frequencies.

This limits the length of aT–1 line to 1 km and requires repeaters every 6000 ft to

boost the signal strength. HDSL uses 2BIQ encoding. HDSL is becoming popular as

it provides full duplex symmetric data communications at rates up to 1.544 Mbps

(2.048 Mbps in Europe).

The 2BIQ encoding is less susceptible to attenuation and hence the good rating can be

achieved without repeaters upto a distance of 3.6 km. HDSL 2 was designed to

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transport T1 signalling at 1.544 Mbps over a single copper wire. HDSL 2 uses

overlapped phase Trellis-code interlocked spectrum (OPTIS).

SDSL.

Symmetric Digital Subscriber Line (SDSL) is a 2 wire implementation of HDSL. It

achives the same data rate of HDSL. It supports T1/E1 on a single pair to a distance

of 3.2 km.

IDSL.

ISDN based DSL (IDSL) uses 2BIQ line coding and typically supports data transfer

rates of 128 kbps. This technology is similar to ISDN, but uses the full bandwidth of

two 64 kbps bearer channels plus one 16 kbps delta channel.

G. SHDSL.

It is a ITU standard and offers a rich set of features (e.g. rate adaptive) and offers

greater rate than many current standards. This technology is able to replace T1, E1,

HDSL, SDSL, HDSL 2, ISDN, IDSL technologies.

OTHER.

The popular proprietary DSL includes (a) Consumer installable DSL (CiDSL) is a

splitterless DSL proposed by Globespan’s (b) Etherloop proposed by Nortel. (c) Multi

rate digital subscriber line (MDSL) and (d) Rate adaptive DSL (RADSL) designed by

Glopes pan semiconductor).

ADSL:

The DSL Forum was formed in December 1994 to promote the DSL concept and

facilitate developement of DSL system architectures, protocols, and interfaces for

major DSL applications.The ANSI, working group TIE1.4, approved the first ADSL

in 1995. It supported data rates upto 6.1 Mbps. The ETSI contributed an annex to

T1.413 to reflect European requirements.

The ITU-T standards are most commonly referred to as G.lite (G.992.2) and G.dmt

(G.992.1). The ATM Forum has recognised ADSL as a physical layer transmission

protocol for unshielded twisted pair media. ADSL, a modem technology, converts

existing twisted pair telephone lines (subscriber loop) into access paths for

multimedia and high speed data communications. ADSL can transmit up to 6 Mbps to

a subscriber, and as mush as 832 kbps or more in both directions.

Frequency spectrum.

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ADSL divides the bandwidth of a twisted pair cable into three bands. The

twisted pair cable used in telephone wire has a frequency spectrum of

ADSL.Fig. 10.9 shows the frequency spectrum of ADSL.

ADSL uses various encoding methods to divide the available bandwidth of the

channel into multiple subchannels. Earlier, FDM or Echo cancellation are

used to divide the available channels

Fig. Frequency Spectrum of ADSL

The frequency spectrum from 26 kHz to 138 kHz is used for upstream

transmission, and the frequency spectrum from 1.38 kHz to 1.1 MHz is used

for down stream transmission.

The lower 4 kHz channel is separated by an analog circuit and used in POTS.

The frequency spectrum above 26 kHz is divided into 249 independent

subchannels, each containing 4.3 kHz bandwidth. 25 channels are used for

upstream transmissions and 224 channels are used for downstream

transmissions.

Topology

ADSL modem is connected to each end of twisted pair, one at the subscriber

end and other at the central office.

The ADSL modem at the exchange is called ATU-C (ADSL-Terminal Unit

Central office) and the ADSL modem at the subscriber end is called ATU-R

(ADSL-Remote). Fig. 10.10 shows the ADSL connection between exchange

and subscriber. ADSL modems with various speed ranges and capabilities

available.

Fig. BELOW shows the modem with 1.5 Mbps to 9 Mbps downstream bit rate

and 16 to 640 kbps duplex channel. The fig shows the topology of an ADSL

system. To the access node different types of services such as Digital

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broadcast, broadband network, Narrow band network, Network management

etc enteres. The access node provides interfacing of broadband services to

ATUC (ADSL–terminal unit central office).

ATU-C converts the data into ADSL format. The ADSL format fed to the

splitter multiplexes them onto a single loop line.

Telephone connection from the PSTN enter the system at the splitter level and

are added to POTS-C (POTS–central office) area of the ADSL spectrum.The

splitter in POTS-R (POTS–Remote) de-multiplexes and transfers phone calls

to the phones.

The ADSL formats are transferred to ATU-R (ADSL-Terminating unit

remote) which in turn converts to the orignal format and supplies to the

terminal ends (TE).

ADSL Modem Connection

Topology of an ADSL system

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ADSL Frame.

The transport of data packets over ADSL requires link layer protocols. There are two

ANSI stardards.

(a) Point-to-point protocol (PPP) variable length data units within and HDLC framed

structure (RFC 1662).

(b) The ATM Forum’s standard for ATM Frame UNI, also within an HDLC framed

structure.

Below fig shows the ADSL frame format. The frame begins with the standard HDLC

flag (7E in hex) followed by a PPP address code field of FF and 03 hex. Two bytes

of protocol ID identifies the payload type and the possible protocol that has been

encapsulated in it. The frame check sequence (FCS) field uses CRC-16 for error

detection and the frame ends with another 7E flag.

ADSL Capabilities.

ADSL will play a crucial role over the next ten or more years. Thus the service

providers shall enter new markets for delivering information in video and multimedia

formats.

By bringing movies, television, video catalogs, remote CD-ROMs, Corporate LANs,

and the internet into homes and small businesses, ADSL will make these markets

viable, and profitable for telephone companies. ADSL modems available with

various speed ranges and capabilities.

Downstream data rates depends on a number of factors, including the length of the

copper line, its wire gauge, presence of bridged taps (accidental connection of

another local loop to the primary local loop.

It behaves as an open circuit at DC, but becomes a transmission line stub with

adverse effects at high frequency) and cross coupled interference. Line attenuation

increases with line length and frequency, and decreases as wire diameter increases.

Ignoring bridge gaps, ADSL will perform as shown in table

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Advantages of ADSL

It provides many advantages to telecom companies and users. Some of them are :

It provides a simple, affordable mechanism to get more bandwidth to end users, both

residential and small to medium businesses.

The high speed downstream is increasingly important for internet access, remote

access to corporate server, integrated voice/data access and transparent LAN

interconnection.

It enables carrier to ofter value added, high speed networking services.

In a telecommunication network, signalling systems are as essential as switching

systems and transmission systems. They must be compatible with the switching

systems as they must be able to transmit all the signals required to operate the

switches. They must also be compatible with the transmission system in order to

reach the exchange that they control. Thus, design of signalling systems is directly

influenced by both switching and transmission requirements.

Exchanges usually send signals over the same circuits in the network as the

connections which they control. This is known as channel associated signaling. In

SPC, the need for more signals to be transmitted between exchanges arise. These

signals are transmitted between two processors of two different exchanges over a

separate data channel. This is known as common channel signalling (CCS). Signaling

can be classified as follows.

Summary of Various DSL Technologies

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Transmission Plan

A hybrid is required to convert a 2-wire circuit into a 4-wire circuit between

the subscriber and a digital exchange. In analog exchanges, local calls are

established on 2-wire circuits. But, long distance calls require 2-wire to 4-wire

conversion at the subscriber line-trunk interfaces Inter exchange or intercity

trunk lines carry a number of conversations on a single bearer circuit which

may be derived from a coaxial cable, microwave or satellite system.

Due to the long distances involved the bearer circuits need amplifiers or

repeaters at appropriate intervals to boost the signals. The amplifiers are

almost invariably one-way devices and cannot handle bidirectional signals.

Since the telephone conversation calls for signal transmission both ways, long

distance trunks require separate circuits for each direction, leading to 4-wire

circuits.

Hence, the need for 2-wire to 4-wire conversion in long distance connections

The conversion is done by the hybrids.

Short delay echos are controlled by using attenuators and long delay ones by

echo suppressors or cancellers. CCITT recommends the use of echo

suppressors or cancellers if the round trip delay exceeds 50m seconds .

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Use of echo suppressors is mandatory in satellite circuits as the round trip

delay involved is several hundred milliseconds. For delays up to 50 ms,

simple attenuators in the transmission path limit the loudness of echo to a

tolerable level,

The fig below shows that the attenuation required increases as the delay increases

Attenuation vs Echo delay

If the echo path delay is 20 ms, ll-dB attenuators must be introduced in the

transmission paths. It may be noted that this loss must be accounted for in the

overall transmission loss budget, i.e. in ORE

The operation of an echo suppressor is illustrated in Fig. Echo J suppressors

are voice activated attenuators. Normally, the echo suppressors remain in a

deactivated state, i.e. the attenuators are bypassed. Speech in one channel

activates the echo suppressor in the return path In Fig. 14

Fig Echo suppressor operation

speech activates the echo suppressor EB and B speech EA. Figure Y.W' depicts the situation

when B is talking and A is silent.

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Should A attempt to talk at this juncture, his talk is also attenuated. He can, however, turn off

the echo suppressor by interruptingB loudly. The echo suppressor is deactivated

automatically a few milliseconds after the talker stops speaking

One drawback of echo suppressors is that they may clip the beginning portion

of speech segments. If subscriber A begins talking at the tail end of B's

speech, his talkspurt is not transmitted until the echo suppressors have had

time to reverse directions. New designs of echo suppressors attempt to

minimize the time required to reverse directions.

Typically reversal times are in the range of 2-5 ms The operation of a system

with echo suppressors is clearly half duplex When telephone circuits are used

for data transmission (see Section 10.1), full duplex operation is required.

Moreover, with several milliseconds of interruption while an echo suppressor

in one direction is turned off and the one in the other direction is turned on, it

is very difficult to organize data transmission.

Hence, echo suppressors are usually disabled while the circuits are used for

data transmission. This is done by providing a disabler feature in the echo

suppressor and triggering the same with a special signal

The amount by which the reflected signal is attenuated is known as return

loss. This is given by

RL= 20 LOG (Z4+Z2 ) / (Z4+Z2) dB

Where

RL = return loss

z4 = impedance of the 4-wire circuit (or) in terms of power,

RL= 20 LOG (P4) / (P4+P2) dB

Where

P4 = incoming power on the 4-wire circuit

P2 = power reaching the 2-wire circuit

P4 -P2 = power reflected on to the return path

or , in terms of signal voltages,

RL= 20 log V4 / (V4 - V2)db.

= 20 log 1 / r c

Where r c is the reflection co-efficient defined as

r c= reflected signal / incident signal

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If the two networks are perfectly balanced, then Z4= Z2 .therefore we have

RL (balanced ) = 20 log 2z2 / 0 = α

The return loss is infinite, i.e. the return signal experiences an infmite attenuation and

hence there is no reflected signal

Transmission System

Laser Communication System

Laser communication system is another mode of telecommunication which

occurs through wireless connections in the atmosphere. Laser

communication came into existence in 1960 and since then many

advancements have been made in this discipline.

In this mode of communication, the information is transferred through free

space. In the laser mode of communication; the signals are transmitted from

the wireless transmitter to a wireless receiver without any hindrance or

obstruction. Such condition is also called line of sight condition where the

signals are transmitted without any obstruction. Laser diode is the major

carrier in this mode of communication.

It does not require any kind of wires and cables and hence is not a very

expensive mode of communication. This mode of communication is also

faster as compared to the other modes and thus is mostly preferred over other

types of communication system. One thing to be careful in a laser mode of

communication is that the signals must flow without any hindrance.

Microwave communication system

Microwave communication occurs with the help of microwaves.

The radiowaves whose wavelengths can be conveniently measured in small

numbers of centimeters with the help of various electronic technologies are

called as Microwaves.

Thus a microwave communication is majorly with the help of radiowaves

with a small and a measurable wavelength. In this mode of communication,

antennas of convenient sizes are instrumental in carrying the radiowaves to

facilitate the microwave communication.

The technology for microwave communication came into picture in 1940 in

Western Union. After five years in 1945, the first microwave message was

sent between towers, one located in New York and the other tower located in

Philadelphia.

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After the successful transmission of microwave message, this became the

most commonly used mode of data transmission. There are two types of

Microwave communication. Analog and Digital Microwave communication,

where digital is more advanced as compared to analog microwave system.

Both the above are communication types which help in the transmission of

signals for the process of communication to take place. Whether it is satellite,

or optical, or laser, or microwave their objective is same - the transmission of

signals from one party to the other.

Numbering Plan

The numbering plan is used to identify the subscribers connected in a

telecommunication network.

The main objective of numbering plan by any nation is to standardise the number

length wherever practical according to CCITT recommendations. Other objectives

includes (a) to meet the challanges of the changing telecom environment (b) to

meet subscriber needs for meaningful and user friendly scheme (c) to reserve

numbering capacity to meet the undefined future needs. In this section,

recommendations of ITU, International and National numbering plan are

discussed. The numbering plan in India is also focussed.

ITU Recommendations in Numbering

Some important recommendations of ITU are discribed below :

Recommendation E.164: It provides the number structure and functionality for three

categories of numbers used for international public telecommunication. The three categories

of numbers are :

National telephone services . An international public telecommunication number (for

geographic areas) is also referred to as the national significant number (NSN). NSN

consists of the country code (CC), national destination code (NDC) and the subscriber

number (SN).

Global telephone services. An international public telecommunication number for

global telephone service consists of a three digit country code and global subscriber

number. The country code is always in the 8XX or 9XX range.

International networks. An international public telecommunication number for

international networks consists of three digit country code, a network identification

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code and a subscriber number. The country code is alsways in the 8XX range. The

identification code is one to four digits.

Recommendation E.123. This defines a standard way to write telephone numbers,

email addresses and web addresses. It recommends how to use hyphen (-), space ( ),

or period (•) . ( ) are used to indicate digits that are sometimes not dialled, / is used to

indicate alternate numbers and • is used in web addresses.

Recommendation E.162. This recommendation describes that the originating

country must analyse a maximum of seven digits of the E.164 international number.

When a number is being analysed, it will be done according to this recommendations.

Also, the international numbering plan or world numbering plan has been defined in

recommendations E.160 ; E.161 and E.162.

International Numbering Plan

This plan has to be implemented irrespective of a country’s national

numbering plane and implemented in accordance to the recommendations of

ITU. With some standard international framework, subscribers from different

countries can call each other.

This plan makes it possible to access all countries with the same country code

anywhere in the world. For the international numbering plan, the world has

been divided into nine geographical area as given below.

Table shows the zone code, Zone and two examples in each zone. Digit ‘0’ is not used to

indicate zone. Generally ‘0’ is used as Trunk prefix and ‘00’ is used for international prefix.

Fig. World numbering zones

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Table World numbering zones

The numbering format for international telephone number is shown in Fig. An inter-

nation telephone number starts with one to three digit country code followed by 9 to 12

subscriber number. The dialling procedure is that the international prefix ‘00’ should be

dialled first followed by the telephone number.

Fig. International telephone number

National Numbering Plan

Each country decides for itself what kind of numbering plan it can have. A

numbering plan may be open, semi open or closed. Each country decides what

rules to follow when issuing telephone numbers. Such a numbering plan is called

national numbering plan.

An open numbering plane or non-uniform numbering scheme allows variations in

the number of digits to be used to identify the subscriber. This plan is used in

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countries equipped extensively with non-director strowger switching system. This

scheme is almost extinct.

A closed numbering plan or uniform numbering plan refers to a numbering plan

which only allows telephone numbers of a predetermined length. Special services

(toll free, premium rate, etc.) are usually excluded from this rule.

The dialling procedure for national numbering plan are also comes in two

categories. A closed numbering plan refers to a numbering plan which requires

users to dial all numbers at all times.

This means that local-local calling also requires the area code to be dialled, as

well as the trunk prefix. In open dialling plan local calls can be placed without the

trunk prefix and area code. National numbering format is shown in Fig.

Fig. National numbering format

Thus, the National significant number (NSN) is the combination of trunk code, exchange

code and the number. The exchange code and line number together called as subscriber

number (SN). NSN length varies from country to country.

Numbering Plan in India

DOT India has released its national numbering plan dated April 2003. It was

last reviewed during 1993. This existing numbering plan was formulated at a

time when there was no competition in the basic telecom services were not

available in the country.

Further, the existing numbering plan was meant to address monopolistic

environment in national and international long distance dialling. The new

numbering plan has been formulated for a projected forecast of 50%

teledensity by the year 2030 and thus making numbering space available for

75 crore telephone connections in the country comprising of 30 crore basic

and 45 crore cellular mobile connections.

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The new national numbering plan will be able to meet the challenges of multi

operator, multi service environment and will be flexible enough to allow for

scalability for next

30 years without any change in basic structure. This plan is aimed at PSTN services, cellular

mobile services and paging services. This section focuses mainly on the PSTN services.

List of commonly used abbreviations

BSO — Basic service operators

CAC — Carrier access code

CC — Country code

CIC — Carrier identification code

ICIC — International carrier identification code

ILD — International long distance

LDCA — Long distance number charging area

LDCC — Long distance charging centre

NDC — National destination code

NLD — National long distance

NLDO — National long distance operators

NSN — National significant number

POI — Point of interconnect

SDCA — Short distance number charging area

SDCC — Short distance charging centre

SN — Subscriber number

TAX — Trunk automatic exchange

TC — Trunk code

LDCA, SDCA, NDA, SN and NSN: Broadly, our country is divided into eight regions with

each region being identified by a single digit code as shown in Fig. The above said notations

are generally used in telecommunications and thus discussed to some extent.

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Fig. Numbering plan in India

LDCA.

Long distance charging centre comprises of one or several SDCA’s. There are at

present 322 LDCA’s. Each LDCA has a long distance charging centre (LDCC) which

is a Trunk Automatic Exchange (TAX).

SDCA.

There are 2645 SDCA’s. Each SDCA is allocated a unique trunk code. Each

SDCA has one or more number of exchanges. Therefore, there are 2645 codes

required to identity the complete country based on SDCA linked numbering scheme.

The length of the trunk codes shall vary from 2 to 4 digits.

SDCA has a short distance charging centre (SDCC). SDCC is an integrated local cum

tandom or a transit switch. The size of SDCA generally varies between 800 sq. kms.

to 2000 sq. kms.

NPA.

Numbers in an Numbering Plan Area (NPA) are not duplicated and called

‘subscriber numbers’.

To make a call from one subscriber to another subscriber in the same NDA, only the

subscriber number needs to be dialled. At present NDA is same as SDCA.

SN.

Subscriber number (SN) is a 6 to 8 digit number which includes 2 to 4 digit

Telephone exhcnage code.

NSN.

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The numbering scheme envisages the use of SDCA based linked numbering with 10

digit National Significant Number (NSN). The variants of SNS is shown in Table.

Table Variants of SNS

National Numbering Scheme.

There are ten levels of numbering schemes starting from 0 to 9. In each level, there

are many sublevels as a classification of services. All the levels are defined briefly

and some sublevels are explained in the following paragraph. Level 0.

Prefix codes. There are various sublevels. Some are described. Sublevel ‘000’ as a

prefix shall be used for home country direct service (Bilateral) and international toll

free service (Bilateral).

The format is 000 + country code + operator code ‘000800’ is used for bilateral

international toll free service. Sublevel ‘00’ as a prefix shall be used for international

dialling. The format as per E.164 recommendation is 00 + country code + NSN.

Sublevel ‘0’ as a prefix shall be used for national long distance calls. The format is 0

+ SDCA code + subscriber number the sublevel ‘09’ is used for cellular mobile

services, satellite based services and Intelligent Network (IN) Services.

Level 1. Special services.

This level is used for accessing special services like emergency services,

supplementary services, inquiry and operator assisted services. The format contains 3

to N digit depending on service.

Level 2 to 6. PSTN subscriber number. This format contains the telephone exchange code

and subscriber number.

Level 7 and 8. These two levels not being allocated and the same are reserved for new

services.

Level 9. Services. The range of numbers in level ‘9’ except ‘90’, ‘95’ and ‘96’ (‘96’ is used

in paging services) are reserved for cellular mobile services. Starting from 90 to 99, there are

9 sublevels. ‘90’ is used as spare not allocated for any service.

Numbering format :

The PSTN numbering format shall be as per the table given below :

Table PSTN numbering format

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The trunk code is 2 to 4 digits. The telephone exchange code and last n digits of

subscriber number together called subscriber number and is from 6 t0 8 digits. Hence

national number to call a subscriber is 8 to 12 digits. To call a subscriber in another SDCA,

prefix ‘0’ must be dialled first.

SDCA code.

Digit A can have any value from 1 to 8. Digit B, C and D can have any value between

0 to 9. AB codes can have any value between 11 to 89 (79 trunk codes).

The code 10 is earmarked for carrier access code for NLD service and ILD service.

Code 11, 20, 22, 33, 40, 44, 79 and 80 (8 codes) are used presently. 39, 50, 60, 69 and

70 (5 codes) are available for allotment to SDCA’s with 3-digit code depending on

the requirement. Certain three digit spare codes like 555, 666 and 888 are not to be

used as SDCA codes. These are reserved for future services.

Telephone exchange code.

Digit E can have any value between 2 to 6. The 0, 1, 7, 8,

9 are not allowed as ‘0’ is used as trunk prefix, level ‘1’ is used for special services, 7 and 8

are kept spare for future services and level ‘9’ is used for cellular mobile services, paging

services and access to adjacent areas. Digit F, G and H can have any value from 0 to 9. EFG

can take value between 200 and 699 (500 exchange codes).

Last n digits of subscriber number.

Digit P, Q, R, S and T can have any value from 0 to 9.

Dialling.

For a call within a local area i.e. SDCA, subscriber number (Telephone exchange

code + last n digits of subscriber code) only need to be dialled.

Thus the number of digits needed to be dial is 6 to 8. For calls outside the SDCA, 0 +

NSN is need to be dialled (NSN = SDCA code + Telephone exchange code + last n

digit of subscriber number). However, access to adjacent areas can also be possible

by dialling ‘95’ followed by NSN.

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Charging Plan

The cost of providing a telecommunication network consists of the capital

cost and the current operating expanses.

The capital cost includes switching systems, buildings, lines and land.

Operating cost includes staff salaries, maintenance costs, water and electricity

charges and miscellaneous expenses.

The telecom operator charges the subscribers for its services by the following

three ways.

An initial charge for providing a network connection (as installation charges)

A rental or leasing charge

Call charges

The initial costs are covered partly from installation charges and partly from rental. The

operating costs of the telephone exchange are recovered through rental and call charges.

According to the government policy, the rental may be levied on a monthly, bimonthly or

by some other modes.

The quantity of equipments used, routing exchanges, switching systems, lines

carrying voice/data and human involvement in establishing a connection between

subscribers differs with respect to the distance between the subscribers, the time

at which the call is made (at busy hour or off peak hour), the area (business or

residential) etc. The charging methods for individual calls fall under two broad

categories.

Duration independent charging

Duration dependent charging

Traditionally, charges for long distance calls have been proportional to distance

multiplied with duration. The local calls within a numbering area are usually charged on a

duration independent basis.

A meter for each subscriber counts the number of charging units based

on the service providers policy decision.

As the billing procedure changes time to time according to the Govt.

policies and to meet service providers expenses, the charging plans cannot be

explained. The readers can refer the day to day changes through websites or

newspapers.

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Signalling Techniques

A subscriber can be able to talk with or send data to someone in any part of

the world almost instantly and an exchange is able to set path and clear it after

the conversation instantly by an effective signalling system.

A signalling system link the variety of switching system, transmission systems

and subscriber equipments in a telecommunication network to enable the

network to function as a whole.

The signalling are classified according to the internal signalling of an

exchange, signalling between exchanges and signalling between an exchange

and subscriber. Thus a signalling system must be obviously be compatible

with the switching systems which itself partitioned into subsystems in a

network.

Traditional exchanges sent signals over the same circuit in the network. The

introduction of SPC in exchanges enhances the services and introduced new

services to the subscriber. These services require more signals to be

transmitted and hence needs a separate data channel. The former method of

signalling is referred as channel associated signalling and the latter is common

channel signalling.

In the following sections, different forms of signalling, types of signalling,

various types of inchannel signalling, common channel signalling, networks

and the world wide popular signalling system 7 (SS7) are described is shown

in figure.

Signalling Classifications

Communication networks generally connect equipments such as telephones

and fax machines via several line sections, switches and transmission media

for exchange of speech, text and data.

To achieve this, control information has to be transferred between exchanges

for call control. Call control is the process of establishing and releasing a call.

This is referred as signalling.

In general, signalling is defines as follows. ‘‘Signalling is the process of

generating and exchanging information among components of a

telecommunication system to establish, monitor or release connections and to

control related network and system operations’’.

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Forms of Signalling

The information that must be transmitted between subscribers, and between switching

centres falls broadly under three classes.

Supervisory signals or line signals. These are the signals necessary to initiate a call

setup and to supervise it, once it has been established. It is also referred as subscriber

loop signalling. Line signals can be transmitted by the use of a single control channel

in each direction line.

Routing signals or register signals. Information transfer related to call setup is

usually referred to as register signals. The basic information is the dialled code which

indicates to the subsequent switching centres the required routing. In addition to the

basic information, signals such as route information, terminal information, register

control signals, acknowledgement signals, status of called terminal etc are also

involved.

Management signals or interregister signalling . These signals are used to convey

information or control between exchanges. This signalling also referred as inter

exchange signalling. This signalling involves remote switching of private circuits,

routing plans, modification of routing plans, traffic over load, priority of the call,

class of service etc. The signaling may be performed by link-by-link basis, which

passes signals exchange to exchange or end-toend signalling which is between

originating and terminating exchange referred as line signalling.

Classification

Traditional signalling uses the same channel to carry voice/data and control signals to

carry out the path setup for speech or data transfer.

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Signalling

Inchannel Common channel

DC Low frequency Voice frequency PCM

Inband Outband

Associated Non-associated

Fig. Signalling techniques

This signalling is referred as in channel signaling or per trunk signalling (PTS). An

alternative to inchannel signalling is called as common channel signalling (CCS).

CCS, uses a separate common channel for passing control signals. It couples the

signals for a large number of calls together and send them on a separate signalling

channel.

Hence basically, signalling technique is classified into

In channel signalling/per trunk signalling (PTS)

Common channel signalling (CCS)

The inchannel signalling is classified further into four categories as shown in Fig..

Fig. Inchannel signalling

The common channel signalling (CCS) is classified according to the transmission of signals

between exchangers. They are:

Associated signalling

Non-associated signalling

Quasi associated signalling

Common Channel Signalling

Introduction of SPC digital switching systems with high speed processors in

the telecom network has necessiated modernisation of signalling.

Also, in order to meet the transfer of varieties of informations for call

management and network management and to satisfy the subscribers

requirement on various features, the uninterrupted, high speed signalling has

become inevitable. The rapid dvelopment of digital systems paved way for the

new signalling system called common channel signalling.

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Instead of using the same link for signalling information and message as in

Inchannel signalling, the common channel signalling (CCS) uses a dedicated

line for the signaling information between the stored program control

elements of switching systems. Fig. shows the basic schematic of CCS.

Fig. Basic schematic of CCS.

The data link sends messages that identify specific trunks and events. Two

signaling channels, one for each direction are used in a dedicated manner to

carry signalling information. Hence, they are capable of carrying information

for a group of circuits.

At the bit rate of 2.6 kbps, CCS can carry signals for 1500–2000 speech

circuits. CCS network is basically a store and forward network. In CCS

network, the signalling information travels in a link-by-link basis along the

route.

The information arrived at a node is sotred, processed and forwarded to the

next node in the route. The CCS technique is also called the transparent mode

for signalling.

CCS Signalling Message Formats

CCS signalling information is transferred as signalling units, which is of

varying length with one or more fixed length.

A signalling unit is divided into a number of fields. The fields may be address

unit, centralised service message unit, acknowledgement unit, synchronisation

or idle unit, management message unit etc.

The signalling information includes data related to routing, addressing digits,

inter exchange information, routing status to the originating exchange

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maintenance request/details etc. Fig. shows the typical CCS signalling

message format.

Fig. Basic CCS signalling message formats.

CCS Network

CCS network improved the performance of the existing network and also

established a plat from for the introduction of new facilities.

The first CCS of AT & T were installed in the toll network between a No. 4 A

cross bar switch in Madison, Wisconsin and a No. 4 ESS switch in Chicago,

Illinois in 1976. The CCS network links between SPC switching offices

increases the speed of long distance call connect timings and reduces the cost

compared to the inchannel SF/ MF signalling facilities. Fig. shows a typical

CCS network.

Fig. CCS basic structure.

The CCS network consists of two types of nodes. They are signal transfer

points (STP) and Signal control points (SCP).

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These nodes are interconnected by signalling links. The original

communication protocols used between CCS entities was CCITT signalling

system No. 6 (CCS 6). The modified protocol introduced in 1980 by CCITT is

signalling system No. 7.

The STP’s are the packet switching nodes of the CCS network. They receive

and route incoming signalling messages towards the proper destination. They

also perform specialized routing functions.

Signal control points (SCP) are data bases that provide information necessary

for advanced call processing capabilities. SCP also serves how to route calls,

verify credit cards, process special services etc. The same basic structure is

also installed within local Access Transport Area (LATA)’s to extend CCS

feature all the way to end offices or local exchanges (LE).

CCS Implementation

Common channel signalling may be implemented in three ways. They are (a)

Channel Associated mode (b) Channel Non-Associated mode and (c) Quasi-

Associated mode. Fig. shows all the three modes of CCS signalling.

In associated CCS signalling mode, there is a direct link between two

exchanges. In this mode, the signalling path passes through the same set of

switches as does the speech path. Network topologies of the signalling

network and the speech network are the same. This mode of operation is

simple, economic and easy to control. This involves in delayed operation for

long distance communication.

In non-associated CCS signalling, there are separate control of the networks

from the switching machines themselves. In multiexchange network, signal

message passing through several intermediate nodes is referred as non-

associated signalling.

The network topologies for the signalling and the speech networks are

different. Between exchanges, many STP’s are placed. This approach is

flexible as far as the routing is concerned. It demands more comprehensive

scheme for message addressing than is needed for channel associated

signalling.

In practice, CCS messages are routed through one intermediate node for short

distance communication. This is known as quasi-associated signalling. It establishes

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simplified predetermined paths between exchanges. The signalling paths are not associated

but are fixed for given speech connections

Advantages and Disadvantages of CCS

Advantages. The major advantages of CCS are listed below:

1. For each associated trunk group, only one set of signalling facility is required. The channel

used for CCS need not be associated with any particular group. Fig. 3.2 shows a CCS

network that is disassociated from the message network structure.

2. The introduction of SPC switching machines and CCS provides efficient routing

procedure.

3. CCS allows for signalling at any time in the entire duration of a call, not only at the

beginning.

4. CCS removes most of the signalling costs associated with inter office trunks.

5. Information can be exchanged between processor at high speed.

6. The CCS provides acceptable quality for network related signalling tones such as DTMF,

MF and SF achieved.

7. As there is no need of line signalling equipment on every function, considerable cost

savings can be achieved.

8. The D1 channel bank use 1 bit per time slot for signalling and 7 bits for voice which

provides signalling rate 8 kbps. D2 channel provides higher data rate which use 1 bit in every

sixth frame for signalling. This is referred as ‘‘robbed bit signalling.’’

When CCS is utilized, the associated T carrier system no longer need to carry signaling

information on a per channel basis and a full 8 bits of voice can be transmitted in everytime

slot of every frame.

9. As separate channels are used for voice and control. There is no chance of mutual

interference and the error rate is very low.

10. CCS enables more services to the subscribers. A signalling link operating at 64 k bit/s

normally provides signalling up to 1000 or 1500 speech circuits.

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Fig. Three modes of CCS signalling

Disadvantages of CCS:

Some disadvantages of CCS are listed below:

1. The CCS network is basically a store and forward network. So, in a established circuit,

the signalling informations are stored, processed and then forwarded to next node. This

causes

additional overhead and disconnect to the continuity.

2. If one node fails to transmit properly, the facilities downstream from the disconnect

will not be released. Thus, a high degree of relability is required for the common channel.

3. Proper interfacing facility is necessary, as most of the present day telephone networks

are equipment with inchannel signalling systems.

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4. As the signalling information is not actually sent over speech paths in CCS, the integrity

of speech path is not assured. As a remedy, routing testing of idle paths and the continuity

test

of an established path become necessary in CCS.

5. Different trunks in a group may terminate at different switch, say local exchange,

other foreign exchange circuits etc. With CCS, all trunks are first terminated to the local

central office and then forward to the different destination.

.

In channel and common channel signalling comparison is tabulated in Table 3.1

UNIT IV

Data Transmission in PSTN

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The transmission medium is the physical foundation for all the data

communications. The amount of data carried across the networks crossed the

voice traffic level.

In the beginning, data transmission was organized using telegraph or telex

networks as they could carry digital signals directly. But teletype machines

were slow, noisy and consumed large amounts of power. The speed was

limited to 110 bauds. (Baud rate is a measure of the rate at which binary data

are transmitted and received).

But data rates for transmission have been on the rise. With public switched

telephone network, there is a possibility of carrying signals at higher speeds.

Public switched telephone networks and electronic PABX’s are designed to

carry analog voice signals. They can be used for data transmission by

employing suitable interfaces.

Data Rates in PSTN

Baud rate.

The maximum rate of signal transitions that can be supported by a channel is known

as baud rate. Baud rate is a close measure of information throughput, or the effective

information data transfer rate from sender to receiver.

Thus, baud rate is one that can be supported in a noiseless channel. We know, a voice

channel in a PSTN is band limited with a nominal bandwidth of 3.1 kHz. A

maximum data rate that a noiseless or ideal voice channel can support can be

obtained from the Nyquist theorem

D = 2 B log2 L bps

where D = Maximum data rate (in Baud or bps)

B = Bandwidth of the channel

L = Number of discrete levels in the signals.

For a 3 kHz channel, and a binary signal, the maximum data rate is 6000 bps,

if the signal level is two.

For higher data rates, we translate information rate into symbols per second. A

symbol is any element of an electrical signal that can be used to represent one

or more binary data bits. The rate at which symbols are transmitted is the

symbol rate. This rate may be represented as a systems baud rate.

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Fig. illustrates the pulse representation of the binary numbers used to code the samples

In Fig. shown, each three digit binary number that specifies a quantized sample value

is called a word. Cg is called guard time between pulses.

Bit rate.

In the noisy channel, there is an absolute maximum limit for the bit rate. This limit

arises because the difference between two adjacent signal levels become comparable

to the noise level when the number of signal level is increased. For noisy channel,

data rate is calculated by

Db = B log2 (1 + S/N)

Where Db = Data rate in noisy channel (in bps)

B = Bandwidth of the channel

S/N = Signal to noise ratio.

For S/N of 30 dB and 3 kHz Bandwidth noisy channel, Db is 30000 bps.

Relation between baud rate (or symbol rate) and bps : The baud rate and bit rate are

related as

Db = D × n

where n = number of bits required to represent signal levels.

In the example considered for baud rate explanation n is assumed as one. Hence baud rate is

equal to bps. Fig. 1.2 illustrates the relation between baud and bit rates.

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Above Fig. (a) shows the baud rate equal to bit rate. Fig. (b) and (c) shows the baud

rate equal to one-half and one-fourth of bit rate respectively. It is proved that up to

2400 bauds may be transmitted reliably through a PSTN voice channel. By increasing

the signal levels, the effective bit rate increases.

For low-speed applications, the difference between baud and bit rate are insignificant.

Thus 300 and 1200 bps modems orginally used with personal computers were frequently

referred to as 300 or 1200 baud modems.

Data Communications Link

In order to communicate from a terminal, computer or any equipment, the following six parts

have to be put together in proper order.

The transmission medium that carries the traffic between source and destination.

Data communication equipment or data circuit terminating equipment (DCE).

Data terminal equipment (DTE).

Communication protocols and software.

Terminal devices.

Interface.

Fig. below shows the typical arrangement of the communication link for the data

communication. Data link refers to the process of connecting or linking two stations together.

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Fig. The typical arrangement of the communication link for the data communication

Transmission medium. The transmission medium include communication channels, path,

links, trunks and circuits. The transmission medium may be a telephone lines, coaxial cable,

twisted pair, Fiber cable, radio waves (free space), microwave link or satellite link.

Terminal devices. These are the end points in a communication link. Terminal devices are

also called as nodes. For the two point network, the node points are the primary station and

the remote or secondary station. A primary station is responsible for establishing and

maintaining the data link between it and a secondary station. The terminal devices includes

main frame computer, personal computer, peripherals such as printers, keyboards, FAX

machines and data display terminals.

Data terminal equipment (DTF). The terminal devices, communication station, UART, and

line control unit (LCU) grouped together and named as DTE. Fig. 11.4 shows typical

arrangement of DTE.

Connection-Oriented and Connectionless Services

Layers can offer two different types of service to the layers above them:

connection-oriented and connectionless. In this section we will look at these

two types and examine the differences between them.

Connection-oriented service is modeled after the telephone system. To talk to

someone, you pick up the phone, dial the number, talk, and then hang up.

Similarly, to use a connection-oriented network service, the service user first

establishes a connection, uses the connection, and then releases the

connection.

The essential aspect of a connection is that it acts like a tube: the sender

pushes objects (bits) in at one end, and the receiver takes them out at the other

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end. In most cases the order is preserved so that the bits arrive in the order

they were sent.

In some cases when a connection is established, the sender, receiver, and

subnet conduct a negotiation about parameters to be used, such as maximum

message size, quality of service required, and other issues. Typically, one side

makes a proposal and the other side can accept it, reject it, or make a

counterproposal.

Each service can be characterized by a quality of service. Some services are

reliable in the sense that they never lose data. Usually, a reliable service is

implemented by having the receiver acknowledge the receipt of each message

so the sender is sure that it arrived. The acknowledgement process introduces

overhead and delays, which are often worth it but are sometimes undesirable.

A typical situation in which a reliable connection-oriented service is

appropriate is file transfer. The owner of the file wants to be sure that all the

bits arrive correctly and in the same order they were sent. Very few file

transfer customers would prefer a service that occasionally scrambles or loses

a few bits, even if it is much faster.

Reliable connection-oriented service has two minor variations:

message sequences and byte streams. In the former variant, the message

boundaries are preserved.

As mentioned above, for some applications, the transit delays

introduced by acknowledgements are unacceptable. One such application is

digitized voice traffic. It is preferable for telephone users to hear a bit of noise

on the line from time to time than to experience a delay waiting for

acknowledgements. Similarly, when transmitting a video conference, having a

few pixels wrong is no problem, but having the image jerk along as the flow

stops to correct errors is irritating.

Not all applications require connections. For example, as electronic mail

becomes more common, electronic junk is becoming more common too.

The electronic junk-mail sender probably does not want to go to the trouble of

setting up and later tearing down a connection just to send one item. Nor is

100 percent reliable delivery essential, especially if it costs more. All that is

needed is a way to send a single message that has a high probability of arrival,

but no guarantee. Unreliable (meaning not acknowledged) connectionless

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service is often called datagram service, in analogy with telegram service,

which also does not return an acknowledgement to the sender.

In other situations, the convenience of not having to establish a connection to

send one short message is desired, but reliability is essential. The

acknowledged datagram service can be provided for these applications.

It is like sending a registered letter and requesting a return receipt. When the

receipt comes back, the sender is absolutely sure that the letter was delivered

to the intended party and not lost along the way.

Still another service is the request-reply service. In this service the sender

transmits a single datagram containing a request; the reply contains the

answer.

For example, a query to the local library asking where Uighur is spoken falls

into this category. Request-reply is commonly used to implement

communication in the client-server model: the client issues a request and the

server responds to it. Below fig summarizes the types of services discussed

above.

Switching Techniques for Data Transmission

This section describes various techniques used to establish connections

between users exchanges. Switches are hardware and/or software devices used

to connect two or more users temporarily.Message switching, circuit

switching and packet switching are the most important switching methods.

The terminals of the message switching systems are usually teleprinters. In

this switching, delays are incurred but no calls are lost as each messages are

queued for each link.

Thus much higher link utilisation is achieved. The reason for the delay is that

the system is designed to maximise the utilisation of transmission links by

queueing message awaiting the use of a line. This switching is also called

store and forward switching.

The circuit switching sets up connection between the telephone, telex

networks etc. which interchange information directly. If a subscriber or

system to which connection to be made as engaged with other connection,

path setup cannot be made. Thus circuit switching is also referred as lost call

system.

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The modified form of message switching is called packet switching. Packet

switching system carries data from a terminal or computer as a short packets

of information to the required destination. This system is midway between

message switching and circuit switching.

Message Switching

In message switching, the messages are stored and relayed from secondary

storage. So, message switching is best known by the term store and forward.

In message switching, there is no direct link between the sender and the

receiver.

A message delivered to the destination is rerouted along any path before it

reaches the destination. It was common in 1960’s and 1970’s. Typical

message switching network is shown in Fig.

Fig. Message Switching

Message switching offers the possibility of greatly imporved economy. The

working of message switching is as follows.

Source sends message M1 to the destination. Suppose that the transmission

path selected is A––C––D. In message switching no complete connection is

required. Thus the each message includes a header contains the destination

address, routing information and priority information (for special cases).

The node A will transmit message M1 to switching centre C. Where it will be

stored in a buffer. The processor in each node maintains message queues for

each outgoing link. These queues are normally serviced on a first come, first

served basis. In the header have priority information, the message will be

passed earlier.

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Then, after storage and possible delay, the message will go from node C to D

and then to the destination. To store the messages, large capacity storage

media is necessary at each node.

In increases the node cost high. Also, most message switching networks could

deliver a message on a delayed basis if destination node is busy or otherwise

unable to accept traffic. Also, the message switching was used in unintelligent

devices such as telegraphy. Since these devices has been replaced by the high

speed intelligent devices, this switching is unpopular for direct

communications.

In message switching network, the transmission links are never idle.

Utilization of transmission link of a message switching network is directly

related to the actual flow of information. With increased store and forward

queuing delays, utilization efficiency can be increased.

Circuit Switching

Circuit switching creates a direct physical connection between two devices

such as phones or computers. In order to setup a direct connection over many

links it is necessary that each link to be simultaneously free.

This implies that the average utilization of the links must be low if the

probability of demand for connection is more. It is therefore used in voice

networks mainly and not in networks designed for data transfer. A circuit

switch is a device with n inputs and m outputs that creates a temporary

connection between source and destination.

The inputs n and outputs m need not be equal. In order to transmit

information, a circuit switched network finds a route along which it has free

circuits. The network connects the circuits together and reserves them for the

transmission. Fig. illustrates the circuit switching.

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Fig. Circuit Switching

In the Fig. shown, the end device A is connected to the end device D through

switches II, III and V. Circuit switching involves three phases. First, the

source, request the network for the route.

The network assigns a route. Second, data transfer now occurs, the duration of

the data transfer is called holding time. Third, once the data transfer is

competed, the path setup is disconnected.

By moving the level of switches, any end devices can be connected to any

other end devices. Circuit switching is usually accomplished by TDM.

As the data transfer takes place in three phases, the time taken for the data transfer (T)

is expressed as

T = Tp + Td + Tr

where Tp = Path setup time (N – 1) Trs

Td = data transfer time = M/R

Trs = average route selection time

Tr = data release time = NTn

N = Number of switches in the path

M = Message length in bits

R = data rate in bits per sec

Th = house keeping entries time.

Thus

T = (N – 1) Trs + M/R = NTh

The propagation time is not considered as it is comparatively very small. In our case N = 3, If

Trs = 2 sec, Tn = 2 sec, R = 2400 bps and the message is 300 bytes long, the time for the data

transfer is T = (3 – 1) × 2 + 300 × 8/2400 + 3 × 0.2 = 4 + 1 + 0.6 = 5.6 sec.

Comparison of Circuit Switching and Message Switching

Message switching Circuit switching

The source and destination do not

interact in real time

Message delivery is on delayed basis if

destination node is busy or otherwise

unable to accept traffic.

The source and destination are

connected temporarily during data

transfer.

Before path setup delay, may be there

due to busy destination node. Once the

connection is made the data transfer

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Destination node status is not required

before sending message.

Message switching network normally

accepts all traffic but provides longer

delivery time because of increased

queue length.

In message switching network, the

transmission links are never idle.

takes place with negligible

propagation time.

Destination node status is necessary

before setting up a path for data

transfer.

A circuit switching network rejects

excess traffic if all the lines are busy.

In circuit switching, after path setup, if

the users denied service, the line will

be idle. Thus, the transmission

capacity will be less, if the lines are

idle.

Packet Switching

The datastream originating at the source is divided into packets of fixed or

variable size. The data communication system typically have bursty traffic.

Thus, the time interval between consecutive packets may vary, depending on

the burstiness of the data stream.

A typical upper bound on packet length is 512 octets (bytes). Each packet

contains a portion of the user’s data plus some control information. As the bits

in a packet arrive at a switch or router, they are read into a buffer.

When the entire packet is stored, the switch routes the packet over one of its

outgoing links. The packet remains quenced in its buffer until the outgoing

link becomes idle. This technique is called store and forward technique. Fig.

illustrates the flow of packet switching.

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Fig. Packet Switching

In Fig. two packets (P1, P2) are entering at station A and B. The packet (P1)

entering at A is targeted to the station H and the packet (P2) entering at B is

targeted to the station G.

Depends on the path availability the packet P1 chooses the path as station A–

C–E– H. Similarly the packet P2 chooses the path as station B–C–D–G. In

each station, the packets are stored in a buffer and forwarded to the next

station after the availability.

Routing Control

From the previous section, it is clear that in packet switching, messages are

broken into packets and sends one at a time to the network.

Routing control decides how the network will handle the stream of packets as

it attempts to route them through the network and deliver them to the intended

destination. The routing decision is determined in one of two ways. They are

Datagram and

Virtual circuit.

Datagram.

In datagram, each packet within a stream is independently routed. A routing table

stored in the router (switch) specifies the outgoing link for each destination. The table

may be static or it may be periodically updated. In the second case, the routing

depends on the router’s estimate of the shortest path to the destination.

Since the estimate may change with time, consecutive packets may be routed over

different links. Therefore each packet must contain bits denoting the source and

destination. Thus may be a significant overhead. Fig. shows a simple communication

network where the concept of datagram is explained. The circled one are called the

switching nodes whose purpose is to provide a switching facility that will move the

data from node to node until they reach the destination.

The squared one are called the stations. The stations may be computers, terminals,

telephones or other communication devices. These stations also referred as end

devices are the communicating devices.

In the datagram approach, each packet is treated independently. In the Fig. shown, the

station A is assumed to send three packets of message namely P1, P2 and P3 (for

explanation purpose named so).

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At first, A transmits these packets to node 1. Node 1 makes decision on routing of

these packets. Node 1 finds node 4 as shortest compared to node 3.

Fig. Datagram concept

Thus it passes P1 and P2 to node 4. Accidently, if node 4 is not accessible,

node 1 finds node 3 as shortest and sends packet P3 to node 3. Node 3 and 4

sends its received messages to the destination C through node 6. It is shown

that the order of the packet is changed due to the different routing of the

packets.

Thus in datagram, it is the responsibility of destination station to reorder the

packets in proper sequence. Also if a packet crashes in a switching node, the

destination C may not receive, all packets. In such a case also, it is the

responsibility of station C to recover the lost packet.

Virtual circuit.

In virtual circuit, a fixed route is selected before any data is transmitted in a call setup

phase similar to circuit switched network. All packets belonging to the same data

stream follow this fixed route called a virtual circuit. Packet must now contain a

virtual circuit identifier.

This bit string is usually shorter than the source and destination address identifiers

needed for datagram. Once the virtual circuit is established, the message is

transmitted in packets. Fig. shows the concept of virtual circuit.

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Fig. Virtual circuit concept

In the Fig. shown, suppose that end station B has two messages to send to the

destination D. First B sends a control packet referred as call-request packet to

node 2, requesting logical connection to D.

Node 2 decides to route the request and the subsequent message packets

through node 3 and 4 to destination D. If D prepared to accept the connection,

it sends a call-accept packet to node 4. Node 4 sends the call-accept packet to

B through node 3 and 2.

Because the route is fixed for the duration of the logical connection, it is

somewhat similar to a circuit switching network and is referred to as a virtual

circuit.

Every data packet with virtual circuit identifier and data from B intended for

D traverses node 2, 3 and 4. Similarly every data packet from D intended for

B traverses nodes 4, 3 and 2.

Any station can terminate the connection with a clear-request packet. At any

time, each station can have more than one virtual circuit to any other station

and can have virtual circuits to more than one station. As all packets follow

the same route, they reach the destination in the same order. So there is no

need of recording work for destination station. Error control is the additional

advantage of virtual circuit.

Error control is a service that assures error free reception. For example, if a

packet in a sequence from node 3 to 4 fails to arrive at node 4, or arrives with

an error, node 4 can request a retransmission of that packet from node 4. As

there is no necessity of routing decision during transition of packets from

source to destination, with virtual circuit, packets transit rapidly. Currently

available packet switching networks make use of virtual circuits for their

internal operations.

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Packet size.

If an organization has large amounts of data to send, then the data can be delivered to

a packet assembler/disassembler (PAD). The PAD (software package) receives the

data and breaks it down into manageable packets. In the data communication, a

packet can be a variable length.

Usually up to 128 bytes of data is in one packet. X-25 services have created packets

up to 512 bytes, but the average is 128. The 128 byte capability is also referred to as

fast select. There is a significant relationship between packet size and transmission

time.

The process of using more smaller packets (for example 30 byte information may be

sent as a single packet with header of 3 byte or two packets with 15 byte each plus the

header in each packet or 5 packets with 6 bytes plus header) increases the speed of

transmission.

Comparison of Circuit Switching and Packet Switching

There are two types of approaches in packet switching. Datagram and virtual circuit.

The circuit switching is compared with these two approaches. Datagram switching achieves

higher link utilization than circuit switching especially when the traffic is bursty. No

dedicated path is required as circuit switching. But the datagram have the disadvantage over

virtual circuit wire.

End to end delay may be so large or so random as to preclude applications that

demand guaranteed delay.

The overhead due to source and destination identifiers and bits needed to delimit

packets may waste a significant fraction of the transmission capacity if the packet are

very short.

A datagram switch does not have the state information to recognize if a packet

belongs to a particular application. Hence the switch cannot allocate resources

(bandwidth and buffers) that the application may require.

Virtual circuits are more advantages and currently the packet switching network uses

the virtual circuit approach. The overhead is comparable to circuit switching. As the

packets arrive in sequence, no resequencing is needed.

The OSI Reference Model

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Fig. The OSI reference model

The OSI model has seven layers. The principles that were applied to arrive at the

seven layers can be briefly summarized as follows:

A layer should be created where a different abstraction is needed.

Each layer should perform a well-defined function.

The function of each layer should be chosen with an eye toward defining

internationally standardized protocols.

The layer boundaries should be chosen to minimize the information flow across the

interfaces.

The number of layers should be large enough that distinct functions need not be

thrown together in the same layer out of necessity and small enough that the

architecture does not become unwieldy.

Below we will discuss each layer of the model in turn, starting at the bottom layer.

Note that the OSI model itself is not a network architecture because it does not

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specify the exact services and protocols to be used in each layer. It just tells what

each layer should do. However, ISO has also produced standards for all the layers,

although these are not part of the reference model itself. Each one has been published

as a separate international standard.

The Physical Layer

The physical layer is concerned with transmitting raw bits over a communication

channel. The design issues have to do with making sure that when one side sends a 1

bit, it is received by the other side as a 1 bit, not as a 0 bit.

The design issues here largely deal with mechanical, electrical, and timing interfaces,

and the physical transmission medium, which lies below the physical layer.

The Data Link Layer

The main task of the data link layer is to transform a raw transmission facility

into a line that appears free of undetected transmission errors to the network

layer.

It accomplishes this task by having the sender break up the input data into

data frames (typically a few hundred or a few thousand bytes) and transmit the

frames sequentially. If the service is reliable, the receiver confirms correct

receipt of each frame by sending back an acknowledgement frame.

Another issue that arises in the data link layer (and most of the higher layers

as well) is how to keep a fast transmitter from drowning a slow receiver in

data.

Some traffic regulation mechanism is often needed to let the transmitter know

how much buffer space the receiver has at the moment. Frequently, this flow

regulation and the error handling are integrated.

Broadcast networks have an additional issue in the data link layer: how to

control access to the shared channel. A special sublayer of the data link layer,

the medium access control sublayer, deals with this problem.

The Network Layer

The network layer controls the operation of the subnet. A key design issue is

determining how packets are routed from source to destination. Routes can be

based on static tables that are ''wired into'' the network and rarely changed.

They can also be determined at the start of each conversation, for example, a

terminal session (e.g., a login to a remote machine). Finally, they can be

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highly dynamic, being determined anew for each packet, to reflect the current

network load.

If too many packets are present in the subnet at the same time, they will get in

one another's way, forming bottlenecks. The control of such congestion also

belongs to the network layer. More generally, the quality of service provided

(delay, transit time, jitter, etc.) is also a network layer issue.

When a packet has to travel from one network to another to get to its

destination, many problems can arise. The addressing used by the second

network may be different from the first one. The second one may not accept

the packet at all because it is too large. The protocols may differ, and so on. It

is up to the network layer to overcome all these problems to allow

heterogeneous networks to be interconnected.

In broadcast networks, the routing problem is simple, so the network layer is

often thin or even nonexistent.

The Transport Layer

The basic function of the transport layer is to accept data from above, split it

up into smaller units if need be, pass these to the network layer, and ensure

that the pieces all arrive correctly at the other end.

Furthermore, all this must be done efficiently and in a way that isolates the

upper layers from the inevitable changes in the hardware technology.

The transport layer also determines what type of service to provide to the

session layer, and, ultimately, to the users of the network.

The most popular type of transport connection is an error-free

point-to-point channel that delivers messages or bytes in the order in which

they were sent.

The transport layer is a true end-to-end layer, all the way from

the source to the destination. In other words, a program on the source machine

carries on a conversation with a similar program on the destination machine,

using the message headers and control messages.

In the lower layers, the protocols are between each machine

and its immediate neighbors, and not between the ultimate source and

destination machines, which may be separated by many routers. The

difference between layers 1 through 3, which are chained, and layers 4

through 7, which are end-to-end, is illustrated in Fig.

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The Session Layer

The session layer allows users on different machines to establish sessions

between them.

Sessions offer various services, including dialog control (keeping track of

whose turn it is to transmit), token management (preventing two parties from

attempting the same critical operation at the same time), and synchronization

(checkpointing long transmissions to allow them to continue from where they

were after a crash).

The Presentation Layer

Unlike lower layers, which are mostly concerned with moving bits around, the

presentation layer is concerned with the syntax and semantics of the

information transmitted.

In order to make it possible for computers with different data representations

to communicate, the data structures to be exchanged can be defined in an

abstract way, along with a standard encoding to be used ''on the wire.'' The

presentation layer manages these abstract data structures and allows higher-

level data structures (e.g., banking records), to be defined and exchanged.

The Application Layer

The application layer contains a variety of protocols that are commonly

needed by users. One widely-used application protocol is HTTP (HyperText

Transfer Protocol), which is the basis for the World Wide Web. When a

browser wants a Web page, it sends the name of the page it wants to the server

using HTTP. The server then sends the page back.

Other application protocols are used for file transfer, electronic mail, and

network news.

Satellite Based Data Networks

Terrestrial networks such as fiber, DSL and leased lines offer broadband

connectivity for data networks but the coverage of these networks is typically

restricted to densely populated urban regions with limited, or no coverage in

the remote and rural regions.

This is an obstacle for organizations that have geographically dispersed

locations as it means that the sites that out of the reach of the terrestrial cannot

be connected.

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The flat architecture of VSAT networks introduces advantages for private

networks:

Uniformity the same level of service can be provided to all of the customer’s sites regardless

of their geographic location.

Security customer sites can communicate with each other directly (the transmissions are sent

up to the satellite and bounced directly back down to the receiving end) and do not need to

traverse through a network core along with other customer’s traffic .

Bandwidth utilization the network bandwidth available for the customer is shared between

all of their sites and it allocated automatically wherever it is needed. There is no “last -mile

bottleneck” as there is in terrestrial networks.

Rapid expansion new sites can be added instantaneously at any location without any

infrastructure pre-requisites and at a low marginal cost. All that is required is to point the

satellite dish in the right direction and the VSAT modem is connected.

Capacity upgrade bandwidth can be added to the network for higher speed connectivity

without the need to replace any of the installed equipment.

The high reliability of satellite networks allows provides private network

customers with always-on, high speed access to all of the bandwidth-intensive

content of their intranet.

This also saves expenses related to outages and poor network performance

that are occasionally encountered on DSL or leased-line networks. Satellite

networks have the inherent capability to perform multicast transmissions to all

of the locations simultaneously.

This can be used for distributing content such as software update files, virus

updates or large files quickly and reliably. One of the main reasons that

organizations choose to implement private network s is because their business

relies on confidential information that needs to be protected from

unauthorized access.

A Satellite Based Private Network is an inherently secure end-to-end network

solution. Gilat’s VSAT modems implement the latest technologies for delivering high

performance IP connectivity with guaranteed bandwidth and advanced QoS capabilities.

Fiber optics network

In recent years it has become apparent that fiber-optics are steadily replacing

copper wire as an appropriate means of communication signal transmission.

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They span the long distances between local phone systems as well as

providing the backbone for many network systems. Other system users

include cable television services, university campuses, office buildings,

industrial plants, and electric utility companies.

Optical Fiber

A technology that uses glass (or plastic) threads (fibers) to transmit data. A

fiber optic cable consists of a bundle of glass threads, each of which is capable

of transmitting messages modulated onto light waves.

An optical fiber (or fibre) is a glass or plastic fiber designed to guide light

along its length. Fiber optics is the overlap of applied science and engineering

concerned with such optical fibers.

Optical fibers are widely used in fiber-optic communication, which permits

transmission over longer distances and at higher data rates than other forms of

wired and wireless communications.

Fibers are used instead of metal wires because signals propagate along them

with less loss, and they are immune to electromagnetic interference. Optical

fibers are also used to form sensors, and in a variety of other applications.

In fibers with large core diameter, the confinement is based on total internal

reflection. In smaller core diameter fibers, (widely used for most

communication links longer than 200 meters) the fiber acts as a waveguide.

There are many different designs of optical fibers, including graded-index

optical fibers, step-index optical fibers which are characteristics of an optical

fiber and different types of optical fiber as singlemode fibers (SMF) in which

there are three kinds of fibers, non-dispersion shifted fibres (NDSF), nonzero

dispersion-shifted fibers (NZDSF) and dispersion-shifted fibers (DSF),

multimode fibers (MMF), birefringent polarization-maintaining fibers (PMF)

and more recently photonic crystal fibers (PCF), with the design and the

wavelength of the light propagating in the fiber dictating whether or not it will

be multi-mode optical fiber or single-mode optical fiber.

Because of the mechanical properties of the more common glass optical

fibers, special methods of splicing fibers and of connecting them to other

equipment are needed. Manufacture of optical fibers is based on partially

melting a chemically doped preform and pulling the flowing material on a

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draw tower. Fibers are built into different kinds of cables depending on how

they will be used.

Working

Fiber optic technology is based on the use of light energy to transmit data.

Basically, the encoded data is converted from electrical signals to optical light

pulses and then transmitted through the medium to its destination, where it is

then converted back.

From this, we can see that there are basically three main elements in any fiber

optic data link: a transmitter, an optical cable (the transmission medium), and

a receiver. The transmitter handles the conversion from electrical to light

energy, the optical cable carries the light waves, and the receiver handles the

conversion from light pulses back to the original electrical format.

After translating the electrical signals, the transmitter uses either a light

emitting diode (LED) or an injection laser diode (ILD) to generate the light

pulses. Using a lens, this light energy is then sent down the fiber optic cable.

The principle that makes this possible is referred to as total internal reflection.

According to John Huber in an article in R&D Magazine, this principle of

total internal reflection states that when the angle of incidence exceeds a

critical value, light cannot get out of the glass; instead, the light bounces back

in (Huber 115).

This happens when two materials with different refractive indices cause the

angle of incidence to be too large for refraction (bending) of light to take

place. Since the light cannot be bent and exit the material, this means that 100

percent is reflected back.

Thus, when a fiber optic cable, which consists of a glass or plastic core

surrounded by a cladding with a lower refractive index, receives a light ray,

the light ray is confined and travels down the core to the receiving end.

Simply put, the difference in the materials used for the core and the cladding

make an extremely reflective surface at the point where they interface, which

makes the principle of total internal reflection possible. This is the

fundamental concept behind all fiber optic transmissions.

In addition to the core and the cladding, a fiber optic cable also has an outer

jacket that protects it from abrasion and other forces. Most high end cabling

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will also have a protective buffer and strength material between the cladding

and the outer jacket.

These outer layers are added to help protect the fragile core and cladding from

damage. There are two common types of cabling used for most fiber optic

applications: single-mode and multi-mode. Single-mode fiber is generally

used for long distance communications. It has a narrower core diameter,

generally 8-10 microns, with a 125-micron cladding.

Single-mode optical fiber only allows one mode of light to travel down its

core. On the other hand, multi-mode fiber generally has a 62.5-micron core

diameter, with a 125-micron cladding.

In order to receive the signal and then convert it back to its original format, a

fiber optic receiver uses a phototransistor to convert the light energy into an

electrical current.

This current is then sent into an amplifier in order to boost the electrical signal

back to its original level, and then a digitizer circuit is used to convert the

signal into the appropriate digital voltage levels to be used by the external

logic. At this point, the electronic signal is ready to be received by the

communications device, whether it is a switch, router, computer, etc.

Advantages

Because of the Low loss, high bandwidth properties of fiber cable they can be

used over greater distances than copper cables, in data networks this can be as

much as 2km without the use of repeaters.

Their light weight and small size also make them ideal for applications where

running copper cables would be impractical, and by using multiplexors one

fibre could replace hundreds of copper cables.

fiber optic systems have many attractive features that are superior to electrical

systems. These include improved system performance, immunity to electrical

noise, signal security, and improved safety and electrical isolation.

Fiber optic transmission systems - a fiber optic transmitter and receiver,

connected by fiber optic cable - offer a wide range of benefits not offered by

traditional copper wire or coaxial cable. These include: The ability to carry

much more information and deliver it with greater fidelity than either copper

wire or coaxial cable.

Disadvantages

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Because of the relative newness of the technology, fiber optic components are

expensive. Fiber optic transmitters and receivers are still relatively expensive

compared to electrical interfaces.

The lack of standardization in the industry has also limited the acceptance of

fiber optics. Many industries are more comfortable with the use of electrical

systems and are reluctant to switch to fiber optics. However, industry

researchers are eliminating these disadvantages.

Application

Optical fiber communication

Optical fiber can be used as a medium for telecommunication and networking

because it is flexible and can be bundled as cables.

It is especially advantageous for long-distance communications, because light

propagates through the fiber with little attenuation compared to electrical

cables.

This allows long distances to be spanned with few repeaters. Additionally, the

light signals propagating in the fiber can be modulated at rates as high as 40

Gb/s and each fiber can carry many independent channels, each by a different

wavelength of light (wavelength-division-multiplex WDM). In total, a single

fiber-optic cable can carry data at rates as high as 14.4 Pb/s (circa 14 million

Gb/s

A fiber-optic Christmas Tree

Optical fibers doped with a wavelength shifter are used to collect scintillation

light in physics experiments.

Optical fiber can be used to supply a low level of power (around one watt) to

electronics situated in a difficult electrical environment. Examples of this are

electronics in high-powered antenna elements and measurement devices used

in high voltage transmission equipment.

Protocol stacks

SS7 is structured in a multi-layered stack which corresponds closely to the layers of

the standard OSI model, although some SS7 components span a number of layers, as

illustrated in here.

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Fig. SS7 protocol stack

The SS7 component parts are:

Layer 1 (Physical): MTP-1 (Message Transfer Part-1)

MTP-1 defines the physical means by which SS7 messages are transferred

from one node to another. For E1 ot T1 networks, the physical layer is usually

a timeslot of an E1 or T1 frame respectively.

The physical layer specifies only how a sequence of bits is conveyed from one

SS7 node to another. It says nothing about the actual meaning of the bits or

how they are grouped together to form a message.

Layer 2 (Data Link): MTP-2

MTP–2 defines how an MTP-1 bit transfer mechanism is used to reliably pass

variable length messages from one SS7 node to another. MTP-2 uses a variant

of the High level Data Link Control (HDLC) used in most modern data

transfer protocols. This uses a delimiter to define the start and end of a data

frame, prevents flags occurring in a frame (bit-stuffing) and protection for the

entire frame (CRC at the end). It also defines how CRC errors are handled (by

error response and retransmission).

MTP-2 says nothing about the actual content of a message. It simply defines a

mechanism by which a message of any length can be sent 100% reliably

between SS7 nodes and can be used by higher layers of the SS7 protocol.

MTP-2 knows nothing beyond the single point-to-point link it operates on.

Layer 3 (Network Layer): MTP–3

MTP-3 builds on top of the lower-level MTP layers to allow the creation of a

network of telephony network nodes interconnected by SS7 links. Each node

is assigned a unique address in the network (known as a Signaling Point Code

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or SPC). Messages can be sent at the MTP-3 level in one node to a

topologically distant node (that is with one or more intermediate SS7 nodes)

simply by specifying the Destination Point Code (DPC). MTP-3 entities on

the SPC node, the DPC node, and all intermediate nodes coordinate the

transfer of a higher-layer message through the network.

MTP-3 can use multiple parallel routes from SPC to DPC through the network

to take account of link loading and availability (there should always be more

than one way to get from any SPC to any DPC).

Upper Layers: TUP (Telephone User Part)

The Telephone User Part (TUP) is used to set up a telephone call between two

SS7 nodes. It defines a set of messages and a protocol using these messages

that allows a telephone call to be set up and torn down.

TUP messages flow only immediately before a call is established and then

immediately before it is terminated. No TUP messages are exchanged when a

call is in progress.

TUP was one of the first SS7 protocols and designed to support simple analog

phones (with little function over and above call setup and tear-down).

Upper Layers: ISUP (Integrated Services User Part)

The ISUP performs the same function as the TUP (that is, it handles the setup

and tear-down of telephone calls) but it is much more sophisticated providing

function available with primary rate ISDN.

This includes calling and called number notification (or suppression), the

ability to control billing (charging) rates, advanced telephony functions such

as transfer, and control over whether the voice channel is used for voice, fax,

or data.

As with TUP, ISUP messages flow only during the setup and tear-down phases of a call.

Upper layers: SCCP (Signalling Connection Control Part)

The SCCP runs above the MTP layers and provides a set of facilities similar

to those provided by the UDP and TCP layers of TCP/IP. Specifically, SCCP

provides five classes of service such as connectionless (like UDP) and

connection-oriented (like TCP) with options of error recovery and flow

control. It also provides what is known in SS7 as Global Title Translation.

Upper layers: TCAP (Transaction Capabilities Application Part)

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The TCAP is designed to implement functions in the SS7 network which are

unrelated to the origination and termination of actual telephone calls. TCAP

provides a means by which information can be transferred from an application

at a switch location to another application in another network entity.

One example of TCAP usage is number translation and database transactions

and lookup. Another example of the use of TCAP is the Message Waiting

Indicator (MWI) on some telephones which indicates that a voice message is

waiting for the subscriber. An SS7-connected voice mail system sends a

TCAP message to the network to set the MWI flag in a subscriber's database.

Note that TCAP can be used by itself (on top of SCCP and the MTP layers),

or it can be used as a transport layer for higher-level layers such as MAP and

INAP (described in following sections).

Upper layers: MAP (Mobile Application Part)

Mobile Application Part (MAP) is the most complex SS7 component and is used in

GSM mobile telephone systems to pass information between the components of the network.

Upper layers: INAP (Intelligent Network Application Part)

The Intelligent Network Application Part (INAP) is used to implement services

within a network, which involve accesses to an SCP and might also involve the use of an

Intelligent Peripheral (IP). INAP messages are sent between network entities using TCAP

transactions.

Upper layers: OMAP (Operations and Administration Application Part)

The OMAP is typically used by a network administration facility to control an

entire network from a central point. Facilities provided in OMAP include

administration of system databases, maintenance access and performance

monitoring.

SS7 Support for Web Sphere Voice Response, which is discussed in the next

section, supports the MTP layers and the ISUP. If you need other layers to be

supported contact your IBM representative.

Internetworking

Internetwork

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An internetwork is a collection of individual networks, connected by

intermediate networking devices, that functions as a single large network.

Internetworking refers to the industry, products, and procedures that meet the

challenge of creating and administering internetworks.

Figure illustrates some different kinds of network technologies that can be

interconnected by routers and other networking devices to create an

internetwork.

Working

The first networks were time-sharing networks that used mainframes and

attached terminals. Such environments were implemented by both IBM’s

Systems Network Architecture (SNA) and Digital’s network architecture.

Local-area networks (LANs) evolved around the PC revolution. LANs

enabled multiple users in a relatively small geographical area to exchange

files and messages, as well as access shared resources such as file servers and

printers.

Wide-area networks (WANs) interconnect LANs with geographically

dispersed users to create connectivity. Some of the technologies used for

connecting LANs include T1, T3, ATM, ISDN, ADSL, Frame Relay, radio

links, and others. New methods of connecting dispersed LANs are appearing

every day.

Today, high-speed LANs and switched internetworks are becoming widely

used, largely because they operate at very high speeds and support such high-

bandwidth applications as multimedia and videoconferencing.

Internetworking evolved as a solution to three key problems: isolated LANs,

duplication of resources, and a lack of network management. Isolated LANs

made electronic communication between different offices or departments

impossible.

Internetworking Challenges

Implementing a functional internetwork is no simple task. Many challenges

must be faced, especially in the areas of connectivity, reliability, network

management, and flexibility. Each area is key in establishing an efficient and

effective internetwork.

Because companies rely heavily on data communication, internetworks must

provide a certain level of reliability. This is an unpredictable world, so many

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large internetworks include redundancy to allow for communication even

when problems occur.

Furthermore, network management must provide centralized support and

troubleshooting capabilities in an internetwork. Configuration, security,

performance, and other issues must be adequately addressed for the

internetwork to function smoothly. Security within an internetwork is

essential. Many people think of network security from the perspective of

protecting the private network from outside attacks.

However, it is just as important to protect the network from internal attacks,

especially because most security breaches come from inside. Networks must

also be secured so that the internal network cannot be used as a tool to attack

other external sites.

Because nothing in this world is stagnant, internetworks must be flexible enough to

change with new demands.

Integrated Service Digital Network (ISDN):

ISDN is a set of digital transmission protocols defined by ITU. ISDN is a

modified version of Integrated Digital Network (IDN) that provides an end-to-

end digital connectivity to support a wide range of services, including voice

and non-voice services, to which users have access by a limited set of standard

multi-purpose user network interfaces.

Digital networks are important for two basic reasons, high quality service and

speed.

One of the advantages of ISDN is its compatibility with much of the existing international

telecommunication infrastructure.

ISDN Services :

ISDN must have the ability to

Handle voice, audio, interactive data, fax, video

Capability to transport continuous traffic and bursty traffic

Fast call establishment

Allocate bandwidth on demand

To handle wide range of transmission speeds

Low bit error rates

Communication security

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Types of ISDN channels

Bearer B, channels for user voice and data

B channels for circuit switched connection just like a regular telephone connection

B channels 64 kbps

Data D channels carry setup, signalling and tear down information

D-channel use packet-switched connections

D-channel 16 kbps / 64 kbps

Two forms of ISDN access are as follows:

Basic Rate Interface (BRI)

Two 64 kbps B channel + one 16 kbps D-channel => 2 B + D

Primary Rate Interface (PRI)

-23B + D channel for T1

-30 B + D channel for E1

BRI is used in residential services.

Data Rate for BRI in 2 x 64 + 16 + (overhead for synchronization and framing)

=144 + (overhead for synchronization and framing)

=192 kbps

PRI is used for connections between a PBX and a

CO. Data rate for PRI = 1.544 Mbps for T1

= 2.048 Mbps for E1

ISDN Architecture:

Reference points: Conceptual points used to separate groups of functions.

Functional group: A group of functions that might be found in a typical device.

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Local exchange (LE): The ISDN central office.

Terminal equipment functional group, TE1: Includes telephones, fax machines,

computers.

Terminal adapter functional group, TA: Provides the functions needed to attach a non-

ISDN device to ISDN.

Network termination functional group, NT1: Connects between the signals/ wiring from

ISDN device and the signalling/ electrical standards adhered to by the local office.

NT2: A digital PBX or a LAN responsible for switching and multiplexing ISDN services.

When NT1 and NT2 are used. The S and T interfaces are separated by the NT2 which is able

to reallocate channels in the form of BRI or PRI connections.

ISDN Call Control Signalling

ACM: Address Complete Message

ANM: Answer Message

IAM: Initial Address Message

REL: Release

RLC: Release Complete

The calling party starts the sequence by sending a SETUP message to the network. In

the SETUP message, the user’s terminal includes the information that the network needs in

order to establish a call. Examples of the information are the desired bearer service

capability, the identity of the called party, the B-channel used for call etc. Bearer service

capability is the essence of ISDN. Bearer capability can be offered in circuit, frame or packet

mode.

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Frame relay

Frame relay is a form of packet switching that provides a streamlined

interface compared to X.25 with improved performance. Accordingly, there

is a large installed base of frame relay products. Interest has since shifted to

ATM for high-speed data networking, but because of the remaining

popularity of frame relay, we provide a survey in this section.

Background

              At each hop through the network, the data link control protocol

involves the exchange of a data frame and an acknowledgment frame.

Furthermore, at each intermediate node, state tables must be maintained for

each virtual circuit to deal with the call management and flow control/error

control aspects of the X.25 protocol.

All of this overhead may be justified when there is a significant probability

of error on any of the links in the network. This approach may not be the

most appropriate for modern digital communication.

             Frame relaying is designed to eliminate much of the overhead

involved in X.25. The key differences between frame relaying and a

conventional X.25 packet-switching service are as follows;

Call control signalling is carried on a separate logical connection from user data.

Thus, intermediate nodes need not maintain state tables or process messages relating

to call control on an individual per-connection basis.

Multiplexing and switching of logical connections takes place at layer 2 instead of

layer 3, eliminating one entire layer of processing.

There is no hop-by-hop flow control and error control. End-to-end flow control and

error control are the responsibility of a higher layer, if they are employed at all.

Thus , with frame relay, a single user data frame is sent from source to destination, and an

acknowledgment, generated at a higher layer, is carried back in a frame. There are no hop-

by-hop exchanges of data frames and acknowledgments.

Disadvantages

The ability to do link-by-link flow and error control.

Hop-by-hop link control is lost.

Advantage

Streamlined communications process

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Frame relay protocol architecture

The fig. depicts the protocol architecture to support the frame mode bearer service.

There are two separate planes of operation: a control(C) plane, which is involved in the

establishment and termination of logical connections, and a user (U) plane, which is

responsible for the transfer of user data between subscribers.

Fig. Frame relay protocol architecture

Control plane

The control plane for frame mode bearer services is similar to that for common

channel signalling for circuit-switching services, in that a separate logical channel is used for

control information. At the data link layer, LAPD(Q.921) is used to provide a reliable data

link control service, with error control and flow control, between user(TE) and network(NT)

over the D channel.

User plane:       

For the actual transfer of information between end users, the user-plane protocol is LAPF

(Link Access Procedure for Frame Mode Bearer Services), which is defined in Q.922. Only

the core functions of LAPF are used for frame relay:

Frame delimiting, alignment, and transparency

Frame multiplexing/de-multiplexing using the address field

Inspection of the frame to ensure that it consists of an integral number of  octets prior

to zero bit insertion or following zero bit extraction

Inspection of the frame to ensure that it is neither too long nor too short

Detection of transmission errors

Congestion control functions

Based on the core functions, a network offers frame relaying as a connection oriented

link layer service with the following properties:

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Preservation of the order of frame transfer from one edge of the network to the other

A small probability of frame loss

User data transfer

Consider the frame format, this is the format defined for the minimum-function LAPF

protocol (known as LAPF core protocol). The format is similar to that a LAPD and LAPB

with one obvious omission: there is no control field. This has the following implications:

There is only one frame type, used for carrying user data. There are no control

frames.

It is not possible to use in band signalling; a logical connection can only carry user

data.

It is not possible to perform flow control and error control, because there are no

sequence numbers.

The Flag and Frame Check Sequence (FCS) fields function as in HDLC. The information

field carries higher-layer data.

TCP/IP Reference Model

Let us now turn from the OSI reference model to the reference model used in

the grandparent of all wide area computer networks, the ARPANET, and its

successor, the worldwide Internet. Although we will give a brief history of the

ARPANET later, it is useful to mention a few key aspects of it now.

Given the DoD's worry that some of its precious hosts, routers, and

internetwork gateways might get blown to pieces at a moment's notice,

another major goal was that the network be able to survive loss of subnet

hardware, with existing conversations not being broken off.

In other words, DoD wanted connections to remain intact as long as the source

and destination machines were functioning, even if some of the machines or

transmission lines in between were suddenly put out of operation.

Furthermore, a flexible architecture was needed since applications with

divergent requirements were envisioned, ranging from transferring files to

real-time speech transmission.

The Internet Layer

All these requirements led to the choice of a packet-switching network based on a

connectionless internetwork layer. This layer, called the internet layer, is the linchpin

that holds the whole architecture together.

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Its job is to permit hosts to inject packets into any network and have them travel

independently to the destination (potentially on a different network). They may even

arrive in a different order than they were sent, in which case it is the job of higher

layers to rearrange them, if in-order delivery is desired. Note that ''internet'' is used

here in a generic sense, even though this layer is present in the Internet.

The analogy here is with the (snail) mail system. A person can drop a sequence of

international letters into a mail box in one country, and with a little luck, most of

them will be delivered to the correct address in the destination country.

Probably the letters will travel through one or more international mail gateways along

the way, but this is transparent to the users. Furthermore, that each country (i.e., each

network) has its own stamps, preferred envelope sizes, and delivery rules is hidden

from the users.

The internet layer defines an official packet format and protocol called IP (Internet Protocol).

The job of the internet layer is to deliver IP packets where they are supposed to go.

Packet routing is clearly the major issue here, as is avoiding congestion. For these

reasons, it is reasonable to say that the TCP/IP internet layer is similar in functionality

to the OSI network layer. Figure shows this correspondence.

Fig. The TCP/IP reference model

The Transport Layer

The layer above the internet layer in the TCP/IP model is now usually called the

transport layer. It is designed to allow peer entities on the source and destination hosts

to carry on a conversation, just as in the OSI transport layer.

Two end-to-end transport protocols have been defined here. The first one, TCP

(Transmission Control Protocol), is a reliable connection-oriented protocol that

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allows a byte stream originating on one machine to be delivered without error on any

other machine in the internet.

It fragments the incoming byte stream into discrete messages and passes each one on

to the internet layer. At the destination, the receiving TCP process reassembles the

received messages into the output stream. TCP also handles flow control to make sure

a fast sender cannot swamp a slow receiver with more messages than it can handle.

The second protocol in this layer, UDP (User Datagram Protocol), is an unreliable,

connectionless protocol for applications that do not want TCP's sequencing or flow

control and wish to provide their own.

The relation of IP, TCP, and UDP is shown in Fig. Since the model was developed,

IP has been implemented on many other networks.

Fig.. Protocols and networks in the TCP/IP model initially

The Application Layer

The TCP/IP model does not have session or presentation layers. No need for

them was perceived, so they were not included. Experience with the OSI

model has proven this view correct: they are of little use to most applications.

On top of the transport layer is the application layer. It contains all the higher-

level protocols. The early ones included virtual terminal (TELNET), file

transfer (FTP), and electronic mail (SMTP), as shown in Fig. 1-22.

The Host-to-Network Layer

Below the internet layer is a great void. The TCP/IP reference model does not

really say much about what happens here, except to point out that the host has

to connect to the network using some protocol so it can send IP packets to it.

This protocol is not defined and varies from host to host and network to

network. Books and papers about the TCP/IP model rarely discuss it.

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Asynchronous Transfer Mode (ATM)

ATM was going to solve all the world's networking and telecommunications

problems by merging voice, data, cable television, telex, telegraph, carrier

pigeon, tin cans connected by strings, tom-toms, smoke signals, and

everything else into a single integrated system that could do everything for

everyone. It did not happen.

In large part, the problems were similar to those we described earlier

concerning OSI, that is, bad timing, technology, implementation, and politics.

Having just beaten back the telephone companies in round 1, many in the

Internet community saw ATM as Internet versus the Telcos: the Sequel. But it

really was not, and this time around even diehard datagram fanatics were

aware that the Internet's quality of service left a lot to be desired. To make a

long story short,

ATM was much more successful than OSI, and it is now widely used deep

within the telephone system, often for moving IP packets. Because it is now

mostly used by carriers for internal transport, users are often unaware of its

existence, but it is definitely alive and well.

ATM Virtual Circuits

Since ATM networks are connection-oriented, sending data requires first

sending a packet to set up the connection. As the setup packet wends its way

through the subnet, all the routers on the path make an entry in their internal

tables noting the existence of the connection and reserving whatever resources

are needed for it.

Connections are often called virtual circuits, in analogy with the physical

circuits used within the telephone system. Most ATM networks also support

permanent virtual circuits, which are permanent connections between two

(distant) hosts.

They are similar to leased lines in the telephone world. Each connection,

temporary or permanent, has a unique connection identifier. A virtual circuit

is illustrated in Fig.

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Figure A virtual circuit.

Once a connection has been established, either side can begin transmitting

data. The basic idea behind ATM is to transmit all information in small, fixed-

size packets called cells.

The cells are 53 bytes long, of which 5 bytes are header and 48 bytes are

payload, as shown in Fig. Part of the header is the connection identifier, so the

sending and receiving hosts and all the intermediate routers can tell which

cells belong to which connections.

This information allows each router to know how to route each incoming cell.

Cell routing is done in hardware, at high speed. In fact, the main argument for

having fixed-size cells is that it is easy to build hardware routers to handle

short, fixed-length cells.

Variable-length IP packets have to be routed by software, which is a slower

process. Another plus of ATM is that the hardware can be set up to copy one

incoming cell to multiple output lines, a property that is required for handling

a television program that is being broadcast to many receivers. Finally, small

cells do not block any line for very long, which makes guaranteeing quality of

service easier.

Fig. An ATM cell.

All cells follow the same route to the destination. Cell delivery is not

guaranteed, but their order is. If cells 1 and 2 are sent in that order, then if

both arrive, they will arrive in that order, never first 2 then 1. But either or

both of them can be lost along the way. It is up to higher protocol levels to

recover from lost cells.

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Note that although this guarantee is not perfect, it is better than what the

Internet provides. There packets can not only be lost, but delivered out of

order as well. ATM, in contrast, guarantees never to deliver cells out of order.

ATM networks are organized like traditional WANs, with lines and switches

(routers). The most common speeds for ATM networks are 155 Mbps and 622

Mbps, although higher speeds are also supported.

The 155-Mbps speed was chosen because this is about what is needed to

transmit high definition television. The exact choice of 155.52 Mbps was

made for compatibility with AT&T's SONET transmission system.The 622

Mbps speed was chosen so that four 155-Mbps channels could be sent over it.

The ATM Reference Model

ATM has its own reference model, different from the OSI model and also

different from the TCP/IP model. This model is shown in Fig. It consists of

three layers, the physical, ATM, and ATM adaptation layers, plus whatever

users want to put on top of that.

Fig. The ATM reference model

The physical layer deals with the physical medium: voltages, bit timing, and

various other issues. ATM does not prescribe a particular set of rules but

instead says that ATM cells can be sent on a wire or fiber by themselves, but

they can also be packaged inside the payload of other carrier systems. In other

words, ATM has been designed to be independent of the transmission

medium.

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The ATM layer deals with cells and cell transport. It defines the layout of a

cell and tells what the header fields mean. It also deals with establishment and

release of virtual circuits. Congestion control is also located here.

Because most applications do not want to work directly with cells (although

some may), a layer above the ATM layer has been defined to allow users to

send packets larger than a cell. The ATM interface segments these packets,

transmits the cells individually, and reassembles them at the other end. This

layer is the AAL (ATM Adaptation Layer).

Unlike the earlier two-dimensional reference models, the ATM model is

defined as being three-dimensional, as shown in Fig. The user plane deals

with data transport, flow control, error correction, and other user functions. In

contrast, the control plane is concerned with connection management. The

layer and plane management functions relate to resource management and

interlayer coordination.

The physical and AAL layers are each divided into two sublayers, one at the

bottom that does the work and a convergence sublayer on top that provides the

proper interface to the layer above it. The functions of the layers and sublayers

are given in Fig.

Fig. The ATM layers and sublayers, and their functions

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The PMD (Physical Medium Dependent) sublayer interfaces to the actual

cable. It moves the bits on and off and handles the bit timing. For different

carriers and cables, this layer will be different.

The other sublayer of the physical layer is the TC (Transmission

Convergence) sublayer. When cells are transmitted, the TC layer sends them

as a string of bits to the PMD layer. Doing this is easy. At the other end, the

TC sublayer gets a pure incoming bit stream from the PMD sublayer.

Its job is to convert this bit stream into a cell stream for the ATM layer. It

handles all the issues related to telling where cells begin and end in the bit

stream. In the ATM model, this functionality is in the physical layer. In the

OSI model and in pretty much all other networks, the job of framing, that is,

turning a raw bit stream into a sequence of frames or cells, is the data link

layer's task.

As we mentioned earlier, the ATM layer manages cells, including their

generation and transport. Most of the interesting aspects of ATM are located

here. It is a mixture of the OSI data link and network layers; it is not split into

sublayers.

The AAL layer is split into a SAR (Segmentation And Reassembly) sublayer

and a CS (Convergence Sublayer).

The lower sublayer breaks up packets into cells on the transmission side and

puts them back together again at the destination. The upper sublayer makes it

possible to have ATM systems offer different kinds of services to different

applications (e.g., file transfer and video on demand have different

requirements concerning error handling, timing, etc.).

As it is probably mostly downhill for ATM from now on, we will not discuss

it further in this book. Nevertheless, since it has a substantial installed base, it

will probably be around for at least a few more years. For more information

about ATM, see (Dobrowski and Grise, 2001; and Gadecki and Heckart,

1997).

Packet Switching

With message switching, there is no limit at all on block size, which means

that routers (in a modern system) must have disks to buffer long blocks. It also

means that a single block can tie up a router-router line for minutes, rendering

message switching useless for interactive traffic.

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To get around these problems, packet switching was invented, as described in

Chap. 1. Packet-switching networks place a tight upper limit on block size,

allowing packets to be buffered in router main memory instead of on disk. By

making sure that no user can monopolize any transmission line very long

(milliseconds), packet-switching networks are well suited for handling

interactive traffic.

A further advantage of packet switching over message switching is shown in

Fig. 2-39(b) and (c): the first packet of a multipacket message can be

forwarded before the second one has fully arrived, reducing delay and

improving throughput. For these reasons, computer networks are usually

packet switched, occasionally circuit switched, but never message switched.

Circuit switching and packet switching differ in many respects. To start with,

circuit switching requires that a circuit be set up end to end before

communication begins.

Packet switching does not require any advance setup. The first packet can just

be sent as soon as it is available. The result of the connection setup with

circuit switching is the reservation of bandwidth all the way from the sender

to the receiver. All packets follow this path.

Packet switching is more fault tolerant than circuit switching. In fact, that is

why it was invented. If a switch goes down, all of the circuits using it are

terminated and no more traffic can be sent on any of them. With packet

switching, packets can be routed around dead switches.

Setting up a path in advance also opens up the possibility of reserving

bandwidth in advance. If bandwidth is reserved, then when a packet arrives, it

can be sent out immediately over the reserved bandwidth.

With packet switching, no bandwidth is reserved, so packets may have to wait

their turn to be forwarded. Having bandwidth reserved in advance means that

no congestion can occur when a packet shows up (unless more packets show

up than expected). On the other hand, when an attempt is made to establish a

circuit, the attempt can fail due to congestion. Thus, congestion can occur at

different times with circuit switching (at setup time) and packet switching

(when packets are sent).

If a circuit has been reserved for a particular user and there is no traffic to

send, the bandwidth of that circuit is wasted. It cannot be used for other

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traffic. Packet switching does not waste bandwidth and thus is more efficient

from a system-wide perspective. Understanding this trade-off is crucial for

comprehending the difference between circuit switching and packet switching.

The trade-off is between guaranteed service and wasting resources versus not

guaranteeing service and not wasting resources.

Packet switching uses store-and-forward transmission. A packet is

accumulated in a router's memory, then sent on to the next router

. With circuit switching, the bits just flow through the wire continuously. The

store-and-forward technique adds delay.

Another difference is that circuit switching is completely transparent. The

sender and receiver can use any bit rate, format, or framing method they want

to. The carrier does not know or care.

With packet switching, the carrier determines the basic parameters. A rough

analogy is a road versus a railroad. In the former, the user determines the size,

speed, and nature of the vehicle; in the latter, the carrier does. It is this

transparency that allows voice, data, and fax to coexist within the phone

system.

A final difference between circuit and packet switching is the charging

algorithm. With circuit switching, charging has historically been based on

distance and time.

For mobile phones, distance usually does not play a role, except for

international calls, and time plays only a minor role (e.g., a calling plan with

2000 free minutes costs more than one with 1000 free minutes and sometimes

night or weekend calls are cheaper than normal).

With packet switching, connect time is not an issue, but the volume of traffic

sometimes is. For home users, ISPs usually charge a flat monthly rate because

it is less work for them and their customers can understand this model easily,

but backbone carriers charge regional networks based on the volume of their

traffic.

Both circuit switching and packet switching are important enough that we will

come back to them shortly and describe the various technologies used in

detail.

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