bsnl summer

8/6/2019 Bsnl Summer

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COMPANY PROFILE

BHARAT SANCHAR NIGAM LIMITED

Bharat Sanchar Nigam Limited (BSNL) is Indias leading telecommunication provider

and the countrys largest public sector firm. It is fourth largest department of

telecommunication company in Asia and seventh in world today. BSNL provides local -

exchange access and domestic long-distance services through a network of more than 45

million access lines covering most of India. It also offers wireless communication, data

and Internet services,as well as business voice and data services.

HIGHLIGHTS

Bharat Sanchar Nigam Limited has a vast reservoir of highly skilled and

experienced work force of about 3,57,000 personnel.

To meet the technological challenges, employee are trained for technology up-

gradation, computerization etc in BSNLs training centers spread across Country.

To apex training centers of BSNL i.e. Advance level Telecom Training

Center(ALTTC) at Ghaziabad & Bharat Ratna Bhimrao Telecom Training Center

at Jabalpur are comparable to any world class Telecom Training Center.

Moreover, 43 zonal training centers and a national Academy Of Telecom Finance

andManagement have been running for several years now.


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VISION

To become the largest telecom Service Provide in South East Asia.

MISSION

y To provide world class State of Art technology telecom services on demand at

affordable price.

y To provide world class telecom infrastructure to develop countrys economy.

PARTICULARS OF THE ORGANISATION

Date of Incorporation Incorporated on 15.9.2000, vide Registration No. 55-107739,dated the 15th sep.2000 and became entitled to commence

business with effect from 19th

sep.2000.

The company(BSNL) took over the business of providingtelecom services and network management throughout the

country except the metro cities of Delhi andMumbai of theerstwhile services providing dept.of the Govt.of India.

Type Of Company Govt. Company under Section 617 of the Companies Act,1956

AdministrativeMinistry Govt. of India,Ministry of Communication and Information

Technology, Department of Telecommunications.

Details Of Investments The entire share capital of the company is held by the Govt. ofIndia.

Shareholding Pattern Govt. of India is holding 100% of the share capital of thecompany.

Listing with StockExchanges

Not applicable, as the BSNL is an unlisted company.


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GLIMPSES OF MAIN SERVICES OFFERED

- Basic and limited mobile telephone services

- Cellular mobile telephone services- Internet services

- Intelligent network- Broadband services

OBLIGATIONS

Towards customers and dealers

To provide prompt, courteous and efficient service and quality of products/services at fair

and reasonable services.

Towards employees

Develop their capability and advancement through appropriate training and careerplanning.

Expeditious redressed of grievances.Fair dealings with recognized representatives of employees in pursuance of healthy trade

union practices and sound personnel policies.

Towards the Society

BSNL is committed to provide quality Telecom Services at affordable price to the

citizens of the remotest part of the Country. BSNL is making all effort to ensure that themain objectives of the new Telecom Policy 1999 achieved.

Manufacturing Units

Telecom factories to manufacture telephone switching boards and accessories at Bhilai,

Mumbai, Kolkata and Jabalpur.

y I.T.I. Bangalore for the manufacturing for the carriers, VFT, Coaxial andmicrowave equipment.

y I.T.I. Gonda for the manufacturing of E-10B electronic exchange equipment.

y Hindustan Cables LTD. Hyderabad & Rupnarainpur for manufacturingunderground cables.

y Hindustan teleprinters LTD. Chennai for manufacturing teleprinters.


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HISTORY OF TELEPHONE EXCHANGE

The Telephone was invented by Mr. Graham Bell. During early stage ofdevelopment of telephone exchange, the connections are established with help of human

operator.Those type of exchange were called manual Telephone Exchange. Thetechnology of telephony was going on progress with the introduction of automatic

exchange. Manual telephone is replaced & automatic exchange became in use there werelot of advantage of automatic exchange over manual.

In manual telephony, the type of exchange used is Central Battery (C.B.).In certain case local battery exchange (L.B.) is also used. The local battery exchange is

also called magnet exchange because the set has a magneto generator which thesubscriber is required to rotate, to generate the A.C. necessary to operate the indicator at

the exchange.In the central battery exchange, the battery is located at the central place

which is the exchange. This arrangement has many advantages over L.B. exchange & can

be used for even for large & medium capacity exchange.In automatic telephony connections between two subscriber are

established with the help of human operator. Obviously the junction of human operator is

carried out by the machine known as switching or selector stages. After the developmentof automatic telephone exchange technology as a subscriber directly & it has many

advantages over manual telephone exchange. Now a day electronic Automatic exchangeis widely used due to their advantages.

YEAR INCIDENTS

1876 Invention of Telephone1915 first transcontinental telephone(NY-SF)

1920 first automatic switches1956 TAT-1 transatlantic cables(35 lines)

1962 Digital transmission(T1)1965 1ESS analog switches

1974 Internet packet voice1977 4ESS digital switches

1980s Signaling System (out-of-band)1990s Advanced Intelligent Network (AIN)


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ADVANTAGES OF ELECTRONIC EXCHANGES OVER

ELECTROMECHANICAL EXCHANGES

y In electromechanical exchanges category analysis, routing, translation, etc is done

by relays while in electronic exchanges translation, speech path, subscriber

facilities, etc are managed by map and other data.

y Electromechanical exchanges have limited flexibility while electronic exchanges

are highly flexible.

y In electromechanical exchanges any change in facilities require addition of

hardware changes whereas in electronic exchanges can be carried out by simple

commands.

y Testing is done manually externally and time consuming process in

electromechanical exchanges whereas in electronic exchanges testing is carried

out automatically analysis is printed out.y In electromechanical exchanges, there is partially full-availability hence blocking

problem is there. Electronic exchanges are fully available hence no blocking.

y Limited facilities are available to subscriber in electromechanical exchanges than

electronic exchanges.

y Electromechanical exchanges are slow in speed as compared to electronic

exchange.

y Switch room occupies large volume in electromechanical exchanges.

y There is lot of switching noise in electromechanical exchanges as compared to

electronic exchanges.y Longer installation time is required in electromechanical exchanges.


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TELEPHONE LINES

Telephone lines are very important in telephone exchange. By the cable the

subscriber is connected with each other. In the whole process, we use two types ofcable:

1. Switching Board Cable(S.B. cable)2. Under Ground cable(U.G. cable)

SWITCH BOARD CABLE: S.B. cable is used for the indoor process. By this cable

we connect exchange to the M.D.F. room. This cable has pairs. One S.B. cable haseight gaps & each gap has eight pair. One gap is connected with a L.C.C.(Line Circuit

Card) & One module. The Cable is Connected to left switch of module.

COLOUR CODING OF S.B. CABLE: There are two colours wire in one pair. One

colour is called prime colour & other one is called made colour.

PRIME COLOUR:

1. Blue 2. Orange

3. Green 4. Brown 5. Slate

MADE COLOUR:

1. White 2. Yellow

3. Black 4. Red

Colour Combination of S.B. cable:

1. Blue-White 2.Brown-White 3. Orange-White

4. Green-White 5. Slate-White 6. Blue-Red

7. Orange-Red 8. Green-Red

There are Eight different colors of wraps in S.B. cable which are given below:

1. White 2. Yellow 3. Brown 4. Blue

5. Grey 6. Black 7. Pink 8. Green


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UNDER GROUND CABLE: U.G. cables are used for outdoor process. By the cablewe connect M.D.F. to subscribers from there left-right to a module. This cable has

hundred pairs to thousands of pairs. There is twenty pair in one band.

COLOUR CODING IN U.G. CABLE: In one pair there are two colour wires. In

this first is called Primary Color & other isMade Color like S.B. cable. Chart of colorcoding is given below:

The pair of colour is made according to this chart respectively

Made

colour

WHITE RED YELLOW BLACK

Primecolour

Blue 1 6 11 16

Orange 2 7 12 17

Green 3 8 13 18

Brown 4 9 14 19

slate 5 10 15 20


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INTRODUCTION OF OCB 283

OCB-283 is digital switching system which supports a variety of communication needs

like basic telephony, ISDN, interface to mobile communication ,data communication etc.This system has been developed by CIT ALCATEL of France and therefore has manysimilarities to its predecessors E-10B also known as OCB-181 in France.

The first OCB-283 exchange Of R-11 version was commissioned in Brest,France and Beijing, china in 1911, the first OCB_283 exchange came to India in 1993

subsequently the system has been upgraded and current version R-20 was fully validatedin January 1994. The exchanges, which are being supplied to India, belong to R-20

version. Thereafter time up gradation to this OCB-283 system was upgraded toR-25version.The basic architecture remaining same, more facilities to subscriber and

administration are supported by later versions.

O-----------------ORGAN

C-----------------COMMAND OR CONTROLB-----------------BOARD

2-----------------2ND GENERATION

83----------------MICRO PROCESSOR UNIT

SALIENT FEATURE OF OCB-283

1. It is a digital switching system with an angle T stage switch a maximum of 2048PCM can be connected.

2. It supports both analog & digital subscribers.

3.The system supports all the existing signaling system like decadic,M1(R2),CAS & also

CCITT#7 signaling system.

4. It provides telephony,ISDN data communication,cellular radio & other value addedservices.

5. The system has automatic recovery feature. When a serious fault occurs in control

unit,it gives the message to SMM. The SMM pulls this unit out of service, loads softwareof this unit in a backup unit & brings it in to service diagnostic programme are runned on

the faulty unit & a diagnosis is printed on terminal.

6.OCB-283 has a double remoting facility. Subscriber access unit CSNL can be placed ata remote place & connected to main exchange through PCM links. Further line


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connectors can also be placed at remote location & connected to CSNL or CSND throughPCM this special feature can meet entire range of necessity viz urban,semi-urban & rural.

7.Various units of OCB-283 system are connected over Token ring. This enables fast

exchange of information & avoid complicated links & wiring between various units.

8. The charge account of subscriber are automatically saved in a disk once in a day. Thisavoid loss of revenue in case of total power supply failure.

9. The traffic handling capacity is 8 lacs BHCA & 25 thousands elands of traffic.

Depending upon traffic a maximum Of 2 lakhs subscriber of 60 thousands can beconnected.

10. The exchange can be managed either locally or from an NMC through 64 KBps links.

11. All the control units are implemented on same type of hardware. This is called a

station. Depending on the requirement of processing capacity, software of either one orseveral control units can be located on the same station. For all these control units only

one backup station is provided enabling Automatic Recovery in case of fault.

12. The OCB-283 system is made up of only 35 type of cards. This excludes the cardsrequired for CSN. Because of this number of space card to be kept for maintenance are

drastically reduced.

13. The system has modular structure. The expansion can be very easily carried out byadding necessary hardware or software.

14. The SMMs (O&M Units) are duplicated with one active & other hot stanby. In case

of faults , switch over takes place automatically.More over as disk are connectedSMMs,there is no necessity of changing cables from one system to another.

15. The hard disk of memory capacity 9.2 Gb is very compact & maintenance free. The

detail billing data regularly saved in the disk itself from where they can be transferred tomagnetic tapes for processing.

16. The space requirement is very small no separate room is required for OMC.

17. There is no fixed or rigid rack and suite configuration in the system , it provides great

flexibility and adjustment in the available space.

18. The environment requirement of the system is very flexible. False floor and ceilingare not essential. Air conditioning requirement are also not stringent. This system can

work at temperature 5 degree Celsius to 45 degree Celsius, though optimum temperatureis 22 degree Celsius.


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A Typical Telephone Exchange -OCB-283

FUNCTIONAL ARCHITURE

The Alcatel E10 system is located at the heart of the telecommunication networksconcerned. It is made up of three independent functional units:

- The Subscriber Access Subsystem which carries out connection of analogue and

digital subscriber lines,

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

- Operation and Maintenance which is responsible for all functions needed by thenetwork operating authority.

Each functional unit is equipped with softwares which are appropriate for handling thefunctions for which it is responsible.

HARDWARE CONFIGURATION

OCB 283 exchange comprises following hardware units.

1. Subscriber Access Units(CSNL, CSND, CSED)2. Trunk and Junction connection Units(SMT)3. Switching Matix(SMX)4. Auxiliary Equipments(SMA)5. Control Units(SMC)6. Communication Multiplex7. Time Base Generator(BTS)8. Operation and Maintenance Unit(SMM)


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Description of Hardware Units

1) Subscriber Access Units:

Subscriber connection units(CCN) are so designed that they can be equipped with either

analogue subscriber or digital subscriber or both. The cards for analogue subscriber and

digital subscriber are different, but can be equipped in any slot of the shelf.CSN can beeither placed in the exchange switch room or at a remote location. Depending upon theirlocation, CSN is known as CSNL or CSND and the subscriber shelf is known as local or

remote concentrator CNL or CNE.

CSN can have one one basic rack and up to 3 extension racks.Its architecture can

be broadly divided into two parts.

Digital Control Unit(UCN): It is the interface between concentrators and the exchange.

It is in the basic rack placed in switch room for CSNL and at remote location for CSND.It can be further broken into two parts.

Concentrators(CNL or CNE): The shelf, which accommodates subscriber line cards, is

known as Concentrators. The concentrators can either be co-located with the digitalcontrol unit in which case they are known as local concentrators CNE. The max.capacityof a concentrator is 256 subscribers.

Fig: Connection of local and remote concentrators to CSNL

2) Trunk and Junction Connection Unit(SMT):

PCM time slot by SMT. The SAB function (branch selection and amplification) is alsoknown as PCM trunk control station. This SMT provides an operational interface

between the PCMs coming from the exchange (CSND or CSNE) and the switchingcenter. The current version of SMT being supplied to our exchanges in India is SMT2G.

It is new functional variant of the SMT station.

Subs CNL

UCX SMXSubs

CNE ICNE


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3) Switching Matrix(SMX):

The SMX station is an element of the central connection matrix of the OCB283 system.

Under the control of the control stations, it performs the following functions.

y Clock reception and distribution

y Control of the station

y Interface with the connection units and the other SMX stations

y Connection of CX input lines to 256 CX output lines

y Connection security

y Help in fault location of the LOCOVAR

y Station alarm processing

The switching network in OCB283 is single T stage system. It is made up of

a) Hot SwitchingMatrixb) Branch Selection and Amplification(SAB)

4) Auxiliary Equipment Control station(SMA):

The SMA station receives the auxiliaries from the OCB283 exchange. These are

y Frequency receive/generator used for setting up calls

y Conference circuitsy Tone generators, time management operator

y CCITT number 7 signaling receiver/transmitter.

The SMA contains the following two functional units.

i. ETA

ii. PUPE

ETA contains the following subscriber components:

y Frequency receiver/generators

y Conference call circuits

y Tone generators

5) Control Units(SMC):

Since all the control units like MR, MQ, TX, TR etc. and SMA are implemented on a

common type of hardware architecture known as station. It is worthwhile to understand

the architecture and concept of station. A station is built around a multiprocessor station


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bus BSM. One or more processor and one or more intelligent couplers connected to this

bus. They interchange data through the common memory. The principal or main

processor is connected to common memory through a 32bit private bus apart from

through BSM.

A block schematic of the station is shown in the figure below.

Main

Processor

Unit

Common

Memory

Secondary

Processor

Unit1

Secondary

Processor

Unit 4(PUS)

Main Coupler

(CMP)

Secondary

Coupler

(CMS)

Secondary

Coupler 4

(CMS)

Specific

Coupler


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OCB 283

SUBCRIBER

ACCESS

SUBSYSTE

MCONNECTION

AND

CONTROL

OPERATION

AND

MAINTENANCE

DATA

NETWORK

TELEPHONE

NETWORK

VALUE ADDED

NETWORK

CCITT N07 SIGNALLING

NETWORK

NT

PABX

ALCATEL 1000 E10

OPERATION

AND

MAINTENANCE

NETWORK


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MULTIPLEXING TECHNIQUES

There are basically two types of multiplexing techniques

i. Frequency DivisionMultiplexing (FDM)

ii Time DivisionMultiplexing (TDM)

Frequency Division Multiplexing Techniques (FDM)

The FDM techniques is the process of translating individual speech circuits (300-

3400 Hz) into pre-assigned frequency slots within the bandwidth of the transmissionmedium. The frequency translation is done by amplitude modulation of the audio

frequency with an appropriate carrier frequency. At the output of the modulator a filternetwork is connected to select either a lower or an upper side band. Since the intelligence

is carried in either side band, single side band suppressed carrier mode of AM is used.This results in substantial saving of bandwidth mid also permits the use of low power

amplifiers. refer Fig. 1.

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

analog transmission system is one which is used for transmitting continuously varyingsignals.

Fig. 1 FDM Principle

Time Division Multiplexing

Basically, time division multiplexing involves nothing more than sharinga transmission medium by a number of circuits in time domain by establishing a

sequence of time slots during which individual channels (circuits) can be transmitted. Thusthe entire bandwidth is periodically available to each channel. Normally all time slots

1are

equal in length. Each channel is assigned a time slot with a specific common repeatitionperiod called a frame interval. This is illustrated in Fig. 2.


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Fig. 2 Time Division MultiplexingEach channel is sampled at a specified rate and transmitted for a fixed duration. All

channels are sampled one by, the cycle is repeated again and again. The channels are

connected to individual gates which are opened one by one in a fixed sequence. At thereceiving end also similar gates are opened in unision with the gates at the transmitting

end.

The signal received at the receiving end will be in the form of discrete

samples and these are combined to reproduce the original signal. Thus, at a giveninstant of time, only one channel is transmitted through the medium, and by sequential

sampling a number of channels can be staggered in time as opposed to transmitting all

the channel at the same time as in EDM systems. This staggering of channels in timesequence for transmission over a common medium is called Time DivisionMultiplexing (TDM).

Pulse Code Modulation

It was only in 1938, Mr. A.M. Reaves (USA) developed a Pulse Code Modulation(PCM) system to transmit the spoken word in digital form.

PCM systems use TDM technique to provide a number of circuits on the sametransmission medium viz open wire or underground cable pair or a channel provided by

carrier, coaxial, microwave or satellite system.


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Basic Requirements for PCM System

To develop a PCM signal from several analogue signals, the following

processing steps are required

Filtering

Sampling

Quantisation

Encoding

Line Coding

3.1 FILTERING

Filters are used to limit the speech signal to the frequency band 300-3400 Hz.

3.2 SAMPLING

It is the most basic requirement for TDM. Suppose we have an analogue

signal Fig. 3 (b), which is applied across a resistor R through a switch S as shown in Fig. 3(a) . Whenever switch S is closed, an output appears across R. The rate at which S is closed

is called the Sampling frequency. Fig. 3(d) is a stream of samples of the input signalwhich appear across R. The amplitude of the sample is depend upon the amplitude of the

input signal at the instant of sampling. The duration of these sampled pulses is equal to theduration for which the switch S is closed.

Fig. 3: Sampling Process


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Sampling Theorem.

A complex signal will have certain amplitudes for all frequency components of which the

signal is made. Let us say that these frequency components occupy a certain bandwidth B.If a signal does not have any value beyond this bandwidth B, then it is said to be band

limited. The extent of B is determined by the highest frequency components of the signal.

Sampling Theorem States

"If a band limited signal is sampled at regular intervals of time and at a rate equal toor more than twice the highest signal frequency in the band, then the sample contains

all the information of the original signal." Mathematically, if fH is the highest frequencyin the signal to be sampled then the sampling frequency Fs needs to be greater than 2 fH.

i.e. Fs>2fH

Let us say our voice signals are band limited to 4 KHz and let sampling frequency be 8

KHz.

Time period of sampling Ts = 1 sec

8000

or Ts = 125 micro seconds

If we have just one channel, then this can be sampled every 125 microseconds andthe resultant samples will represent the original signal. But, if we are to sample N

channels one by one at the rate specified by the sampling theorem, then the time availablefor sampling each channel would be equal to Ts/N microseconds.

FIG. 4: Sampling and combining Channels

Fig. 4 shows how a number of channels can be sampled and combined. The

channel gates (a, b ... n) correspond to the switch S in Fig. 3. These gates are opened bya series of pulses called "Clock pulses". These are called gates because, when closed these

actually connect the channels to the transmission medium during the clock period and


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isolate them during the OFF periods of the clock pulses. The clock pulses are staggered sothat only one pair of gates is open at any given instant and, therefore, only one channel

is connected to the transmission medium. The time intervals during which the commontransmission medium is allocated to a particular channel is called the Time Slot for that

channel. The width of this time slot will depend, as stated above, upon the number of

channels to be combined and the sampling frequency.In a 30 channel PCM system. TS i.e. 125 microseconds are divided into 32 parts.

That is 30 time slots are used for 30 speech signals, one time slot for signalling of all

the 30 chls, and one time slot for synchronization between Transmitter &Receiver.

The time available per channel would be Ts/N = 125/32 = 3.9 microseconds. Thusin a 30 channel PCM system, time slot is 3.9 microseconds and time period of sampling

125 microseconds. This duration i.e. 125 microseconds is called Time Frame.

The signals on the common medium (also called the common highway)

of a TDM system will consist of a series of pulses, the amplitudes of which are

proportional to the amplitudes of the individual channels at their respective samplinginstants. This is illustrated in Fig. 5

i

Fig 5 : PAM Output Signals

The original signal for each channel can be recovered at the receive end by

applying gate pulses at appropriate instants and passing the signals through low pass filters.(Refer Fig. 6).

Fig. 6 : Reconstruction of Original Signal


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Quantization

In PCM, we convey the speech in discrete form. The sampler selects a number of

points on the analogue speech signal (by sampling process) and measures their instantvalues. The output of the sampler is a PAM signal as shown in Fig. 3; Therefore, in PCM

systems, PAM signals are converted into digital form by using Quantization

Principles. The discrete level of each sampled signal is quantified with reference to acertain specified level on an amplitude scale.

The process of measuring the numerical values of the samples and giving them

a table value in a suitable scale is called "Quantising". Of course, the scales and thenumber of points should be so chosen that the signal could be effectively reconstructed

after demodulation.

Quantising, in other words, can be defined as a process of breaking down a

continuous amplitude range into a finite number of amplitude values or steps.

The discrete value of a sample is measured by comparing it with a scale having a

finite number of intervals and identifying the interval in which the sample falls. The

finite number of amplitude intervals is called the "quantizing interval". Thus, quantizingmeans to divide the analogue signal's total amplitude range into a number ofquantizing intervals and assigning a level to each. intervals.

Quantizing Process

Suppose we have a signal as shown in Fig. 7 which is sampled at instants a, b,c, d and e. let us suppose that the signal has maximum amplitude of 7 volts.

In order to quantize these five samples taken of the signal, let us say the totalamplitude is divided into eight ranges or intervals as shown in Fig. 7. Sample (a) lies in the

5th range. Accordingly, the quantizing process will assign a binary code correspondingto this i.e. 101, Similarly codes are assigned for other samples also. Here the

quantizing intervals are of the same size. This is called Linear Quantizing.

FIG. 7: QUANTIZING-POSITIVE SIGNAL


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Quantizing is done for both positive and negative swings. As shown inFig.6, eight quantizing levels are used for each direction of the

analogue signal. To indicate whether a sample is negative withreference to zero or is positive with reference zero, an extra digit is

added to the binary code. This extra digit is called the "sign bit". In Fig.

8. positive values have a sign bit of ' 1 ' and negative values have signbit of'0'.

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

3.1.1 Relation between Binary Codes and Number of levels.Because the quantized samples are coded in binary form, the quantization intervals

will be in powers of 2. If we have a n bit code, then we can have

Quantization level=2^n

Practical PCM systems use an eight bit code with the first bit as sign bit. It means wecan have 2" = 256 (128 levels in the positive direction and 128 levels in the negative

direction) intervals for quantizing.

..

Encoding

Conversion of quantised analogue levels to binary signal is called encoding. Torepresent 256 steps, 8 level code is required. The eight bit code is also called an eight bit

"word".


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The 8 bit word appears in the form

P ABC WXYZ

Polarity bit 1 Segment Code Linearencoding

for + ve 'O' for - ve. in the segment

The first bit gives the sign of the voltage to be coded. Next 3 bits gives the segment

number. There are 8 segments for the positive voltages and 8 for negative voltages.Last 4 bits give the position in the segment. Each segment contains 16 positions.

Referring to Fig. 9(b), voltage Vc will be encoded as 1 1 1 1 0101.

FIG. 9 (b) : Encoding Curve with Compression 8 Bit Code

The quantization and encoding are done by a circuit called coder. The coderconverts PAM signals (i.e. after sampling) into an 8 bit binary signal. The coding is

done as per Fig. 9 which shows a relationship between voltage V to be coded andequivalent binary number N. The function N = f(v) is not linear.

The curve has the following characteristics.

y It is symmetrical about the origins. Zero level corresponds to zero voltage to be

encoded.

y It is logarithmic function approximated by 13 straight segments numbered 0 to 7in positive direction and 'O' to 7 in the negative direction. However 4 segments

0, 1, 0, 1 lying between levels + vm/64 -vm/64 being colinear are taken as onesegment.


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y The voltage to be encoded corresponding to 2 ends of successive segments are inthe ratio of 2. That is vm, vm/2, vm/4, vm/8, vm/16, vm/32, vm/64, vm/128 (vm

being the maximum voltage).

There are 128 quantification levels in the positive part of the curve and 128 in the

negative part of the curve. In a PCM system the channels are sampled one by one by

applying the sampling pulses to the sampling gates. Refer Fig. 10. The gates open onlywhen a pulse is applied to them and pass the analogue signals through them for theduration for which the gates remain open. Since only one gate will be activated at a given

instant, a common encoding circuit is used for all channels. Here the samples are quantizedand encoded. The encoded samples of all the channels and signals etc are combined in the

digital combiner and transmitted.

Fig. 10

The reverse process is carried out at the receiving end to retrieve the originalanalogue signals. The digital combiner combines the encoded samples in the form of

"frames". The digital separator decombines the incoming digital streams into individual

frames. These frames are decoded to give the PAM (Pulse Amplitude Modulated)samples. The samples corresponding to individual channels are separated byoperating the receive sample gates in the same sequence i.e. in synchronism with the

transmit sample gates.


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CONCEPT OF FRAME

In Fig. 10, the sampling pulse has a repetition rate of Ts and a pulse width of

"St". When a sampling pulse arrives, the sampling gate remains opened during the time"St" and remains closed till the next pulse arrives. It means that a channel is activated for

the duration "St". This duration, which is the width of the sampling pulse, is called the

"time slot" for a given channel.

Since Ts is much larger as compared to St. a number of channels can be sampledeach for a duration of St within the time Ts. With reference to Fig. 10, the first sample of

the first channel is taken by pulse 'a', encoded and is passed on the combiner. Then the firstsample of the second channel is taken by pulse 'b' which is also encoded and passed on to

the combiner, Likewise the remaining channels are also sampled sequentially and areencoded before being fed to the combiner. After the first sample of the Nth channel is

taken and processed, the second sample of the first channel is taken, this process isrepeated for all channels. One full set of samples for all channel taken within the

duration Ts is called a "frame". Thus the set of all first samples of all channels is one frame;the set of all second samples is another frame and so on.

For a 30 channel PCM system, we have 32 time slots.

Thus the time available per channel would be 3.9 microsecs.

Thus for a 30 channel PCM system,

Frame = 125 microseconds

Time slot per channel = 3.9 microseconds.

Structure of Frame

A frame of 125 microseconds duration has 32 time slots. These slots arenumbered Ts 0 to Ts 31. Information for providing synchronization between trans and

receive ends is passed through a separate time slot. Usually the slot Ts 0 carries thesynchronization signals. This slot is also called Frame alignment word (FAW).

The signaling information is transmitted through time slot Ts 16. Ts 1 to Ts 15 areutilized for voltage signal of channels 1 to 15 respectively. Ts 17 to Ts 31 are

utilized for voltage signal of channels 16 to 30 respectively.

SYNCHRONIZATION

The output of a PCM terminal will be a continuous stream of bits. At thereceiving end, the receiver has to receive the incoming stream of bits and

discriminate between frames and separate channels from these. That is, the receiver

has to recognise the start of each frame correctly. This operation is called framealignment or Synchronization and is achieved by inserting a fixed digital patterncalled a "Frame Alignment Word (FAW)" into the transmitted bit stream at regular

intervals. The receiver looks for FAW and once it is detected, it knows that in nexttime slot, information for channel one will be there and so on.

The digits or bits of FAW occupy seven out of eight bits of Ts 0 in the followingpattern.


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Bit position of Ts0 B1 B2 B3 B4 B5 B6 B7 B8

FAW digit value

X 0 0 1 1 0 1 1

The bit position B1 can be either ' 1 ' or '0'. However, when the PCM system is to belinked to an international network, t he B1 position is fixed at '1 '.

The FAW is transmitted in the Ts O of every alternate frame.

Frame which do not contain the FAW, are used for transmitting

supervisory and alarm signals. To distinguish the Ts 0 of frame carryingsupervisory/alarm signals from those carrying the FAW, the B2 bit position of the

former are fixed at T. The FAW and alarm signals are transmitted alternatively asshown in Table - 2.

TABLE-2

Frame Remark

Numbers B1 B2 B3 B4 B5 B6 B7 B8

FO X 0 0 1 1 0 1 1 FAW

F1 X 1 Y Y Y 1 1 1 ALARM

F2 X 0 0 1 1 0 1 1 FAW

F3 etc X 1 Y Y Y 1 1 1 ALARM

In frames 1, 3, 5, etc, the bits B3, B4, B5 denote various types of alarms. Forexample, in B3 position, if Y = 1, it indicate Frame synchronization alarm. If Y = 1 in

B4, it indicates high error density alarm. When there is no alarm condition, bits B3B4 B5 are set 0. An urgent alarm is indicated by transmitting "all ones". The code

word for an urgent alarm would be of the form.

X 111 1111


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SIGNALLING IN PCM SYSTEMS

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

of a call between two subscribers, for providing certain status information like dialtone, busy tone, ring back. NU tone, metering pulses, trunk offering signal etc. All

these functions are grouped under the general terms "signaling" in PCM

systems. The signaling information can be transmitted in the form of DC pulses (asin step by step exchange) or multi-frequency pulses (as in cross bar systems) etc.

The signaling pulses retain their amplitude for a much longer period than the

pulses carrying speech information. It means that the signaling information is aslow varying signal in time compared to the speech signal which is fast changing in

the time domain. Therefore, a signaling channel can be digitized with less number of bits than a voice channel. In a 30 chl PCM system, time slot Ts 16 in each frame is

allocated for carrying signaling information.

The time slot 16 of each frame carries the signaling data

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

must transmit 15 frames, each having 125 microseconds duration. For carryingsynchronization data for all frames, one addit ional frame is used. Thus agroup of 16 frames (each of 125 microseconds) is formed to make a "multi-frame". The duration of a multi-frame is 2 milliseconds. The multi-frame has 16

major time slots of 125 microseconds duration. Each of these (slots) frames has 32time slots carrying, the encoded samples of all channels plus the signaling and

synchronization data. Each sample has eight bits of duration 0.400 microseconds (3.9/8= 0.488) each. The relationship between the bit duration frame and multi-frame is

illustrated in Fig. 11 (a) & 11 (b).

Fig. 11 (a) Multi-frame Formation


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FIG. 11 (b) 2.048 Mb/s PCM Multi-frame

We have 32 time slots in a frame; each slot carries an 8 bit word.

The total number of bits per frame = 32 x 8 = 256

The total number of frames per seconds is 8000

The total number of bits per second is 256 x 8000 = 2048 K/bits.

Thus, a 30 channel PCM system has 2048 K bits/sec.

8.0 Multi-frame Structure

In the time slot 16 of FO, the first four bits (positions 1 to 4) contain the multi-frame

alignment signal which enables the receiver to identify a multi-frame. The other four bits(no. 5 to 8) are spare. These may be used for carrying alarm signals. Time slots 16 of

frames F1 to FT5 are used for carrying the signaling information. Each frame carriessignaling, data for two VF channels. For instance, time slot Ts 16 of frame F1 carries the

signal data for VF channel 1 in the first four bits. The next four bits are used for carryingsignaling information for channel 16. Similarly, time slot Ts16 of F2 carries signalling

data of channels 2 and 17. Thus in multi-frame structure, four signaling bits are providedfor each VF channels. As each multi-frame includes 16 frames, so the signaling of each

channel will occur at a rate of 500 per sec.

bsnl summer

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