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 upgradation, 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 and Management have been running for several years now.

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 providing telecom services and network management throughout the country except the metro cities of Delhi and Mumbai of the erstwhile services providing dept.of the Govt.of India.

Type Of Company

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

Administrative Ministry

Govt. of India, Ministry of Communication and Information Technology, Department of Telecommunications. The entire share capital of the company is held by the Govt. of India. Govt. of India is holding 100% of the share capital of the company. Not applicable, as the BSNL is an unlisted company.

Details Of Investments

Shareholding Pattern Listing with Stock Exchanges

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

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 career planning. 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 the main 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 y y y I.T.I. Bangalore for the manufacturing for the carriers, VFT, Coaxial and microwave equipment. I.T.I. Gonda for the manufacturing of E-10B electronic exchange equipment. Hindustan Cables LTD. Hyderabad & Rupnarainpur for manufacturing underground cables. Hindustan teleprinters LTD. Chennai for manufacturing teleprinters.

HISTORY OF TELEPHONE EXCHANGE

The Telephone was invented by Mr. Graham Bell. During early stage of development of telephone exchange, the connections are established with help of human operator.Those type of exchange were called manual Telephone Exchange. The technology of telephony was going on progress with the introduction of automatic exchange. Manual telephone is replaced & automatic exchange became in use there were lot 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 the subscriber 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 development of automatic telephone exchange technology as a subscriber directly & it has many advantages over manual telephone exchange. Now a day electronic Automatic exchange is widely used due to their advantages. YEAR 1876 1915 1920 1956 1962 1965 1974 1977 1980s 1990s INCIDENTS Invention of Telephone first transcontinental telephone(NY-SF) first automatic switches TAT-1 transatlantic cables(35 lines) Digital transmission(T1) 1ESS analog switches Internet packet voice 4ESS digital switches Signaling System (out-of-band) Advanced Intelligent Network (AIN)

ADVANTAGES OF ELECTRONIC EXCHANGES OVER ELECTROMECHANICAL EXCHANGES

y y

y y y y y 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. Electromechanical exchanges have limited flexibility while electronic exchanges are highly flexible. In electromechanical exchanges any change in facilities require addition of hardware changes whereas in electronic exchanges can be carried out by simple commands. 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. In electromechanical exchanges, there is partially full-availability hence blocking problem is there. Electronic exchanges are fully available hence no blocking. Limited facilities are available to subscriber in electromechanical exchanges than electronic exchanges. Electromechanical exchanges are slow in speed as compared to electronic exchange. Switch room occupies large volume in electromechanical exchanges. There is lot of switching noise in electromechanical exchanges as compared to electronic exchanges. Longer installation time is required in electromechanical exchanges.

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 of cable: 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 has eight 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 3. Green MADE COLOUR: 1. White 3. Black 2. Yellow 4. Red 2. Orange 4. Brown 5. Slate

Colour Combination of S.B. cable: 1. Blue-White 4. Green-White 7. Orange-Red 2.Brown-White 5. Slate-White 8. Green-Red 3. Orange-White 6. Blue-Red

There are Eight different colors of wraps in S.B. cable which are given below: 1. White 5. Grey 2. Yellow 6. Black 3. Brown 7. Pink 4. Blue 8. Green

UNDER GROUND CABLE: U.G. cables are used for outdoor process. By the cable we 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 is Made Color like S.B. cable. Chart of color coding is given below:

Made colour Prime colour Blue Orange Green Brown slate

WHITE

RED

YELLOW

BLACK

1 2 3 4 5

6 7 8 9 10

11 12 13 14 15

16 17 18 19 20

The pair of colour is made according to this chart respectively

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 many similarities 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 validated in 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-25 version.The basic architecture remaining same, more facilities to subscriber and administration are supported by later versions.

O-----------------ORGAN C-----------------COMMAND OR CONTROL B-----------------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 2048 PCM 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 added services. 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 software of 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 at a remote place & connected to main exchange through PCM links. Further line

connectors can also be placed at remote location & connected to CSNL or CSND through PCM 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. This avoid 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 be connected. 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 or several 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 cards required 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 by adding 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 connected SMMs,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 to magnetic 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 ceiling are not essential. Air conditioning requirement are also not stringent. This system can work at temperature 5 degree Celsius to 45 degree Celsius, though optimum temperature is 22 degree Celsius.

A Typical Telephone Exchange -OCB-283

FUNCTIONAL ARCHITURE The Alcatel E10 system is located at the heart of the telecommunication networks concerned. It is made up of three independent functional units: The Subscriber Access Subsystem which carries out connection of analogue and digital subscriber lines, Connection and Control which carries out connections and processing of calls, Operation and Maintenance which is responsible for all functions needed by the network operating authority.

Each functional unit is equipped with softwares which are appropriate for handling the functions for which it is responsible. HARDWARE CONFIGURATION OCB 283 exchange comprises following hardware units.

1. 2. 3. 4. 5. 6. 7. 8.

Subscriber Access Units(CSNL, CSND, CSED) Trunk and Junction connection Units(SMT) Switching Matix(SMX) Auxiliary Equipments(SMA) Control Units(SMC) Communication Multiplex Time Base Generator(BTS) Operation and Maintenance Unit(SMM)

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 be either placed in the exchange switch room or at a remote location. Depending upon their location, 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 digital control unit in which case they are known as local concentrators CNE. The max.capacity of a concentrator is 256 subscribers.

Subs

CNL

UCX
Subs CNE ICNE

SMX

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 also known as PCM trunk control station. This SMT provides an operational interface between the PCMs coming from the exchange (CSND or CSNE) and the switching center. The current version of SMT being supplied to our exchanges in India is SMT2G. It is new functional variant of the SMT station.

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 y y y y y y Clock reception and distribution Control of the station Interface with the connection units and the other SMX stations Connection of CX input lines to 256 CX output lines Connection security Help in fault location of the LOCOVAR Station alarm processing

The switching network in OCB283 is single T stage system. It is made up of a) Hot Switching Matrix b) Branch Selection and Amplification(SAB)

4) Auxiliary Equipment Control station(SMA):

The SMA station receives the auxiliaries from the OCB283 exchange. These are Frequency receive/generator used for setting up calls Conference circuits Tone generators, time management operator CCITT number 7 signaling receiver/transmitter. ETA PUPE

y y y y i. ii.

The SMA contains the following two functional units.

ETA contains the following subscriber components: y y

y

Frequency receiver/generators Conference call circuits 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

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 (PUS)

Secondary Processor Unit 4(PUS) (PUS)

Main Coupler (CMP)

Secondary Coupler (CMS)

Secondary Coupler 4 (CMS)

Specific Coupler

CCITT N07 SIGNALLING NETWORK

TELEPHONE SUBCRIBER ACCESS SUBSYSTE M CONNECTION AND DATA NETWORK NETWORK

NT

CONTROL

VALUE ADDED NETWORK

OCB 283

OPERATION AND MAINTENANCE

OPERATION AND MAINTENANCE NETWORK

PABX

ALCATEL 1000 E10

OCB 283

MULTIPLEXING TECHNIQUES
There are basically two types of multiplexing techniques i. ii Frequency Division Multiplexing (FDM) Time Division Multiplexing (TDM)

Frequency Division Multiplexing Techniques (FDM) The FDM techniques is the process of translating individual speech circuits (3003400 Hz) into pre-assigned frequency slots within the bandwidth of the transmission medium. The frequency translation is done by amplitude modulation of the audio frequency with an appropriate carrier frequency. At the output of the modulator a filter network is connected to select either a lower or an upper side band. Since the intelligence is carried in either side band, single side band suppressed carrier mode of AM is used. This results 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 varying signals.

Fig. 1 FDM Principle Time Division Multiplexing Basically, time division multiplexing involves nothing more than sharing a transmission medium by a number of circuits in time domain by establishing a sequence of time slots during which individual channels (circuits) can be transmitted. Thus the entire bandwidth is periodically available to each channel. Normally all time slots1 are equal in length. Each channel is assigned a time slot with a specific common repeatition period called a frame interval. This is illustrated in Fig. 2.

Fig. 2 Time Division Multiplexing Each 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 the receiving 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 given instant 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 time sequence for transmission over a common medium is called Time Division Multiplexing (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 same transmission medium viz open wire or underground cable pair or a channel provided by carrier, coaxial, microwave or satellite system.

Basic Requirements for PCM System To develop a PCM signal from several analogue signals, the following processing steps are required 3.1 3.2 Filtering Sampling Quantisation Encoding Line Coding

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

It is the most basic requirement for TDM. Suppose we have an analogue signal Fig. 3 (b), which is applied across a resistor R through a switch S as shown in Fig. 3 (a) . Whenever switch S is closed, an output appears across R. The rate at which S is closed is called the Sampling frequency. Fig. 3(d) is a stream of samples of the input signal which appear across R. The amplitude of the sample is depend upon the amplitude of the input signal at the instant of sampling. The duration of these sampled pulses is equal to the duration for which the switch S is closed.

Fig. 3: Sampling Process

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 to or more than twice the highest signal frequency in the band, then the sample contains all the information of the original signal." Mathematically, if fH is the highest frequency in the signal to be sampled then the sampling frequency Fs needs to be greater than 2 fH. i.e. Fs>2fH Let us say our voice signals are band limited to 4 KHz and let sampling frequency be 8 KHz. Time period of sampling Ts = or Ts = 125 micro seconds If we have just one channel, then this can be sampled every 125 microseconds and the resultant samples will represent the original signal. But, if we are to sample N channels one by one at the rate specified by the sampling theorem, then the time available for sampling each channel would be equal to Ts/N microseconds. 1 sec 8000

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 by a series of pulses called "Clock pulses". These are called gates because, when closed these actually connect the channels to the transmission medium during the clock period and

isolate them during the OFF periods of the clock pulses. The clock pulses are staggered so that only one pair of gates is open at any given instant and, therefore, only one channel is connected to the transmission medium. The time intervals during which the common transmission medium is allocated to a particular channel is called the Time Slot for that channel. The width of this time slot will depend, as stated above, upon the number of channels to be combined and the 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. Thus in 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 sampling instants. 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

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 instant values. 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 a certain specified level on an amplitude scale. The process of measuring the numerical values of the samples and giving them a table value in a suitable scale is called "Quantising". Of course, the scales and the number of points should be so chosen that the signal could be effectively reconstructed after demodulation. Quantising, in other words, can be defined as a process of breaking down a continuous amplitude range into a finite number of amplitude values or steps. The discrete value of a sample is measured by comparing it with a scale having a finite number of intervals and identifying the interval in which the sample falls. The finite number of amplitude intervals is called the "quantizing interval". Thus, quantizing means to divide the analogue signal's total amplitude range into a number of quantizing intervals and assigning a level to each. intervals. 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 total amplitude is divided into eight ranges or intervals as shown in Fig. 7. Sample (a) lies in the 5th range. Accordingly, the quantizing process will assign a binary code corresponding to this i.e. 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

Quantizing is done for both positive and negative swings. As shown in Fig.6, eight quantizing levels are used for each direction of the analogue signal. To indicate whether a sample is negative with reference to zero or is positive with reference zero, an 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 sign bit 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 we can 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. To represent 256 steps, 8 level code is required. The eight bit code is also called an eight bit "word".

The 8 bit word appears in the form P Polarity bit 1 encoding for + ve 'O' for - ve. ABC Segment Code WXYZ Linear 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 coder converts 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 and equivalent binary number N. The function N = f(v) is not linear. The curve has the following characteristics. y y It is symmetrical about the origins. Zero level corresponds to zero voltage to be encoded. It is logarithmic function approximated by 13 straight segments numbered 0 to 7 in positive direction and 'O' to 7 in the negative direction. However 4 segments 0, 1, 0, 1 lying between levels + vm/64 -vm/64 being colinear are taken as one segment.

The voltage to be encoded corresponding to 2 ends of successive segments are in the ratio of 2. That is vm, vm/2, vm/4, vm/8, vm/16, vm/32, vm/64, vm/128 (vm being the maximum voltage).

There are 128 quantification levels in the positive part of the curve and 128 in the negative part of the curve. In a PCM system the channels are sampled one by one by applying the sampling pulses to the sampling gates. Refer Fig. 10. The gates open only when a pulse is applied to them and pass the analogue signals through them for the duration for which the gates remain open. Since only one gate will be activated at a given instant, a common encoding circuit is used for all channels. Here the samples are quantized and encoded. The encoded samples of all the channels and signals etc are combined in the digital combiner and transmitted.

Fig. 10 The reverse process is carried out at the receiving end to retrieve the original analogue signals. The digital combiner combines the encoded samples in the form of "frames". The digital separator decombines the incoming digital streams into individual frames. These frames are decoded to give the PAM (Pulse Amplitude Modulated) samples. The samples corresponding to individual channels are separated by operating the receive sample gates in the same sequence i.e. in synchronism with the transmit sample gates.

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 sampled each for a duration of St within the time Ts. With reference to Fig. 10, the first sample of the first channel is taken by pulse 'a', encoded and is passed on the combiner. Then the first sample of the second channel is taken by pulse 'b' which is also encoded and passed on to the combiner, Likewise the remaining channels are also sampled sequentially and are encoded before being fed to the combiner. After the first sample of the Nth channel is taken and processed, the second sample of the first channel is taken, this process is repeated for all channels. One full set of samples for all channel taken within the duration Ts is called a "frame". Thus the set of all first samples of all channels is one frame; the set of all second samples is another frame and so on. 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 are numbered 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 the synchronization 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 are utilized for voltage signal of channels 1 to 15 respectively. Ts 17 to Ts 31 are utilized for voltage signal of channels 16 to 30 respectively. SYNCHRONIZATION The output of a PCM terminal will be a continuous stream of bits. At the receiving end, the receiver has to receive the incoming stream of bits and discriminate between frames and separate channels from these. That is, the receiver has to recognise the start of each frame correctly. This operation is called frame alignment or Synchronization and is achieved by inserting a fixed digital pattern called a "Frame Alignment Word (FAW)" into the transmitted bit stream at regular intervals. The receiver looks for FAW and once it is detected, it knows that in next time slot, information for channel one will be there and so on. The digits or bits of FAW occupy seven out of eight bits of Ts 0 in the following pattern.

Bit position of Ts0 FAW digit value

B1 X

B2 0

B3 0

B4 1

B5 1

B6 0

B7 1

B8 1

The bit position B1 can be either ' 1 ' or '0'. However, when the PCM system is to be linked to an international network, the B1 position is fixed at '1'. The FAW is transmitted in the Ts O of every alternate frame. Frame which do not contain the FAW, are used for transmitting supervisory and alarm signals. To distinguish the Ts 0 of frame carrying supervisory/alarm signals from those carrying the FAW, the B2 bit position of the former are fixed at T. The FAW and alarm signals are transmitted alternatively as shown in Table - 2. TAB LE- 2 Frame Numbers FO F1 F2 F3 etc B1 X X X X B2 0 1 0 1 B3 0 Y 0 Y B4 1 Y 1 Y B5 1 Y 1 Y B6 0 1 0 1 B7 1 1 1 1 B8 1 1 1 1 FAW ALARM FAW ALARM Remark

In frames 1, 3, 5, etc, the bits B3, B4, B5 denote various types of alarms. For example, in B3 position, if Y = 1, it indicate Frame synchronization alarm. If Y = 1 in B4, it indicates high error density alarm. When there is no alarm condit ion, bits B3 B4 B5 are set 0. An urgent alarm is indicated by transmitting "all ones". The code word for an urgent alarm would be of the form. X 111 1111

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 dial tone, 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 (as in 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 a slow 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 carrying synchronization data for all frames, one additional frame is used. Thus a group of 16 frames (each of 125 microseconds) is formed to make a "multiframe". 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 32 time slots carrying, the encoded samples of all channels plus the signaling and synchronization data. Each sample has eight bits of duration 0.400 microseconds (3.9/8 = 0.488) each. The relationship between the bit duration frame and multi-frame is illustrated in Fig. 11 (a) & 11 (b).

Fig. 11 (a) Multi-frame Formation

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 carries signaling, data for two VF channels. For instance, time slot Ts 16 of frame F1 carries the signal data for VF channel 1 in the first four bits. The next four bits are used for carrying signaling 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 provided for 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.