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INDEX1.About BSNL 2.About Alcatel 3.Description Of E-10B 4.Pulse Code Modulation 5.Conclusion


Bharat Sanchar Nigam Ltd. formed in October, 2000, is World's 7th largest Telecommunications Company providing comprehensive range of telecom services in India: Wireline, CDMA mobile, GSM Mobile, Internet, Broadband, Carrier service, MPLS-VPN, VSAT, VoIP services, IN Services etc. Presently it is one of the largest & leading public sector unit in India. BSNL has installed Quality Telecom Network in the country and now focusing on improving it, expanding the network, introducing new telecom services with ICT applications in villages and wining customer's confidence. Today, it has about 46 million line basic telephone capacity, 8 million WLL capacity, 52 Million GSM Capacity, more than 38302 fixed exchanges, 46565 BTS, 3895 Node B ( 3G BTS), 287 Satellite Stations, 614755 Rkm of OFC Cable, 50430 Rkm of Microwave Network connecting 602 Districts, 7330 cities/towns and 5.6 Lakhs villages. BSNL is the only service provider, making focused efforts and planned initiatives to bridge the Rural-Urban Digital Divide ICT sector. In fact there is no telecom operator in the country to beat its reach with its wide network giving services in every nook & corner of country and operates across India except Delhi & Mumbai. Whether it is inaccessible areas of Siachen glacier and North-eastern region of the country. BSNL serves its customers with its wide bouquet of telecom services. BSNL is numero uno operator of India in all services in its license area. The company offers vide ranging & most transparent tariff schemes designed to suite every customer. BSNL cellular service, CellOne, has 55,140,282 2G cellular customers and 88,493 3G customers as on 30.11.2009. In basic services, BSNL is miles ahead of its rivals, with 35.1 million Basic Phone subscribers i.e. 85 per cent share of the subscriber base and 92 percent share in revenue terms. BSNL has more than 2.5 million WLL subscribers and 2.5 million Internet Customers who access Internet through various modes viz. Dial-up, Leased Line, DIAS, Account Less Internet(CLI). BSNL has been adjudged as the NUMBER ONE ISP in the country. BSNL has set up a world class multi-gigabit, multi-protocol convergent IP infrastructure that provides convergent services like voice, data and video through the same Backbone and Broadband Access Network. At present there are 0.6 million DataOne broadband customers. The company has vast experience in Planning, Installation, network integration and Maintenance of Switching & Transmission Networks and also has a world class ISO 9000 certified Telecom Training Institute. Scaling new heights of success, the present turnover of BSNL is more than Rs.351,820 million (US $ 8 billion) with net profit to the tune of Rs.99,3902

million (US $ 2.26 billion) for last financial year. The infrastructure asset on telephone alone is worth about Rs.630,000 million (US $ 14.37 billion). The turnover, nationwide coverage, reach, comprehensive range of telecom services and the desire to excel has made BSNL the No. 1 Telecom Company of India.

BSNL SERVICESBSNL provides almost every telecom service in India. Following are the main telecom services provided by BSNL:

Universal Telecom Services : Fixed wireline services & Wireless in Local loop (WLL) using CDMA Technology called bfone and Tarang respectively. As of December 31, 2007, BSNL has 81% marketshare of fixed lines. Cellular Mobile Telephone Services: BSNL is major provider of Cellular Mobile Telephone services using GSM platform under the brand name BSNL Mobile[4]. As of Sep 30, 2009 BSNL has 12.45% share of mobile telephony in the country[5]. Internet: BSNL provides internet services through dial-up connection (Sancharnet) as Prepaid, (NetOne) as Postpaid and ADSL broadband (BSNL Broadband). BSNL has around 50% market share in broadband in India. BSNL has planned aggressive rollout in broadband for current financial year. Intelligent Network (IN): BSNL provides IN services like televoting, toll free calling, premium calling etc. 3G:BSNL offers the '3G' or the'3rd Generation' services which includes facilities like video calling etc. IPTV:BSNL also offers the 'Internet Protocol Television' facility which enables us to watch television through internet. FTTH:Fibre To The Home facility that offers a higher bandwidth for data transfer.This idea was proposed on post-December 2009.

New Services introduced/planned by BSNL

Prepaid Broadband:BSNL is also offering prepaid Broadband services. The customers availing prepaid broadband have many advantages over post paid broadband like control on usage, Mobility etc.3

Wi-MAX: In addition to wireline broadband services, BSNL is also in the process of rolling out its Wi-MAX network in rural areas to take an initial lead and provide wireless broadband services in all rural blocks in the country during 2010-11. The Urban Wi-Max is also being deployed in Kerala & Punjab Circles and shall cover all the mojor cities in these circles. Wi-Max services are also being provided through a Franchisee agent with M/s SOMA in three states of Gujrat, AP and Maharashtra Value Added Services : BSNL is focussing on provision of value added services/features to attract high end customers and to double its revenues from VAS Mobility in WLL: BSNL is planning to provide full mobility on its WLL network.

ABOUT ALCATELHistoryAlcatel has a long history begininig in 1898 with the founding of Compagine Generale d'Electricite(CGE). The original home of the company was the Alsace region and it still maintains R&D operations in the Strasbourg area. The current4

name of "Alcatel", comes from the acquisition in 1968 of Societe Alsacienne de constructions atomiques,de Telecommunications et d' Electronique In 1991 CGE changed its name to Alcatel Alsthom,and in 1998 to Alcatel There were a number of merges and acquisitions as well as divestments since 1898.To understand the current company and its focus on telecommunications,the most important were the acquisition of the European telecommunications, the most important were the acquisition of the European telecommunications activities of ITT 1986. The combined companies were called Alcatel Alsthom. Alcatel maintains significant R&D presence in France(Paris region, Brittany, south of France), Antwerp (Belgium), at the former ITT operatoins there (Bell Telephone), in Stuttgart (Germany), in Italy(Vimercate, genoa, Rieiti,Battiapaglia), in India(Gurgaon, Noida and Chennai), since 2000 in Shanghai(China)and since 2005 in Saint-Petersburg(Russia). Since 1990, various North America companies were acquired- Spatial Wireless, Rockwell Technologies, DSC communications, Xylan, Packet Engines,Assured Access, Newbridge, iMagicTV, TiMetra, and eDial - giving alcatel a strong U.S. and Canadian presence. Alcatel has its North American headquaters in Plano, Texas, and R&D operations in Ottawa, Mountain View, California, Petalumn, California, Saint John, New brunswick, Calabasas, California, Murray, Utah, and, Raleigh, North Carolina Early in 2006, Alcatel setup a new joint venture with TCL of china forming a new mobile business, TCL and Alcatel Mobile Phones Limited(TAMP).

Alcatel-LucentAlcatel-lucent is one of the world's biggest industryplayers in telecommunications that provides hardware,software and services to service providers and enterprise all over the globe. The company is incoprated in france and has it headquaters at rue de la Boetie in Paris. The company does business in more than 130 countries,with almost equal sales distribution coming from both its European and North American regions, and an additional third of its channel located elsewhere ni the world. Alcatel-Lucent was formed after Alcatel merged with Lucent technologies on december 1,2006.

Areas of businessAlcatel was mostly well known for its DSL multiplexers, used for high speeed internet access over ADSL and VDSL, wheras Lucent was well- known for its class 4 nad 5 voice switches (central office) and its optical products, Alcatel had over 40% of the world DSLAM market in 2007, with more than 143 million lines shipped and has been evolving this from an ATM-backhauled device to an IP-backhauld device. It has a partnership with Microsoft as of 2004 to provide IPTV services via its TPSDA(Triple5

Play Servives Delivery Architecture) over DSL and using its 7*50 VPLS/MPLS routers and switches to service provides such as AT&T in the united states. It also leading provider of optical transmission equipment, especially for submarine communication cable. Genesys, a U.S. subsidary, is a leading provider of a call centre software which operates both with Alcatel-Lucent equipment and 3rd-party equipment. The company is generally organised into carrier, enterprise and services business groups. Alcatel-Lucent is also thebworld leader in point-to-point microwave radios and wireless transmission; with over 50 years expertise and over 17% of the global market share in 2005, it has a field-proven experience in deploying and managing wireless transmission networks and 3G, 2.5G, and 3G mobile backhauling. Alcatel-Lucent has several notable non-network-based businesses. It has a transport solutions division that provides routing, control and network mangement for railway and mass tranisit operators, such as city undergrounds in Berlin, London and New York.

Types of Exchanges Used in BSNLThere are various types of digital switching systems used in BSNL: E10B SWICHING SYSTEM C-DOT DIGITAL SWICHING SYSTEM(MAX) EWSD SWICHING SYSTEM6


E-10B SWITCHING SYSTEMEvolution and architecture of E-10B-System1.0. IntroductionThe policy perused by Telecom. Administrations all over the world at presentation introduced digital switching in their network in view of significance techno economic advantage of digital network. The digitalization of our network have begun in right earnest by large scale introduction of digital switching and digital7

transmission system. At present, E-10B electronic digital switching developed by CIT Alcatel of France is being introduced into our network on a very large scale.2.0.

Present status of digital switching in Indian network:For our 2 lac. E-10B Local lines were imported from the manufacturer M/s CID Alcatel of France. These equipments have been installed at 23 stations. The first system (A training model of E-10B Exchange) was commissioned at ALTTC/Ghaziabad July/8f4. The first commercial E-10B Local exchange of 10,000 lines capacity was commissioned at Bombay


Evolution of E 10-B system:The E-10B system is the culmination of a massive R&D effort in the filed of digital switching systems over many years in FRANCE. It has evolved from the E-10A system which was commissioned in French network in early seventies. Based on the architecture of E-10A system, a more powerful system, having a significantly higher call handling/traffic capacity was developed in early 80s the first E-10B system was commissioned in June 81 at BREST in FRANCE. Since then, the system has been in operational use in several countries. The system has several versions. The version supplied to India is the 384 PCM versions which can handle a maximum traffic of 4000 Erlangs. The number of Busy Hour Call Attemps (BHCA) that the system can handle is found 1, 90,000

4.0. Applications of E-10B system:(a) Local Exchanges These exchanges terminate local subscriber lines and are connected to other exchanges in the local network, within the limits of the maximum traffic handling capacity viz. 4000 Erlangs, any proportion of subscribers and junctions is possible. (b) Local, Transit or tandem Exchanges E-10B system can be used to carry pure transit traffic in this case, subscribers 1 in terminating equipments will not be provided. Only equipments needed for connecting the junctions will be provided. (c) TAX When used as a tax, the system provides for the termination of long distance circuits. Digital TAX of the E-10B type have already been commissioned at various stations. The maximum capacity of an E-10B Digital TAX is limited to 1100 (O/G and I/C in the 384 PCM versions. (d) Local-cum-transit or TAX8

It is possible to combine the functions of local and transit or TAX exchanges in an E-10B system.

5.0. Facilities and services to subscribersE-10B system offers many Custom Calling Services to its subscribers. Some of the popular services are: Follow-me-service (Call forwarding) Short code dialing abbreviated dialing. Identification of malicious calls Call waiting indication (camp-on busy) Detailed (Itemised billing) Automatic alarm call (wake me service) Barred access Hotline facility. DTMF pushbutton telephones. Last number redials. Thanks to its modular structure E-10B system is easily expandable to meet increased demands. It is a system with an open architecture i.e. it can fit in with any existing network configuration. New services can be easily introduced by modification of software, in most of the cases

5.1. Facilities to connect remote subscribers concept of RLUs (Fig.1)



It is possible to extend telecommunication facilities to remote subscribers eg. Subscribers situated at the outskirts of metropolitan centre. These subscribers can be served by remote line units or RLUs. These RLUs are called URADs (Distant Subscribers connection units) these are also known as CSEDs (Distant Electronic Spatial concentrators) Each RLU caters to 1023 remote subscribers and has to be parented to the main E-10B exchange (situated in distance is no constraint in locating these RLUs. The remote subscribers get all the facilities available to the local (main E-10B subscribers. Cabling costs are reduced drastically since the subscribers are now connected by short cable pairs straight to the RLUs instead of to the distant main exchange. However, the URAD (CSED) has no stand-alone capability, i.e. in case of failure of PCM between URAD and parent exchange, the URAD is isolated, i.e. subscribers can neither make any local or outgoing calls nor can receive incoming calls. Subscribers can security call. This facility enables them to get access to predetermined emergency services only viz. police, Fire and Ambulance by dialing special codes. The E-10B URADs are being progressively utilize to fulfill the telecom needs of sparsely populated areas having a community of telecom interest with main towns/urban centers.10


Introduction of E-10B in the Analogue network (Fig.2)

The E-10B system can be introduced into an existing analog network with ease. The analogue electromechanical systems e.g. strowger, crossbar local and TAX system and analog electronic systems like PRX-A, ND-10, FETEX-100L need interface equipments before being connected to the E-10B system. Two interfaces are digital signaling acceptable by E-10B and the other to carry out analog to digital (A to D) conversion (and vice versa). The signaling conversion is carried out by a group of signaling converters known as GAS (Signaling Adapter Group). The A/D and D/A conversion is performed by the PCM multiplexing equipment called TNE or Digital Terminal Equipment. The equipments are situated ideally at the analogue exchange ends. The output of TNE stream. The TNE/GAS equipments are situated in the same rack. Each fully equipped 180 junctions.11

7.0. Basic principles and architecture of E-10B system: 7.1. Basic featuresThe system is based on the following salient features: 1. 2. 3. 4. 5. Stores program control (SPC) TDM digital switching. PCM principles and techniques. Segregation of switching and management functions. Distributed control using dedicated microprocessors (e.g. INTEL 8085) or minicomputer (e.g. ELS-48) 6. Centralized management for group of E-10B exchanges. The summary of Principal features of E-10B system is given in appendix-17.1.1. Stores Program control:

The control functions relating to call processing are carried out by execution of program instructions stores in the memories of computers. It is electromechanical system, these functions are hardware based, while in electronic systems like E-10B these are mostly realized in software. 7.1.2. TDM digital switching: The system switches signaling digital form. Analog signals are converted into Time-division multiplexed digital signals prior to switching. 7.1.3. PCM principles The system has been developed for 30 channels PCM corresponding to relevant CCITT recommendations. 7.1.4. Segregation between switching and management functions. Switching functions like reception of dialed digital their storage, analysis, routing of the call etc. are preformed by the control units in the exchange, which have a decentralized architecture employing dedicated processors. Functions like subscriber lines and circuit group managements, faults and alarm management, testing and diagnostics etc. are performed by a separate mini computer located at centralized operations and maintenance centre (OMC) which is common for a number of E-10B exchanges. The OMC and the switching center can be collocated in the same premises or can be situated apart. The two are interconnected by PCM links (Fig.3)


The various call handling and call-processing functions status, receptions and storage of digitas, analysis and routing, metering etc. are distributed over various functional units. Dedicated micro processors INTEL 8085 and dedicated mini computer ELS-4B handle these functions.

7.1.6. Centralized Management for E-10B Exchanges: The O & M functions for a group of E-10B exchanges (up to a maximum of 6 exchanges or 80,000 lines) are carried out by a single OMC which is connected to the various exchanges by PCM links.



The E-10B exchange consists of three blocks or subsystems:14

1. Connection units. 2. Switching network 3. Control units. 7.2.1. Connection units: The connection units act as interface between the external environment (viz. subs and trunks and the central units. The unit switches manage the generation and transmission of digitalized tones and frequencies and dissemination of recorded announcements to subscribers/trunks are also categorized as connection units. 7.2.2. Switching network (CX): This is a time division digital switching network having a 3 stage Time-SpaceTime architecture. It employs four wire (4w) switching for connecting the time slots of calling and called parties. 7.2.3. Control Units: The control units mainly handle telephone call set up, supervision, clear down and charging functions

Appendix-ISummary of principal features of the E-10B system:


Number of switch able PCM links: 384 Processing capacity: 190,000 BHCA Traffic handling capacity: 4,000 Erlangs Subscriber exchange: 45,000 lines & 5,000 circuits(Typical Values)


Times division switching. Pulse code modulation (PCM) to CCITT standards 2 Mbits /PCM link 30 telephone channels per PCM link. 8 bits per telephone channel Stored program control (SPC) Dedicated processors for switching functions(Intel 8085) Non dedicated processor for operation functions15

SUBSCRIBER LINE: Dial or push button VF telephone (CCITT) standards) Maximum loop resistance inclusive of telephone set: 2400 ohms


Exchange: Ambient temperature of air drawn into racks: 18 to 20 deg. Celsius Relative humidity: 30 to 70% Satellite exchange: Ambient temperature: 5 to 35 deg. Celcius Relative humidity: 20 to 80% OMC: Air- conditioned environment Temperature 15 to 18 deg. C (optimum 22 2 deg.C)

MECHANICAL DATA FLOOR AREA 45,000 lines subscribers Exchange = 154 sq. meter 11,000 circuit transit.Exchange =90 sq. meter Rack dimensions Height: 2.00 m Width: 0.75 Depth:0.50m Distributed floor loading: less than 500 kg/ sq. meter


Exchange and satellite exchange: -48V OMC: 220V 50 Hz Power supply current on line: 23-60 mA Loop resistance (i) 1990 ohms max, for ordinary lines (ii) 2400 ohms max for lines with long line equipment.


CONNECTION UNITS 1.0 The connection units of the E10B system as mentioned earlier provide the required interface between subscribes/trunks and the E10B Central Units. There are four types of connection units. These are: (1) (2) (3) (4) 2.0 URA- Subscriber Connection Unit. This is also known as CSE-Electronic Space Division Concentrator. URM- Multiplex Connection Unit. ETA- Frequency Sender/Receiver Unit. BDA- Auxiliary Equipment Rack.

Subscriber Connection Unit- URA (or CSE) The CSE has maximum capacity to connect 1023 subscribers. It concentrates the traffic of 1023 subscribers on to a maximum of 4 PCM links. It is managed by a duplicated microprocessor-based logic unit. In case, CSE is collocated as the control units. It is called a local subscriber connection unit and is designated as URAL or CSEL. (Fig.1)




To switching n/w


The CSE can also be located remotely from the exchange to economies on outdoor plant investment, and then it is called a Distant Subscribers connection Unit and is designated as URAD or CSED. The CSE is actually is Remote Line Unit (RLU) and is connected to the parent (Main) E10B exchange by PCM links. (Fig. 2) subscribers 4 PCMs To switching n/w





The major functions of CSE are: i. ii. iii. Power feeding for subscriber lines. Detection of line event or line loop status (off hook/on-hook) Transmission of home (Subs premises) metering pulses, battery reversal, ringing current etc towards subscriber lines. iv. Concentration of 1023 subscribers on to 120 timeslots i.e. digital channels. v. vi. vii. Search for a free time-slot and connection or disconnection of a subscribers equipment to/from a time-slot. Analogue to Digital conversion. Routine Subscriber line tests under the control or Operation and Maintenance Center (OMC)


Subsystems of CSE : The CSE can be devided into two subsystems1. 2. The speech path subsystem The control logic subsystem.

2.2.1 The speech path subsystem 1. 2. 3. Singalling equipment Concentration network Transmission equipment. Singalling equipment The Singalling equipment is the Subscriber line card called XEJ. The purpose of this unit is to feed power to telephone sets. Detect line status (off-hook/on-hook) To send the ringing current To subject the line or equipment to various tests. Where applicable to send battery reversal or send remote metering pulses for energizing the meters in subs premises. The Concentration network concentrates 1023 subscriber lines on to 120 connection paths. The concentration is acievedby using CMOS matrices.18

(1) (2) (3) (4) (5)

There are two types of XEJ printed card boards. The card in which 16 ordinary subscribers (2 wire) are accommodated is known as XEJ-16. The facility of having meters in their own premises (home-metering) can also be provided to the subscribers. Such subscribers are accommodated in another type of PCB known as XEJ-8. As the name indicates, this PCB accommodates 8 subscribers 8 subscribers only. The 16 KHz remote metering pulse generator common to all the 8 subscribers is located in this XEJ-8 PCB. The transmission equipment section of the speech path subsystem of CSE handles analogue to digital conversion and comprises 4 PCM modules.

To switching n/w From CSEPCM MODULE 0




FIG-4. TRANSMISSION EQUIPMENT The transmission equipment performs filtering, sampling and encoding functions. When distant URs are used, code converters provide also the line coding function (Binary to HDB3 and vice versa.) Each of the PCM modules are connected to the switching network (CX) by 2 Mbit/sec PCM links Called LR links. These Lr links can carry speech, tones and frequencies in the digital form (Fig. 4). 2.2.2 Control logic Subsystem: The control logic system manages the speech path subsystem and also ensures exchange of message with the central units of the E-10B exchange.19

The logic system is duplicated and is provided by Intel 8085 microprocessors, along with their memories. One of the logic system (Intel 8085) is on-line and the other in hot-stand by mode. Either of the two logic units can be the active logic at a given point of time. Only one logic has access to the speech path at any point of time, while the other logic carries out tests and updates the memories. 2.3 CSED (URAD):-The CSED is parented to Main E-10B Exchange using 2 to 4 PCMs.

These PCM Systems terminate at the URM (Multiplex connection unit) of the parent exchange. Regenerative Repeaters (RR) are provided along the PCM route. The digital line terminal (TNL) provided at each end fulfills the functions specific to the transmission medium (e.g. power feeding of repeaters, fault localization etc). 2.4 The first 2 PCM links emanating from the CSED are designated as PCM-o and PCM-1 and are called the Active PCMs and the other two PCMs designated as PCM-2 and PCM-3 are called passive PCMs. Active PCMs carry speech and signaling while passive PCMs carry, speech only. Signalling for all the four all the four the four PCMs is sent on two active PCMs only. The CSED also offers the possibility of setting local security calls In the event of a breakdown of the announcement informing him of the 2-digit number to be dialed to obtain an emergency service (fire, police, ambulance etc.) Multiplex Connection Unit (URM): The functions of URM are: 1. PCM synchronization 2. Signalling byte injection and extraction (TS-16) 3. Speech-path mixing 4. Fault localization The multiplex Connection Unit is the PCM connection interface between E10B exchange and distant subs terminate upto thirty-two, 30-Channel PCMs i.e., 960 junctions or circuits can be terminated. A URM contain 8 modules. Each module terminates 4 PCMs (i.e.) 120/ Channels, There are two types of modules (1) (2) MRM (Multiplex Connection module) MRS (Satellite connection module)




Both module types may coexist within the same URM in any required proportion.


The URM logic consists of one LOGUR system duplicate (LOGUR-0 and LOGUR-1) for security reasons. Each LOGUR consist of a Main (Master) Processor and two Auxiliary (Slave) Processors a & B which are standard intel 8085 microprocessors.

Each LOGUR can function as Pilot or as Hot-standby unit keeps its memories updated in real time to conform to control (pilot) logic. Logic switch over can take place under the following contingencies. 3.2 Under the control of the other logic. Periodically every 24 hours. Under instruction from OMC (by man-machine command) In case the URM is connected to PCM links incoming from electromechanical exchanges, analogue to digital conversion is necessary and in some cases signaling conversion is also required. TNE is used for A/D conversion and vice versa, while GAS converts DC loop signaling into E&M (digital) signaling and vice versa. Frequency sender and Receiver Unit (ETA) The functions of ETA are: Generation of tones and R2, 2 out of 5 MF Frequencies Reception of MF Push Button telephone (DTMF) frequencies, R2 MF frequencies etc and decoding them. 3. Provide conference call facility. 3.3.1 Architecture of ETA. An ETA rack is connected to the Switching Network, Multiregisters and the Monitoring unit (OC). A fully equipped ETA rack consists of 1. 2.



Three Tone generator units-GTI, GT2 and GT3. Two frequency receiver units-RFI and RF2, each having 31 basic receivers and


Two conference call circuits CCF-1 & CCF-2, each capable of handling 8 conference calls.

A microprocessor- based control logic for controlling the functions of the above units


3.3.2 An E10-8 exchanges is equipped with a minimum of two and maximum of 16 ETAs. i. ii. 4.0 Detection of all alarm conditions and forwarding them to OMC. Distribution of recorded announcements given out the (BDA) rack.

iii. Displaying the alarm conditions on the General indications. The BDA is controlled by a non-duplicated control logic module provide by 8085 microprocessor. There is only one BDA rack in an E-10B exchange. Concept of UR groups: The maximum number of connection units is 192, numbered from UR-0 to RU-191 UR No. 0 (zero) is not used. These URs are divided into 12 UR groups called GUR and numbered from GUR-0 to GUR-11. Every UR group (GUR) comprises a maximum of 16 URs. Ur-1 of GURO is allotted to BDA. A URM is considered as a single UR group and is allotted 16 UR nos (eg) 4B to 63, 64 to 79 etc.


Connection Units Links: All the exchange units in the E-10B system are interconnected by dedicated links carrying message for call processing. Connection units are connected to markers by the LU links link is multiplied to both the markers and serves a group of 16 connection units (i.e. one UR group). These links carry messages required for call processing. LT links are connected to the URA and URM. These carry loop status of subscribers (viz off-hook) and the status of circuits (ie idle, selzure etc). To allow for all possible exchange configurations, 192 LT links are provided for URAs and URMs. There are three types of links between the connection units and the switching network. These are LRE, LRS and LVS links. LRE and LRS links provide the path for speech/tone samples. There are 4 LRE links for each CSEL and for each module of URM (MRS/MRM).



In E-10B System, the switching network CX is connected to the Connection Units, To the Markers (in particular to the UGCXs ie the Switching Network Control Units associated with the marker) and the monitoring unit OC. 1.1 Major functions of CX The CX (1) interconnects calling and called timeslots and enables transfer of speech samples between them. It transmits tones, R2 frequencies, circuit test frequencies from ETA and recorded announcements from BDA towards connection units for onward transmission to subscribers/circuits. It directs DTMF (Dual Tone Multi Frequency i.e. Push button) frequencies from subscribers and R2 (2 out of 5 MF) frequencies from circuits towards ETA., for decoding into digits/appropriate signals. It transmits singalling to connection units relating to subscribers or circuits e.g. order to send ringing current to called sub, seizure signal on the circuits etc. It receives positioning commands from OC (i.e. faults messages to OC for changing of units status commands) and also faults messages to OC transmission to OMC. Traffic Handling by CX




1.2.1 Switching in CX A connection in the switching network for any call will fall in one of the four categories. i. ii. iii. iv. Subscriber Subscriber Subscriber (Local call) Circuit or junction (O/G call)

Circuit or junction-subscriber (terminating call) Circuit or junction- Circuit or Junction (Transit call)

Any such connection involves two, one-way (unidirectional) connections as below: Calling partys timeslot on the LRE- Calling partys timeslot on the appropriate LRS, where LRE and LRS are the input and output PCM highways terminated in the CX. Such Connection are possible via the CX since the and receive paths (LRS) for the speech samples. It is obvious that two timeslot are required per call. 1.2.2 Traffic Handling Capacity of CX23

Max no. of LRs = 384 LRs reserved for ancillary Functions (connection with ETA and BDA etc) 16 Usable LRs carrying speech samples = 384-16 = 368 No. of usable timeslots = 368x30 Two TS are required per call. A switching efficiency of 80% i.g. traffic of O.B. Erlang per Timeslot can be assumed. 11040 X No. of simultaneous calls = = = 2 5520 x 4416 80 100 0.8

Say 4000 (approx) By definition, average no. of simultaneous calls. Traffic carried in Erlangs. Hence maximum traffic handling capacity of CX= 4000 Erlangs (approx). 2.0 Modularity of CX In order to handle such heavy traffic, the CX is provided with a TST architecture and is organized in modular fashion. There are 24 switching modules numbered from 0 to 23. Each module terminates 16 LRs (i.e. 16 LREs and 16 LRs). Each LR can carry speech samples. Each swathe module consists of one input time stage (CTE), one space stage (CS) and one output time stage (CTS). The entire RCX is thus constituted of 24 Input Time stages, 24 space stages and 24 output Time stages. The various stages of each switch module and those of there is full flexibility for interconnecting the switch module can be connected to any timeslot of any LRS located in any switch module (Fig. 2) 4.0 Architecture of TST Each time stage is provided with its own buffer and control memories. the space stage is equipped with a multiplexer type switch matrix and associated24

control memories. The timing signals needed for the read/ write operations are derived from the mean time base(BT).4.1.1 Switching of tones through CX

The tones, R2 frequencies ( 2 out of 5 MF frequencies)m junctions test frequencies and recorded announcements( all in digital form) are connected to the output time stages of the switching modules, so as to avoid blocking while switching thro the CX for dissemination towards subscribers/circuits. Normally the even numbered switch modules (0,2,22) egt their tones, freq, supplies etc from the first ETA and the odd number ones (1,3..23) are supplied with tones etc by the second ETA. However in the case of failure of one ETA the second ETA provides thetone supply to all the switch modules for distribution. Thus, the first ETA ( ETA-1) acts as a normal source for a even switch modules and as stand by source for odd switch modules and vice- versa. TS A- allocated to sub A on the LR TB X- carries dial tone sample from ETA5.0

CX cpmfogiratopm The24 switch modules of the CX are situated in 6 racks and 4 switch modules/rack. The modularity of the CX enables the accurate dimensioning of the switch modules based on a traffic to be handled by the CX. The CX can be expanded to handle additional traffic by adding a required number of switch modules.

6.0 switching network control unit (UGCX) 6.1 the switching network control unit known as UGCX is located in the marker rack. UGCXO is associated with MQ1 and UGCX 1 with MQ2. the UGCX is connected to marker by an internal data bus and by dedicated links to CX,MR.TX. 6.2 Functions of UGCX 1.It receives tone, R2 MF frequency and circuit test frequency connection/ disconnection commands from multiregister and then forwards them to CX for execution. 2.It receives sub-signalling command to send (a) ringing current (b) home metering pulses. (c) battery reversal and (d) stop battery reversal etc. from MR/ TX and sends them to CX for transmission to subscriber connection units.25

3. it receives circuit signaling commands from MR and then sends them to CX for outer transmission to circuits via URM Architechture of UGCX The UGCX has its own high speed processor for executing above procedure with a instruction cycle time of 2 microseconds. The instructions are of 4B Bits but only 32 bits are used. The program memory is organized in a 4k word sarcler card. Besides, the processor has a work memory, instruction decoders etc. There is a buffer memory also to store messages required to be exchanged between UGCX and marker.

CONTROL UNITS1.0 dedicated control units carry out various operations related to call processing. Some of these operations are: reception and storageof dialed digits, analysis and digits, control over a transmission of various tones, path finding via the switching network, metering the call etc. there are 6 control units. These are 1. Marker 2. Multiregister 3. Translater 4. Charging unit 5. Standind by charge recording unit 6. Monitoring unit MQ MR TR TX DSF OC

The call processing and control functions are distributed over the 6 control units. These 48 provided in control units. Marker (MQ) The marker acts as a message distributer and routes the message associated with call from one unit to other. It also finds a path through switching. A marker is seized with: Connection units in case of a new call or on hook condition ( i.e release) of a subscriber26

Multiregisters (1) for free/ busy/ check of subs ( URA)/ cct/ ( URM)/ frequency receiver (ETA). (2) for connections as well as release of connections in sw. network Two markers are provided in a E10B system. These two operate in load sharing mode. Each marker occupies a rack which also houses a switching network control unit (UGCX). UGCX controls the operations of switching networkon the basis of messages received from control units like marker multiregister,charging unit etc. 2.1 Marker BLOCK SCHEMATIC The marker is organized around the processor (ELS 4B) and a 16 bit data bus LIM. For exchange of messages with control units and other control units interchange modules are connected to the ELS by S bus lIM. On the interchange modules, links from other modules are terminated. Depending on the linka connected these modules are named as LM module, LC module, LU module etc. For stripping data necessary for marker operations MAMQ is provided,which is connected to ELS over the LIM In addition, a buffer memory known as MICX is also provided. 3.0 MULTIREGISTER (MR) The multiregister is the heart of the system ad performs overall command and control functions for call set up and call release. An exchange is provided with two to six multiregisters according to traffic requirements. These multiregisters operage on load sharing basis. Each MR can handle a setting up or release of 254 calls simultaneously. The multiregister: - accepts seizer requests and on/ off hook conditions from subscribers and circuits detected by connecting units. - stores dialed digits. - orders connection between time slots of calling and called parties in the switching networks. - orders transmission of tones, frequencies and signaling towards subscribers and circuits. - orders the release of calls. 3.1 MULTIREGISTER BLOCK SCHEMATIC The multiregisters has 2 sections - the interchange unit - regisrer unit27

These two sections exchange messages via buffer memory accessible by the ELS of the interchange units by the processor of register units. The buffer is called the I/Q buffer (TES). 3.1.1 INTERCHANGE UNIT ( EXCHANGER)

The interchange unit comprises - Processor ELS 4B which implements the multiregisters interchange functions - A molecule LM exchanges with controls units (TX, MQ, OC, DSF, TX). - A module LC handeling exchanges ( faults, unit status change) with a monitering unit OC - A module RLTX ,which receives a subscriber line and circuit signaling and retransmitsitv to the signaling reception module (RSI) in the register unit 3.1.2 REGISTER UNIT A register unit comprises: - A register processing system which executes various multiregisters call processing programs . - A data memory modle: this is a RAM containing 256 memory areas Designated as registers each containing 64 words of 16 bits each. - A register processing memory module (MTEN) into which register data is transferred for processing by processing system. - A register transferring system which transfers register data between the Data memory and MTEN module - An auxillary register expansion memory module MAMR, a RAM which Increases the capacity of each register by 64 words* 16 bits. - A module LX for transmitting commands to the switching network ( ex. for connecting/ disconnectingtones and signaling to be sent on sub lines and circuits). These commands are sent to the UGCX via LX links. - A moduleRSI which stores subscribers lines and cct signaling by module RLTX OPERATING PRINCIPLE It is a multiregister, any one out of 254 registers is used to process data for Setting up or release of a call data required for setting up the call is stored in a register of 64 words capacity during its seizure( when calling party goes off hook) and is updated in usual time during call set up phase ( ex calling subscriber dialling digits ). These registers are processed cyclically on a time sharing basis. A register processing system at least once in every tocll set up or release is processed a28

number of times by register processing system executing a number of circuit signaling stored in a RSI module is also made use of while processing. During call set up/ release, exchange and depending on the phase of call sends sutaible message for a call set up/ release. 4.0 TRANSLATOR Data relating to the subscribers i.e. category, class of service and other characterstics required by multiregisters during call processing are supplied by translator. Similar data relating to circuits also stored in a translator. These data are organised in a data memory in a translator in the form of lines. Man machine commands from the OMC depending upon the even by subscribers ( eg. Data relating to additional services) there are two translators which work on load sharing basis. 4.1 TRANSLATOR BLOCKING SCHEMATIC The translator is organized around the ELS 4B processor. Interchange modules and data memory are connected to the processor on the common bus. The processor can address any part of data memory depending upon the type of data required 4.1.1 DATA MEMORY This is a auxillary memory which stores all the data needed to set up and release calls. The data memory (MATR) consists of maximum three memory modules. Depending an amount of data to be stored , the translator will be equipped one, two or three modules. Each memory module can store 256 k words ( 1 word = 16 bits) of data. The data in the memory is organized in the form of files In a typical E- 10B exchange, the data memory consists of a maximum of 523 Files. 4.2 OPERATING PRINCIPL The translator is called by multiregister during call set up and its release.there are two phases of call set up when translator is called up by multiregisters i.e. preselectionand selection. 4.2.1 Preselection The translator provides the multiregister with a class of service of calling subscriber. Calling party may be a subscriber or a incoming circuit. The nature of a output data depends upon the category and a type of a calling source: -subscriber: entitement of dial tone, type of telephone instrument whether STD barred subscriber etc. -incoming circuit: characterstics bf the circuit group( eg. Type of signalling used, total number of circuits etc.)29

Preselection process for a call from a local subscriber: Data is obtained from translator data memory in response to the class of service requests message from multiregister. 4.2.2 Selection when adequate no. of digits has been received by th multiregister, it has to find the routing and charging information to route the call properly. This information is obtained by translatorfrom its files. As soon as MR receives 2/3 digits, it calls TR to determine the type of call i.e local/ regional, national, special services or international.The determination of call is called preanalysis supply selection data such as selection address( eg. UR no. and eqpt of the URA of the called sub.) charging data. characterstics of the circuit group to be used ( in the case of O/G calls). Any selection ( or translation) procedure in the translator may be represented as follows. 5.0 CHARGING UNIT (TX) There are two charging units which operate in load sharing mode, both the char ging units update each other for certain jobs. A charging unit can handle charging of 2000 calls simultaneously. It also stores charge account of all subscribers,transmits a detailed billing data for the subscribers entitled for the facility and transmits traffic measurements results to operation and maintenance centre (OMC) via monitering unit (OC). 5.1 CHARGING UNIT BLOCK SCHEMATIC

The charging unit is organised in the same way as multiregister. There are three Functional subsystems: Interchange unit Register unit I/O interface

5.1.1 The unit is organized around an ELS 4B processor. There are interchange modules for various links. It includes a clock module (HORE) which provides a day, hour minutes and seconds needed by ELS 4B processor to manage a charge rate table as appropriate to the type of day and time. This information is also useful in detailed data; it as also provided to translator for changing the route of timedependent special services( eg. Calls to 198 can be routed to local Xge during30

working hours and to a centralized location during slack hours). The interchange unitalso contains a Auxillary charging memory (MATX). This memory contains Subscriber charge account Charge rate tables Traffic ibservation data Extention registers

This memory is organized into files. 5.1.2 Register unit This unit is analogous to register unit of MR. About 2000 charging registers are Provided in the data memory i.e. 2000 calls can be charged simultaneously. 5.1.3 The I/O interface comprises The I/O buffers for interchange of messages between interchange unit and Register unit The currently applicable charge rate table. 5.2 OPERATING PRINCIPLE The steps in the charging process are as follows :

When calling party lifts the hand set, a charging unit is seized by mul tiregister. This charging unit also calculate the charge and update charge account in the other one at the end of the call. As soon as calling subscriber goes off , multiregister sends a message to start calculation of charge to the charging unit already seized in this message all the information required by charging unit for the charge computation is sent. MR has already received this information during selection( translation) phase from TR. In TX the charging register compute the charge, in case of periodic call, charge is calculated from the entire duration of call, using the data avai lable in currently used charge rate table. At the end of the call, charge account for calling party is incremented in Both the charging units When metering pulses are received from the distant exchange ( eg. TAX), the seized charging register receive the metering pulses and keeps on counting the call the charging register counting value ie the total no. of metering units is transferred to ( MATX). The subs account is thus incremented; subscriber account in the other charging unit is also updated



DSF does not call processing. It is not duplicated. It is provided with magneticTape drive unit and also includes the interchange unit around processor ELS 4B. The DSF has 2 main functions - Data save - Regeneration Data save - this take place in following contingencies: - Exchange - OMC link failure - Monitoring unit in exchange block - Mag tape drive unit in OMC faulty Data like detailed billing and traffic observations will be stored in DSF mag tape In normal course. This data is stored in magnetic tapes mounted on OMC. After restoration of normal conditions, the data stored on DSF magnetic tape command it is to be noted that the DSF does not directly store the data account. In other words it is not standby for charging units. This name is a misnomer. REGENERATION Auxillary memories (RAM) of marker, translatorand charging unit can be regenera ted is reloaded from DSF for this perpose. 7.0 MONITORING UNIT OC

It is a interface between exchange and OMC. It controls the transfer of message between the OMC and various units of exchange. The no. of traffic counters are located in OC 8.0 MISCELLANEOUS UNITS The following miscellaneous units are also available in a E10B exchange. These are not directly involved in call processing functions of an exchange - Time base (BT) unit - General visual display panel (PGV). 8.1 Power Distribution column (MDE). TIME BASE (BT) Each exchange has its own time base, which is not duplicated. The time base gen erates the basic signals ( 2w, 20, h, ti). The timing signals are sent to other exchange units where secondry. Timing signals are required for synchronizing and co-ordinating various operations of E-10B exchange units.32

For security reasons the signal generation system generating the basic timing signals is triplicated 6.144 MHz oscillator is used for signal generation. Provision exists in time base for fault detection & simulation. 8.2 General Visual Display Panel(PGV):

The general visual display panel(PGV) displays simultaneously all alarms relating to an exchange to an exchange together with OMC generated alarms.There is a PGV in each E-10B exchange and it is driven by an auxiliary equipment rack(BDA). When the OMC is informatics to any alarm condition occurring in the exchange of controls the visual indication on the PGV via BDA. 8.3 Power Distribution Column(MDE) :

The power distribution columns(MDE) are end of each suite. The MDE distributes racks of its suite and provides voltage using fuses and circuit breadkers.

Functions of OMC:

Subscriber management Trunk group management Routing and Analysis management Call charging management Semipermanent connection management Processing of permanent line(PG) lock out conditions Subscriber line and telephone set testing Trunk testing Traffic and load measurements Fault message processing Alarm message processing Unit positioning Fault tracing Fault clearing



Stages In A Local Call There are three stages in the setting up of a local call. These are:1. Preselelection 2. Selection 3. Call connection and charging 1. Pre-Selection: There are four operations in the preselection stage, these are: a) Detection of calling subscribers off-hook condition by scanningb) Getting the class of service of the calling party

c) Initiation of traffic observation d) Feeding of dial tone 2. Selection: Selection is the second main stage in the local call set upand commences with subs dialing. This stage consists of five phases of operation. These are: a) Digit reception b) Preliminary digit analysis c) Digit analysis/Translation d) Registration of the type of call by the charging unit e) Called partys loop status testing 3. Connection and charging This is the third main stage in the local call set up and consists of the following operations: a) ) Sending of ringing current to the called party and ring back tone to the calling party b) Called subscriber answer (orr-hook). c) Termination of ringing current and ring back tone.34

d) Connection of calling party and called party time-slots. e) Charging in TX. f) Release of telephone call register.

Pulse-code modulationPulse-code modulation (PCM) is a digital representation of an analog signal where the magnitude of the signal is sampled regularity at uniform intervals, then quantized to a series of symbols in numeric (usually binary) code. PCM has been used in digital telephone systems and 1980s-era electronic musical keyboards. It is also the standard from for digital audio in computers and the compact disc red book format. It is also standard in digital video, for example, using ITU-R BT.601. However, straight PCM is not typically used for video in standard definition consumer applications such as DVD or DVR because the bit rate required is far too high.

ModulationIn the diagram, a sine wave (red curve) is Sampled and quantized for PCM. The sine wave is sampled at regular intervals, shown as ticks on the x-axis. For each sample, one of the available values (ticks on the y-axis) is chosen by some algorithm (in this case, the floor function is used). This produces a fully discrete representation of the input signal (shaded area) that can be easily encoded as digital data for storage or manipulation. For the sine wave example at right, we can verify that the quantized values at the sampling moments are 7, 9, 11, 12, 13, 14, 15, 15, 15, 14, etc.35

Encoding these values as binary numbers would result in the following set of nibbles: 0111, 1001, 1011, 1100, 1101, 1110, 1111, 1111, 1111, 1110, etc.

These digital values could then be further processed or analyzed by a purposespecific digital signal processor or general purpose CPU. Several pules code modulation streams could also be multiplexed into a large aggregate data stream, generally for transmission of multiple streams over a single physical link. This technique is called time-division multiplexing, or TDM, and is widely used, notably in the modern public telephone system. There are many ways to implement a real device that performs this task. In real system, such a device is commonly implement on a single integrated circuit that lacks only the clock necessary for sampling, and is generally referred to as an ADC (Analog-to-digital converter). These devices will produce on their output a binary representation of the input whenever they are triggered by a clock signal, which would then be read by a processor of some sort.

DemodulationTo produce output from the sampled data, the procedure of modulation is applied in applied in reverse. After each sampling period has passed, the next value is read and the output of the system is shifted instantaneously (in an idealized system) to the new value. As a result of these instantaneous transitions, the discrete signal will have a significant amount of inherent high frequency energy, mostly harmonics, the signal would be passed through analog filters that suppress artifacts outside the expected frequency (see square wave). To smooth out the signal and remove these undesirable harmonics, the signal would be passed through analog filters that suppress artifacts outside the expected frequency range (i.e. greater than, the maximum resolvable frequency). Some system use digital filtering to remove the lowest and largest harmonics. In some system, no explicit filtering is done at all; as its impossible for any system to reproduce a signal with infinite bandwidth, inherent losses in the system compensate for the artifacts or the system simply does not require much precision. The sampling theorem suggests that36

practical PCM devices, provided a sampling frequency that is sufficiently greater than that input signal, can operate without introducing significant distortions within their designed frequency bands. The electronics involved in producing an accurate analog signal from the discrete data are similar to those used for generating the digital signal. These devices are DACs (digital-to-analog converters), and operate similarly to ADCs. They produce on their output a voltage or current (depending on type) that represents the value presented on their inputs. This output would then generally be filtered and amplified for use.

LimitationsThere are two sources of impairment in PCM system: Choosing a discrete value near the analog signal for each sample (quantization error) The quantization error swing between q/2 to q/2 so mean = integration ( xf(X) dx ) the int from q/2 to +q/2 which equal zero variance = int (x-mean)^2 f(x) dx the int from q/2 to +q/2 which equal to q^2/12. Between samples no measurement of the signal is made; due to the sampling

theorem this results in any frequency above or equal to (fs being the sampling frequency) being distorted or lost completely (aliasing error). This is also called the Nyquist frequrency. As samples are dependent on time, an accurate clock is required for accurate reproduction. If either the encoding or depending clock is not stable. Its frequency drift will directly affect the output quality of the device. A slight difference between the encoding and decoding clock frequencies is not generally a major concern; a small constant error is not noticeable. Clock error does become a major issue if the clock is not stable, however. A drifting clock, even with a relatively small error, will cause very obvious distortions in audio and video signals, for example. Digitization as part of the PCM process In conventional PCM, the analog signal may be processed (e.g. by amplitude compression) before being digitized .Once the signal is digitized, the PCM signal is subjected to further processing (e.g. digital data compression). Some forms of PCM combine signal processing with coding. Older version of these system applied the processing in the analog domain as part of the A/D process, newer implementation do so in the digital domain. These simple techniques have been largely rendered obsolete by modern transform-based audio compression techniques.37

Differential (or Delta) pulse-code modulation (DPCM) encodes the PCM values as differences between the current and the predicted value. An algorithm predicts the next sample based on the previous samples, and the encoder stores only the difference between this prediction and the actual value. If the prediction is reasonable, fewer bits can be used to represent the same information. For audio, this type of encoding reduces the number of bits required per sample by about 25% compared to PCM. Adaptive DPCM (ADPCM) is a variant of DPCM that varies the size of the quantization step, to allow further reduction of the required bandwidth for a given signal-to-noise ratio. Delta modulation, another variant, uses one bit per sample. In telephony, a standard audio signal for a single phone call is encoded as 8000 analog samples per second, of * bits each, giving a 64 kbit/s digital signal known as DS0. The default signal compression encoding open a DS0 is either u-law (mu-law) PCM (North America and Japan) or a-law PCM (Europe and most of the rest of the world). These are logarithmic compression system where a12 or 13 bit linear PCM sample number is mapped into an * bit value. This system is described by international standard G.711. An alternative proposal for a floating point representation, with % bit mantissa and 3 bit radix, was abandoned. Where circuit costs are high and loss of voice quality is acceptable, it sometimes makes sense to compress the voice signal even further. An ADPCM algorithm is used to map a series of 8 bit -law (or a-law) PCM samples into a series of 4 bit ADPCM samples. In this way, the capacity of the line is doubled. The technique is detailed in the G.726 standard. Later it was found that even further compression was additional standards were published. Some of these international standards describe systems and ideas which are covered by privately owned patents and thus use of these standards requires payments to the patent holders. Some ADPCM techniques are used in Voice over IP communications. Encoding for transmission Pulse-code modulation can be either return-to-zero (RZ) or non-return-to-zero (NRZ). For a NRZ system to be synchronized using in-band information, there must not be long sequences of ones-density before modulation into the channel. In other cases, extra framing bits are added into the stream which guarantee at least occasional symbol transitions. Another technique used to control ones-density is the use of a scrambler polynomial on the raw data which will tend to turn the raw data stream into a stream that looks pseudo-random, but where the raw stream can be recovered exactly by reversing38

the effect of the polynomial. In this case, long runs of zeroes or ones are still possible on the output, but are considered unlikely enough to be within normal engineering tolerance. In other cases, the long term DC value of the modulated signal is important, as building up a DC offset will tend to bias detector circuits out of their operating range. In this case special measures are taken to keep a count of the cumulative DC offset, and to modify the codes if necessary to make the DC offset always tend back to zero. Many of these codes are bipolar codes, where the pulses can be positive, negative or absent. In the typical alternate mark inversion code, non-zero pulses alternate between being positive and negative. These rules may be violated to generate special symbols used for framing or other special purposes.

Sampling (signal processing)In signal processing, sampling is the reduction of a continuous signal to a discrete signal.A common example is the conversion of a sound wave (a continuous-time signal) to a sequences of samples (a discrete-time signal). A sample refers to a value or set of values at a point in time and/or space. A sampler is a subsystem or operator that extracts samples from continuous signal.A theoretical ideal sampler multiplies a continuous signal with a Dirac comb.This multiplication picks out values but the result is still continuous-valued. If thissignal is then discretized ( i.e., converted into a sequence) and quantized along all dimensions it becomes a discrete signal Theory For convenience, we will discuss signals which vary with time. However, the same results can be applied to signals varying in space or in any other dimension. Let x(t) be a continuous signal which is to be sampled ,and that sampling is performed by measuring the value of the continuous signal every T seconds. Thus, the sampled signal x[n] given by x[n]=x(nT),with n=0,1,2,, The sampling frequency or sampling rate fs is defined as the number of samples obtained in one second, or fs=1/T. The sampling rate is measured in hertz or in samples per second. We can now ask: under what circumstances is it possible to reconstruct the original signal completely and exactly (perfect reconstruction)?


A partial answer is provided by the Nyquist-Shannon sampling theorem, which provides a sufficient (but not always necessary) condition under which perfect reconstruction is possible. The sampling theorem guarantees that band limited signals (i.e., signals which have a maximum frequency) can be reconstructed perfectly from their sampled version, if The sampling rate is more than twice the maximum frequency. Reconstruction in this case can be achieved using the Whittaker- Shannon interpolation formula. The frequency equal to one-half of the sampling rate is therefore a bound on the highest frequency that can be unambiguously represented by the sampled signal. This frequency (half the sampling rate) is called the Nyquist frequency of the sampling system. Frequencies above the Nquist frequency fn can be observed in the sampled signal, but their frequency is ambiguous. That is, a frequency component with frequency f cannot be distinguished from other components with frequencies NfN+f and NfN-f for nonzero integers N. This ambiguity is called aliasing. To handle this problem as gracefully as possible, most analog signals are filtered with an antialiasing filter (usually a low-pass filter with cutoff near the Nyquist frequency) before conversion to the sampled discrete representation. A more general statement of the Nyquist-Shannon sampling theorem says more or less that the signals with frequencies higher than the Nyquist frequency can be sampled without loss of information provided their bandwidth (non-zero frequency band) is small enough to avoid ambiguity, and the band limits are known. Sampling interval The sampling interval is the interval T=1/fs corresponding to the frequency. Observation period The observation period is the span of time during which a series of data samples are collected at regular intervals. More broadly, it can refer to any specific period during which a set of data points is gathered, regardless of whether or not the data is periodic in nature. Thus a researcher might study the incidence of earthquakes and tsunamis over a particular time period, such as a year or a century. The observation period is simply the span of time during which the data is studied, regardless of whether data so gathered represents a set of discrete events having arbitrary timing within the interval, or whether the samples are explicitly bound to specified sub-intervals. Practical implication In practice, the continuous signal is sampled using an analog-to-digital converter (ADC), a non-ideal device with various physical limitations. This result in deviations40

from the theoretically perfect reconstruction capabilities collectively referred as distortion. Various types of distortion can occur, including: Aliasing: A precondition of the sampling theorem is that the signal be band limited. However, in practice, no-limited signal can be band limited. Since signals of interest are almost always time-limited (e.g., at most spanning the lifetime of the sampling device in question), it follows that they are not band limited. However, by designing a sampler with an appropriate guard band, it is possible to obtain output that is as accurate as necessary. Integration effect or aperture effect: This results from the fact that the sample is obtained as a time average within a sampling region, rather than just being equal to the signal value at the sampling instant. The integration effect is readily noticeable in photography when the exposure is too long and creates a blur in the image. An ideal camera would have an exposure time of zero. In a capacitor-based sample and hold circuit, the integration effect is introduced because the capacitor cannot instantly change voltage thus requiring the sample to have non-zero width. Jitter or deviation from the precise sample timing intervals. Noise, including thermal sensor noise, analog circuit noise, etc. Slew rate limit error, caused by an inability for an ADC output value to change sufficiently rapidly. Quantization as a consequence of the finite precision of words that represent the converted values. Error due to other non-linear effects of the mapping of input voltage to converted output value (in addition to the effects of quantization). The conventional, practical digital-to analog converter (DAC) does not output a sequence of dirac impulses (such that, if ideally low-pass filtered, result in the original signal before sampling) but instead output a sequence of piecewise constant values or rectangular pulses. This means that there is an inherent effect of the zeroorder hold on the effective frequency response of the DAC resulting in a mild roll-off of gain at the higher frequencies (a 3.9224 db loss at the Nyquist frequency).This zero- order hold effect is a consequence of the hold action of the DAC and is not due to the sample and hold that might precede a conventional ADC as is often misunderstood. The DAC can also suffer errors from jitter noise, slewing and nonlinear mapping of input value to output voltage. Jitter, noise and quantization are often analyzed by modeling them as random errors added to the sample values. Integration and zero-order hold effects can be analyzed as a form of low-pass filtering. The non-linear ties of either ADC or DAC are41

analyzed by replacing the ideal linear function mapping with a proposed nonlinear function.

Quantization (signal processing)In digital signal processing, quantization is the process of approximating a continuous range of values (or a very large set of possible discrete values) by a relatively-small set of discrete symbols or integer values .More specifically, a signal can be multi-dimensional and quantization need not be applied to all dimensions. Discrete signals (a common mathematical model) need not be quantized, which can be a point of confusion. See ideal sampler. A common use of quantization is in the conversion of a discrete signal (a sampled continuous signal) into a digital signal by quantizing. Both of these steps (sampling and quantizing) are performed in analog-to-digital converters with the quantization level specified in bits. A specific example would be compact disc (CD) audio which is sampled at 44,100 Hz and quantized with 16 bits (2 bytes) which can be one of 65,536 (i.e. 216 ) possible values per sample. In electronics, adaptive quantization is a quantization process that varies the step size based on the changes of the input signal, as a means of efficient compression. Two approaches commonly used are forward adaptive quantization and adaptive quantization. Mathematical description The simplest and best-known form of quantization is referred to as scalar quantization, since it operates on scalar (as opposed to multi-dimensional vector) input data. In general, a scalar quantization operator can be represented as Q()=g(f()) Where is a real number to be quantized, . is the floor function, yielding an integer result i= f() that is sometimes referred to as the quantization index, f() and g(i) are arbitrary real-valued functions. The integer-valued quantization index i is the representation that is typically stored or transmitted, and then the final interpretation is constructed using g(i) when the data is later interpreted.


In computer audio and most other applications, a method known as uniform quantization is the most common. There are two common variations of uniform quantization, called mid-rise and mid-tread uniform quantizes. If is a real-valued number between -1 and 1, a mid-rise uniform quantization operator that uses M bits of precision to represent each quantization index can be expressed as Q()=2 m-1 +0.5/2 m-1 In this case the f(x) and g (i) operators are just multiplying scale factors (one multiplier being the inverse of the other) along with an offset in g(i) function to place the representation value in the middle of the input region for each quantization index. The value 2 (m-1) is often referred to as the quantization step size. Using this quantization law and assuming that quantization noise is approximately uniformly distributed over the quantization step size (an assumption typically accurate for rapidly varying x or high M) and further assuming that the input signal x to be quantized is approximately uniformly distributed over the entire interval from -1 to 1 the signal to noise ratio (SNR) of the quantization can be computed as S/Nq=20 log 10 (2m) = 6.0206m db From this equation, it is often that the SNR is approximately 6 db per bit. For mid-tread uniform quantization, the offset of 0.5 would be added within the floor function instead of outside of it. Sometimes, mid-rise quantization is used without adding the offset of 0.5. This reduces the signal to noise ratio by approximately 6.02 db, but may be acceptable for the sake of simplicity when the step size is small. In digital telephony, two popular quantization schemes are the A-law (dominant in Europe) and j-law(dominant in North America and Japan) .These schemes map discrete analog values to an 8-bit scale that is nearly linear for small values and then increases logarithmically as amplitude grows. Because the human ears perception of loudness is roughly logarithmic, this provides a higher signal to noise ratio over the range of audible sound intensities for a given number of bits


ConclusionDuring the training at E10B exchange, the whole procedure of call connecting is well understood. The fact that to connect a call between to end is a tedious task and it require lot of be performed like first the transmission of analog signal from source to destination, conversion of analog signal to digital signal, transmission the signal to the required department. E10B acts as the local exchange and switching according to time division multiplexing to connect the call to destination. Lot of precaution are taken in order to provide smooth, gapless voice transfer between source and destination