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

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

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.99,390
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

BSNL 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.

• 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


Alcatel 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 current
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-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 business
Alcatel 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(Triple
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 BSNL

There are various types of digital switching systems used in BSNL:




Evolution and architecture of E-10B-System

1.0. Introduction
The 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 digital

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

3.0. 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 80’s 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 TAX

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 subscribers

E-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



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 ‘RLU’s. 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

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.

6.0 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.

7.0. Basic principles and architecture of E-10B system:

7.1. Basic features

The system is based on the following salient features:

1. Stores program control (SPC)

2. TDM digital switching.
3. PCM principles and techniques.
4. Segregation of switching and management functions.
5. 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

7.1.1. 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:

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-Space-
Time 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


Summary 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 functions


• 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)



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.


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) URA- Subscriber Connection Unit. This is also known as

CSE-Electronic Space Division Concentrator.
(2) URM- Multiplex Connection Unit.
(3) ETA- Frequency Sender/Receiver Unit.
(4) BDA- Auxiliary Equipment Rack.

2.0 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)

subscriber 4 PCM LINKS

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



2.1 The major functions of CSE are:

i. Power feeding for subscriber lines.

ii. Detection of line event or line loop status (off hook/on-hook)

iii. 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


v. Search for a free time-slot and connection or disconnection of a

subscriber’s equipment to/from a time-slot.

vi. Analogue to Digital conversion.

vii. Routine Subscriber line tests under the control or Operation and
Maintenance Center (OMC)

2.2 Subsystems of CSE : The CSE can be devided into two subsystems-

1. The speech path subsystem

2. The control logic subsystem.

2.2.1 The speech path subsystem

1. Singalling equipment

2. Concentration network

3. Transmission equipment. Singalling equipment

The Singalling equipment is the Subscriber line card called XEJ. The purpose
of this unit is

(1) to feed power to telephone sets.

(2) Detect line status (off-hook/on-hook)
(3) To send the ringing current
(4) To subject the line or equipment to various tests.
(5) 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.
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 CSE





The transmission equipment performs filtering, sampling and encoding


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.
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

2.3 CSED (URAD):-The CSED is parented to Main E-10B Exchange using 2 to 4


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.

2.5 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.)

3.0 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
3.1 The multiplex Connection Unit is the PCM connection interface between E-
10B 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) MRM (Multiplex Connection module)

(2) MRS (Satellite connection module)

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

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.

- Under the control of the other logic.

- Periodically every 24 hours.

- Under instruction from OMC (by man-machine command)

3.2 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.

3.3 Frequency sender and Receiver Unit (ETA)

The functions of ETA are:

1. Generation of tones and R2, 2 out of 5 MF Frequencies

2. 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

- 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

i. Detection of all alarm conditions and forwarding them to OMC.

ii. 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.

4.0 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.

5.0 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
URA’s 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).



1.0 CX Environment

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.

1.2 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

i. Subscriber - Subscriber (Local call)

ii. Subscriber - Circuit or junction (O/G call)

iii. Circuit or junction-subscriber (terminating call)

iv. Circuit or junction- Circuit or Junction (Transit call)

Any such connection involves two, one-way (unidirectional) connections as


Calling party’s timeslot on the LRE- Calling party’s 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 CX

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

11040 X 80

No. of simultaneous calls = 2 100

= 5520 x 0.8

= 4416

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

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 associated

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 ETA

5.0 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

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.

1.0 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 MQ

2. Multiregister MR

3. Translater TR

4. Charging unit TX

5. Standind by charge recording unit DSF

6. Monitoring unit 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

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.


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.


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.


The multiregisters has 2 sections

- the interchange unit
- regisrer unit

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).


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
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



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 a
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.


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


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


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.


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.)
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

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
Any selection ( or translation) procedure in the translator may be represented as


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).


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 time-
dependent special services( eg. Calls to ‘198’ can be routed to local Xge during

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.


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

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.


Auxillary memories (RAM) of marker, translatorand charging unit can be

regenera ted is reloaded from DSF for this perpose.


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


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).

- Power Distribution column (MDE).


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.
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 subscriber’s off-hook condition by scanning

b) 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 party’s 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.

d) Connection of calling party and called party time-slots.

e) Charging in TX.

f) Release of telephone call register.

Pulse-code modulation
Pulse-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.

In 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.

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 purpose-

specific 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.

To 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 it’s 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 that
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.

There 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

• 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 µ-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 reversing
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

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


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 anti-
aliasing 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 deviations

from the theoretically perfect reconstruction capabilities collectively referred as

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 zero-
order 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 non-
linear 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 are

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

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

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



χ is a real number to be quantized,

└.┘ is the floor function, yielding an integer result i= └f(χ) that is sometimes
referred to as the quantization index,

f(χ) 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 χ 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(χ)=└2 m-1 χ┘+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 ear’s 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

During 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