DECLARATION

The Student, ……….…….…………………………. with ID Number of

………………… at the Department of Environmental Engineering in the Faculty of

Engineering, Who Performed …………. Workdays Industrial Training in the Academic

Year of ……-……, and the Aforementioned Training Has Been Accepted as ……………

Workdays.

The above information is true as far as to my knowledge and concern. In case any

fault or error is found in the above said statement I shall be held responsible for the action

arising out of that.

Sandeep Kumar

B.Tech 4th year (ECE)

Poornima Institute Of Engg. And Technology

CHAPTER- 1

INTRODUCTION

1.1 Introduction:

NTPC, India's largest power company, was set up in 1975 to accelerate power

development in India. It is emerging as an ‘Integrated Power Major’, with a significant

presence in the entire value chain of power generation business.NTPC ranked 317th in the

‘2009, Forbes Global 2000’ ranking of the World’s biggest companies. With a current

generating capacity of 32,194 MW, NTPC has embarked on plans to become a 75,000 MW

company by 2017.

Vision: "A world class integrated power major, powering India’s growth, with

increasing global presence."

Mission : “Develop and provide reliable power, related products and services at

competitive prices, integrating multiple energy sources with innovative and eco-friendly

technologies and contribute to society.”

1.2 Background of the company:

NTPC kahalgaon (a venture of NTPC limited )located in Bhagalpur district in Bihar

near the banks of river Ganges. It is basically a super thermal power plant generating

electricity from coal. The plant has nearly 4000 employees from different parts of the

country generally from West Bengal, Jharkhand, Orissa etc. It was established way back in

the year 1990 and from then it has made considerable amount of progress and this

continues till today. The company has been working to serve to the society with the most

basic and essential need that is “electricity”. Moreover it also supports many small scale

and large scale industries in the region of Bihar and other neighboring provinces like West

Bengal, Bihar,Jharkhand, Orissa, Sikkim. Today this branch of NTPC ltd produces around

2300 MW of electricity. (210 X 4 and 500 X 3).

1.3 Organizational Structure:

The company is a government enterprise and comes under the public sector . It is

under the direct control of the Government Of India. The company has its own board of

members and has its headquarters at New Delhi.(NTPC bhawan, Scope complex , 7

institutional area, Lodhi road New delhi-110003). Basically a thermal plant there are three

major companies that help for the the establishment of the thermal plant namely B.H.E.L

(bharat heavy electrical limited) , N.T.P.C and Power grid corporation. The work done by

the companies are listed below.

B.H.E.L – It works in the establishment of a power plant i.e it provides the basic material

and all the machinery required for the establishment of a power plant. The machinery may

include items ranging from turbines to generators and several other essential parts.

N.T.P.C – It works for the smooth running of the power plant i.e production of electricity.

Morever it also works for the maintainence of the power plant and looks after the glitches

that occur in the plant.

Power Grid – The work of N.T.P.C is only to produce electricity but the work of

distribution of the produced electricity is done by Power Grid. The power Grid decides how

much electricity is to be send to which place.

The company has a advisory board which looks after the working of the power plants in an

efficient manner and to solve problems regarding any issues in the company.

ADVISORY BOARD:

Chairman and Managing Director: Mr. R.S.Sharma Director (Finance): Shri A.K.Singhal Director (Commercial): Shri I.J.Kapoor Director (Technical): Shri D.K.Jain Director (Projects): Shri B.P.Singh Chief vigilance officer : Shri T. venkatesh

1.4 Nature of Business:

The company basically deals with the production of electricity from water. Presently,

NTPC generates power from Coal and Gas. With an installed capacity of 32,194 MW, NTPC

is the largest power generating major in the country. It has also diversified into hydro

power, coal mining, power equipment manufacturing, oil & gas exploration, power trading

& distribution. With an increasing presence in the power value chain, NTPC is well on its

way to becoming an “Integrated Power Major.”

The water is first heated to steam and then this steam is used for rotation of

turbines which in turn produces electricity. For this it requires huge amount of coal and

water. The water is taken from the river ganges and coal is taken from Rajmahal Coal

Fields of ECL. in Jharkhand.

1.5 Products:

The product is generally the generated electricity which is supplied to the neighbouring

states.

1.6 Market strength:

NTPC has been ranked 317th rank in the world among the biggest companies. This

has for sure increased the confidence of the investors and shareholders.

Stock quote: 197.00

 

NTPC has been operating its plants at high efficiency levels. Although the company hass

18.79%of the total national capacity it contributes 28.60% of the total power generation due to its focus

on high efficiency.

In October 2004, NTPC launched its Initial Public Offering (IPO) consisting of 5.25% as offer

for sale by Government of India. NTPC thus became a listed company in November 2004 with thee

Government holding 89.5%of the equity share capital. The rest is held by the Institutional investors and

the Public. NTPC is among the largest 5 companies in India in terms of market capitalization.

CHAPTER- 2

COMPANY INFRASTRUCTURE:

2.1 Introduction:

NTPC is one of the leading power generating coprporation in india. The total

installed capacity of the company is 30,144 MW (including JVs) with 15 coal based and 7

gas based stations, located across the country. In addition under JVs, 3 stations are coal

based and another station uses naptha/LNG as fuel. By 2017, the power generation

portfolio is expected to have a diversified fuel mix with coal based capacity of around

53000 MW, 10000 MW through gas, 9000 MW through hydro generation, about 2000 MW

from nuclear sources and around 1000 MW from Renewable Energy Sources (RES). NTPC

has adopted a multi-pronged growth strategy which includes capacity addition through

green field projects, expansion of existing stations, joint ventures, subsidiaries and takeover

of stations.

NTPC has been operating its plants at high efficiency levels. Although the company

has 18.79%of the total national capacity it contributes 28.60% of the total power generation

due to its focus on high efficiency.

2.2 Department Structure:

For the smooth running of the company the overall load is distributed among

various departments. These departments serve as the building blocks which works

individually. They can be broadly classified into the following departments depending upon

the nature of work they do.

i) Finance department

ii) Surveillance department

iii) Project department

iv) Field work department

a) Information technology and communication

b) Control and instrumentation

c) Process management

d) Material management

e) Raw material handling

f) Heavy vehicles maintenance department

v) Employee development department and

vi) H.R department

2.3 Network structure

NTPC Electric Supply Company Ltd. (NESCL)

The company was formed on August 21, 2002. It is a wholly owned subsidiary

company of NTPC with the objective of making a foray into the business of distribution

and supply of electrical energy, as a sequel to reforms initiated in the power sector.

NTPC Vidyut Vyapar Nigam Ltd. (NVVN)

The company was formed on November 1, 2002, as a wholly owned subsidiary company of

NTPC. The company’s objective is to undertake sale and purchase of electric power, to

effectively utilise installed capacity and thus enable reduction in the cost of power. NVVN

NTPC Hydro Ltd. (NHL)

The company was formed on December 12, 2002, as a wholly owned subsidiary company

of NTPC with an objective to develop small and medium hydroelectric power projects of

up to 250 MW. More>>

Pipavav Power Development Co. Ltd. (PPDCL)

A memorandum of understanding was signed between NTPC, Gujarat Power Corporation

Limited (GPCL) and Gujarat Electricity Board (GEB) in 2004 for development of a 1000

MW thermal power project at Pipavav in Gujarat by forming a new joint venture company

between NTPC and GPCL with 50:50 equity participation. Pursuant to the decision of

Gujarat Government, NTPC Ltd. has dissociated itself from this company. PPDCL is under

winding up.

2.4 Software and hardware:

Although the company needs a variety of hardware and software for its smooth

running. So it’s need is catered by some companies which offers a complete set of packages

required. These includes software as well as hardware. The software package is provided

by ABB. The hardware package is provided by IBM.

In addition to this in communication network the whole hardware installation has

been done by the company CORAL. This includes installation of EPABX, IRIS.

The control and instrumentation unit is also controlled with the help of software.

This includes automation of starting and stopping of different equipments such as motors ,

valves etc. this has no doubt decreased human interference and above all risk of any

accident.

2.5 Available Plans:

The company is working hard to achieve greater heights and to earn mare respect

among the companies. For this it has undertaken several projects . Moreover it has several

further plans to increase its strength in the field of power generation. It has also plans for

nuclear power plants.

The Unit # 6 of 490 MW of National Capital Thermal Power Project, Dadri was

synchronized with coal on 16.07.2010. This was the second unit planned to provide

power to Commonwealth Games to be organized in Delhi in October 2010. It is

expected to achieve full load shortly.

NTPC and NPTI to run ITI at Barkagaon in Jharkhand.

PMI will be conducting a training programme on "Advance Course in Power

Station Maintenance Operation and Efficiency" from 11/01/2010 to 03/04/2010.

Programme is meant for working Engineers from Central and State utilities, private

sector companies, funding agencies and other stakeholders.

NTPC, NHPC, Power Grid and DVC signed a Joint Venture Agreement (JVA) for

incorporation of a JV Company to set up an Online High Power Indigenous Test

Laboratory for short circuit test facility in the country to ensure testing of the

electrical equipments as per international standards in New Delhi.

NTPC Coal Stations recorded the highest generation of 588.90 MU, a PLF

102.685% on 31st March, '09. The previous highest generation was 587.99 MU,

with PLF of 102.53% on 26th March '09.

CHAPTER- 3

3.1 Introduction:

The training at NTPC was divided into two sections namely

1. Information technology and communication and

2. Control and instrumentation.

Both department concurrently work for the company. The seamless connectivity and high

speed data transfer is managed by the communication system whereas the control of

various equipments and other processes is controlled by the c&I unit.

3.2 Project description

What the plant does basically is described as follows.

Coal from the coal wagons is unloaded in the coal handling plant. This coal is transported

upto the raw coal bunkers with the help of belt conveyers. Coal is transported to bowl mills

by coal feeders. The coal is pulverized in the bowl mills, where it is grounded to a

powdered form. The mill consists of a round metallic table on which coal particles fall.

This table is rotated with the help of motor. There are three large steel rollers which are

spaced 120 degrees apart.

The crushed coal is taken away to the furnace through coal pipes with the help of

hot and cold air mixture from P.A fan. P.A fan takes atmospheric air, a part of which is sent

to air preheaters for while a part goes directly to the mill for temperature control.

Atmospheric air from F.D fan is heated an the air heaters and sent to the furnace as

combustion air.

Water from boiler feed pump passes through economizer and reaches the boiler

drum. Water from the drum passes through down commers and goes to bottom ring header.

Water from bottom ring header is divided to all the foue sides of the furnace. Due to heat

and density difference the water rises up in the water wall tubes.

Water is partly converted to steam as it rises up in the furnace. The steam and water

mixture is again taken to the boiler drum where the steam is separated from water. Water

follows the same path while the steam is sent to superheaters for superheating. The super

heaters are located inside the furnace and the steam is superheated and finally it goes to

turbine. Flue gases from the furnace are extracted by induced draft fan which maintains

balance draft in the furnace with F.D. fan. This flue gases emits their heat energy to various

super heaters in the plant house and finally passes through the air preheaters and goes to

ESP where the ash particles are extracted. ESP consists of metal plates which are

electrically charged. Ash particles are attracted on this plate so that they do not pass

through the chimney to pollute the atmosphere .

The steam generated is made to pass through a three stage turbine. The steam is first

made to pass through a High Pressure Turbine after which it is passes through a reheater to

regain the lost temperature. The steam then passes through a Intermediate pressure turbine

and a Low Pressure Turbine which are just used for boosting purpose. The blades of turbine

rotate at a speed of 3000 rpm and thus rotate the coils in the generator. Based on Faraday’s

law of Electromagnetic Induction a change in magnetic field induces e.m.f and thus

electricity is generated.

The steam from the turbine is cooled down in a condenser. Cold water from river is

circulated through tubes and as the steam from the turbine passes round them it is rapidly

condensed into water condensate. From the condenser the condensate is pumped through

LP heaters by the extraction pump. it is passed through further feed heaters to the

economizer and the boiler for the reconversion into steam. Where the cooling water for

power station is drawn from large rivers , it can be returned directly to the source after use.

Cooling towers are used where about one hundredth part of the cooling water evaporates

and certain amount is returned to its source to carry away any impurities that collect.

SYSTEM ANALYSIS..

Control and instrumentation

The control & instrumentation package of Kahalgaon project (4x210 MW + 3 x 500) is supplied by Russian with the Main package by M/s. TPE, USSR. In addition to the original Russian supplied system there are Furnace Safeguard Supervisory System (Make Forney, USA), Data Acquisition System (Make Westinghouse, USA), Steam Water Analysis System & uninterrupted Power Supply system (supplied by Keltron).

The C&I package can be broadly classified into the following categories.

1. Process Measurements2. Process Protections3. Interlock & Protections of Boiler and Turbine auxiliaries4. Selective Control System & Short Sequence System5. Auto Control system6. Annunciation system.7. Furnace safeguard supervisory system 8. Data acquisition system 9. SWAS and its auxiliaries

Communication

Satellite communication offers a great number of advantages over terrestrial links, that is

why, it is preferred over terrestrial communication, namely :-

High availability – minimal/nil fading No last mile problem

Extensive coverage-Hemi/Zonal/Spot Broad cast nature allows easy point to multipoint links Easy scalability and expandability of network Security and secrecy can be adequately addressed Less prone to failure due to natural calamities

The disadvantages being:-

High up- front investment in ground segment High recurring cost on satellite segment.

A Satcom exchange has also been provided in NTPC-Khalgaon, which caters to the communication needs all its 96 employees. The layout of the SATCOM exchange BPL ∑ INOX 250, NTPC- Khalgaon has been shown in figure I. The out door unit basically consists of the Antenna. The indoor unit comprises a modem, multiplexer, router, DAMA, DTE, EPBX. The users can connect to the worldwide web or the company’s internet using this system. Using a PC the users can connects to the company LAN via a modem and then via server all the outside mails are router to the Proxy server. From the router the data is routed to the multiplexer from where it goes to the RFT via a modem. The data is transmitted via the antenna and the satellite to the NTPC earth station at Muradnagar (now NOIDA). To connect to the Corporate Center a 2Mbps micro wave link is used. A proxy server is also present in the Corporate Center from where it connects to the Internet w

world via CGI and MTNL/BSNL server. Private Branch Automatic Exchange. The EPBAX connects to

the main plant telephone exchange as well as a member of subscribers. These provide the facility

of voice communication via satellite to the users. This is particularly useful in case of terrestrial

links, which are much more damage prone. GDAMS data is also sent to the corporate centre using

this link.

SYSTEM DESIGN

C&I

1. PROCESS MEASUREMENT

The process parameter measurement is realized by means of Transmitters,

Thermocouples, RTDs and other primary instruments. The signals received from these

primary instruments are utilized for Control /Record /Indication of the process parameter

and protection of the process. In the process where mass monitoring is required like Boiler

metal temperatures, Turbine metal temperatures, Turbine bearing temperatures and

Generator slot temperatures in these cases either multipoint recorders or indicators with a

manual switch for point selection is provided. For remote indications of level, pressure and

flow, transmitters are providing. Explosion proof Transmitters are provided for explosion

prone areas like Fuel oil pump house and Hydrogen plant. All transmitters are of strain

gauge type, which gives an output of 4 – 20mA when the pressure across the diaphragm

varies from minimum range to maximum range.

For some process parameters the signal for DAS & Measurement is shared from a

single transmitter. This sharing is done through Zener diode isolation circuit. Generally for

Auto Control System dedicated transmitters are provided with 100% redundancy.All the

secondary instruments are powered from UPS including Recorders, Indicators, TSI

equipment & all process variation measuring transmitters.

Unit Control Board (UCB) is distinctly divided into two parts. One is operative area

which contain Desk panels, Annunciation panels, Recorder & Indicator panels, Transmitter

power supply panels & Mimic panels .The other is the Non- operative area consisting of

Interlock and Protection panels, Auto Control panels, Selective and Short sequence panels,

RC feeder & FSSS panels.

There is facility to control the operation of all drives, valves & dampers from

operator desk panel. Status of the same is indicated through mimic just above the operating

panel. In the operator desk

The recorders & indicators are of three different types depending upon the inputs

i.e. milliampere, potentimetric and bridge type. These are utilised for measurement of

transmitter signal, thermocouple signal and resistance thermal detector signal receptively.

For the measurement of excess Oxygen in flue gas, an insitu zirconia probe (Make

Westinghouse, supplied by I.L.Kota) is installed at 41 meter in second pass of Boiler. The

output of the probe is mv, which is converted to mA through converter. The cell constant

and cell slope is adjustable. Auto calibration facility is also available.

In Air preheater system, which is of tubular type, differential temperature of air & gas is

measured in a recorder. This gives annunciation when it crosses the alarm value. In case of

fire the differential temperature increases and when it crosses a pre-set value alarm is

obtained at UCB. For cleaning these tubular type

For Drum level protection as per the original design, level switches are provided at

different levels. However presently protection of drum level is of two out of three logic

rooted through Hydrastep (Keyston Make) both left and right side and the pressure

compensated signal from drum level compensation package (supplied by M/s Bells

control). Bearing temperature, winding temperature PA Fans & Mills are shared by DAS

and measurement from dual RTD. Coal pipe temperatures of furnace inlet are exclusively

provided in DAS only.

There are many important parameters, which are measured locally through pressure

gauges & switches. In Boiler Feed Pumps the bearing temperature recorders & winding

temperature indicators are located at local which helps the operator for easy monitoring of

the temp regularly.

For safe operation of Turbine the following Turbo Supervisory measurements are

carried out in continuous basis. .

01. Axial shift.

02. HPC / IPC differential expansion.

03. Eccentricity

04. HPC/IPC thermal expansion

05. Turbine speed measurement

06. Turbine vibration of Bearing 1 to 7.

07. Seal Noise device for Bearing 1 to 7.

08. Control valve servo meter position measurement.

Axial Shift, HPC / IPC/ LPC differential expansion measurement are realised by a

instrument based on the principle of Linear voltage differential transducer principle. Axial

shift pick up is fixed near brg-2 i.e. the pivot point of the turbine rotor.

For Turbine speed measurement a permanent magnet pick up is mounted above a wheel

of 60 teeth attached to turbine shaft located at MOP. This pick up is having a coil which is

induces an A.C current which is processed by a pre amplifier to get a square pulse. This

square pulse is finally counted by the counter located in NOP area which gives three

Analogue & three digital output used for local / UCB analogue / digital measurement.

For Turbine vibration measurement in each bearing there are two pick-ups. One is for

vertical vibration & other one for axial & cross sectional vibration. Vibration signal is

measured by piezoelectric crystal, which induces a millivoltage as per the vibration, which

is subsequently amplified by preamplifier at local. At UCB the signal is processed &

integrated to get velocity measurement output in the range of 0 – 12 mm/sec or 0 –

30mm/sec. The signal is converted into analogue output of 0 – 5 mA who is shared DAS &

measurement. In each pick up there is a facility of on line checking of healthiness by

injection of A.C voltage another piezoelectric crystal placed near to the measuring crystal.

There is a provision of two-alarm contact for alarm & warning setting for the measurement

of each vibration.

In offsite areas including Fly Ash, Bottom Ash, FOPH, UPH, H2 & chlorine plant, for

easy monitoring and operation of the equipment in addition to local Pressure gauge,

pressure switch, Recorders and instruments indication the temperature & current etc. are

also provided near the equipment itself. The ON/OFF status of the major offsite drives are

available in UCB # 1. In case of emergency i.e. when any local annunciation is generated a

parallel contact goes to UCB and ‘control board faulty ‘ annunciation appears. This helps

the UCB personnel to maintain & monitor the offsite area.

In Fuel Oil Pump House their three receiving tanks, each consisting of two oil transfer

pumps. There are four oil storage tanks in which level and temperature are measured. There

are two sets of lift pumps. Each set consists of two stages. Suction and discharge pressure is

measured through local gauges. Fuel oil flow to boiler and return oil flow from boiler is

measured through transmitters.

There are two BottomAsh a pump house. Each Bottom Ash pump house consists of

three sets of pumps. In each set there are two pumps connected in series. This arrangement

is done to give more discharge Pressure to the slurry. In each pump Bearing temperature is

measured & dedicated recorders are provided in local panel. Pneumatic valves are provided

in the discharge of the pumps. Gland sealing is required for each stage of pump. In Air

Compressor house, there are four compressors common to all four units. The Compressor

has got two independent motors winding for two independent speeds (Low speed and High

speed). For each stage air temperature before cooler is measured and independent tripping

is initiated when it rises beyond the alarmed value. Compressor trips when lubrication oil

pressure falls below the set limit and cooling water flow is absent.

2. PROCESS PROTECTION

Protection intended to prevent the damage of the equipment under the alarm conditions

can be broadly divided into the following categories.

1. Unit protection.

2. Turbine protection

3. BFP protection

4. Boiler trip protection

5. Boiler load drop protection

6. QPRDS protection

7. Turbine local protection (HPH,LPH,D/A)

4. AUTO CONTROL SYSTEM

The Auto Control system is intended to maintain process variables within a

predetermined range and to ensure the reception of process data. The system produces

control instructions & actions which helps in achieving the predetermined state.

The system is based on a split architecture. Controllers comprises of two separate units

i.e regulating unit and operative unit.

The Controllers actions are processed through different blocks which are used for

different purposes .Some of the important blocks are mentioned below.

1.Signal input block.

2.Signal converter block,

3.Integration block

4.Summator block

5.Pulse output block

6.Annunciation block

The input to the system is of 4 – 20 mA , millivolts or resistance depending upon the

type of transducer used which in-turn converted to 0 – 10v for block operation through

input signal block. This also optically isolates the input & output signals.

The pulse output block compare the process parameter set point , process parameter

measurements and control valve position and gives output accordingly for the control

valve.There is a signaling unit, which compares the parameter value with a predetermined

set point. Deviation beyond that gives alarm & changes the Auto operation into Manual

Mode. In the event of abrupt failure such as open circuit or short circuit in transmitters

controller trip action takes place through this block. Parallel command goes to annunciation

blocks also.

For some important controllers redundant transmitters are provided. Provisions are

there for manual change over to the standby in case of failure of the working

transmitter .Provisions are there for bumpless transfer of actuator from auto mode to

manual mode.

The main automatic controllers are as detailed below.

A. UNIT CAPACITY CONTROL

This controller is for maintaining a controlled TG power output and mainsteam inlet

pressure to turbine. This controller incorporate Load change limiter, Boiler capacity control

and Turbine capacity control. Boiler capacity controller will receive unit preset capacity

signals from unit load rate of change limiter and the actual power output. It generates a

demand signal to a coal flow controller & total air flow controller. Turbine capacity

controller will receive signal from actual power output , rate of feed forward signals of the

preset output, the error signal representing the discrepancy between the demand and actual

main steam inlet pressure to for time . Turbine capacity control initiates command to

operate the speeder gear motor.

B. COMBUSTION CONTROL

Combustion control is done by means of Coal flow controller , Mill loading

controller, Total Air flow , Furnace draught controller & F.O pr. controller. Summation of

Coal flow & Fuel Oil flow are used for Total air flow controller which maintain a

predetermined value of free oxygen in flue gas. The loop actuates individual actuate of FD

fans dampers for maintaing the specified free oxygen in flue gas. Mill loading controller

regulates feeding of coal to the pulverizes. Furnace draught is controlled through

controlling the inlet vanes of ID fans.

5. ANNUNCIATION SYSTEM

The Annunciation system is intended to make available the information about the

equipment under control, deviation of measured parameters from the normal (preset) value

and interlock action. Alarm is provided with each annunciation block.

Visual status indication of equipment under control i.e Devices and Stop valves

Audio-visual alarm annunciation of emergency tripping and autostart of equipments

Audio-visual process annunciation.

Audio-visual preventive alarm annunciation of parameter deviation from normal values.

There are three major section in the annunciation circuit. Section – I controls the

Boiler & off site annunciations, section – II controls the turbine related annunciations and

section – III looks after the generator related annunciation. The circuit is powered by 110v

DC which is generated by 3 phase 415v AC rooted through a transformer and diode

rectifier circuit.

Operating status of important drives protection status and alarm conditions, when

arises actuate the annunciation system. Once the alarmed value is reached by any signal, it

flashes & hooter circuit is energizes. Flashing takes place till the resetting is not done. If the

alarmed condition still exist then it glows steadily. Each annunciation section will be

provided with test push buttons to check the audio annunciation signal and with the panel

switches for lamp healthiness checking.

6. FURNACE SAFEGUARD SUPERVISORY SYSTEM

The Furnace safeguard supervisory system supplied by Forney, USA i.e of Model

AFS – 1000 is a 16 bit 80186 microprocessor based system. The main task of FSSS is to

control the fuel system equipments i.e. oil guns, igniters, PA Fans, Mills, seal air fans and

their associated dampers etc. so as to ensure safe operation of the Boiler under start up,

transient and normal conditions. Besides, one of the most important FSSS task is to

implement process protections with provision for a first up annunciation facility on

operating console with respect to the first actuated protections.

AFS – 1000 Burner management system employs plug in printed circuit board

assemblies.This permits maintenance by replacing board without causing the total

shutdown. The Burner Managent System logic comprises of ten nos. of subsystems

(Excluding Maintenance subsystem – II) per each unit each responsible for predefined

portion of the overall system. There are two CRT master subsystem ,one purge and fuel

safety subsystem, one oil burner subsystem & six mill subsystems. Each subsystem is

responsible for a portion of overall control.

A subsystem consist of a single - board control card (80186 up based ) , solid state input /

output printed circuit board assemblies, starburst keyboard with intercommunication

through fiber optic Assembly and a shared operation interface through fiber of the cable

assembly and having touch screen facility. There are also some logic derived inputs that got

transmitted from one subsystem from another and all subsystem operate independently of

other subsystem.

CRT master subsystem:

The primary function of each CRT master subsystem is to act as controller of the

intercommunication bus (ICB) used to pass data back and forth between the master

and the other subsystem. It also decodes commands that it receives from CRT

screen. In the CRT screen the last acknowledged alarm, Printer ON/OFF status and

Command elements appear of which print out can be taken if required. For

depicting valves ,dampers and drives different symbols were used and when status

changes the colour changes.

01. Purge and fuel safety subsystem :

The purge and fuel safety subsystem comprises of the following logic.

Purge logic has the major responsibility for insuring that the furnace is purged

thoroughly with air to remove any combustible before the fuel is released into the

headers & into the furnace. For starting the purge the following conditions should

be met.

a). All burner oil valves closed.

b). Oil trip valve closed.

c). Oil re-circulation valve closed

d). No flame detected

e). Scanner Air fan on.

f). Any Induced Draught & Furnace Draught running,

g). All primary fans , pulverizers & Raw coal feeders off.

h). All hot air damper closed.

i). Igniter trip valve closed.

j). No trip (Boiler) present

k). Air flow >30%

After all these conditions are satisfied PURGING takes place for 10Minutes.

03. Pre light Logic: When furnace purging is completed pre light logic is responsible for

the safe release of fuel into the headers. Pre light logic is also responsible for

communicating to the banners subsystem that furnace conditions are satisfactory for banner

operation.

04. Fuel Safety logic – Fuel safety logic is responsible for overall boiler safety. It

monitors the potential danger points throughout the boiler. If the predetermined limits are

crossed, the fuel safety logic shuts off some or all fuel to the furnace depending upon the

conditions. The cause of first out alarm function indicators the cause of the tripping. The

MFT 1 and MFT 2 relays get trip command when any of the following conditions is / are

initiated.

01. Both the ID fans off.

02. Both the FD fans off.03. Operator trip04. Drum level high / low05. Loss of all fuel06. Loss of all flame07. Unit trip from BPS08. Furnace pressure high/ low09. Delayed light off (10 min after purging)10. Air flow (< 30%) low.

05. Burner control sub system – It is responsible for following a predetermined

programme in safety placing burners in service, removal from service, purging of burners

that have been removed from service and for monitoring burner operations.

01. Mill control subsystem- Milling subsystem is responsible for controlling the

coal burners and associated pulverizer, primary Air fans, feeders and associated

program in safety taking burners in service, removal from service and

monitoring burner operation.

02. Maintenance subsystem-II.

The MS-II is an equipment helping for fault finding or trouble shooting of the

systems input as well as to change the program logic. It comprises of on operation console,

a control unit, interfacing cable assembly, graphic CRT / KBD and menu driver software.

7. DATA ACQUISITION SYSTEM

The Data Acquisition System of 4x210 Mw KhSTPP was supplied by M/S

ESPL ,Chandigarh. This is a WDPF family of system manufactured by Westinghouse with

software level version 6.4. This system is a microprocessor based distributed modular type

with independent system, drops connected to each other through high speed standard

Westnet data highway having 100% redundancy. The drops talk to each other directly over

the data highway, which eliminate the need of host computer. The system caters 900 analog

points and 1400 digital points in each unit Each system supports two units. The standard

WDPF drops and products perform specific functions, which allows a custom system to be

configured by determining required functions and selecting the appropriate standard

equipment.

8. UPS PACKAGE

UPS system of capacity 75 KVA has been supplied by M/S KELTRON ,Trivendrum

for each of the four units which caters the power supply for all C&I system including Data

Acquisition system. FSSS, Instrument supply, ACS etc. The system consists of 2 nos.

chargers, 2 nos. inverters, and one paralleling unit, one servo controlled voltage stabiliser

(SVCS) and one set of Battery (550AH) for one-hour backup at 75KVA load. The two

inverters, each of which is rated at 75KVA, are connected together to the O/P busbar to

share 50% of the load. In case of failure of one inverter the other one takes up the full load

automatically. The two chargers, which are meant for feeding the inverters and charging

the battery, also work in parallel and each one can take the load if needed. The set of

battery is connected at the output of the chargers and input of the inverters. In case of

failure of both the chargers the battery automatically takes up the load and starts feeding to

the inverters without any interruption. The SCVS is also connected to the output busbar. In

case of failure of both the inverters the SCVS gets connected to the output busbar through

the static switch with a maximum time delay of 20millisec.

9. ANALYSER AND MISC. C&I EQUIPMENT PACKAGE:

The package was awarded to M/S Keltron, Aroor. The package contents of supply of

the below mentioned instruments.

(1) Analyser for steam and water. Analyse system and Flue gas oxygen analyser.

(2) Miscellaneous C&I equipment such as switch drivers etc. and maintenance and

Calibration equipment for plant.

(i) The supply and erection and commissioning were in the scope of M/S Keltron.

(i) Only the supply was in the scope of M/S Keltron.

Part (1) is consisting of the following major equipment.

(a) Flue gas oxygen Analyser.(b) Secondary coolers.(c) Silica Analyser.(d) Hydrogen Analyser.(e) PH analyser.(f) Conductivity.(g) Phosphate Analysers.(h) Dissolved oxygen.

Part (2) consists of the following major. Equipment's.

(a) Temperature gauges.(b) Temperature switch.(c) D.P switch.(d) Level switch.(e) Flow switch.(f) Thermocouples(g) Flow elements.

Maintenance and calibration equipment.

(a) Pneumatic test bunch.(b) Vacuum tester.(c) Thermocouple test furnace.

(d) Flowmeter calibrator.

(e) Coil winding machine.

(f) Soldering iron

(g) Radial drilling machine.

(h) Solder sucker.

(i) Variac.

(j) Rheostat/Potentiometer.

(k) RCL Bridge.

(l) Logic Probs.

(m)Portable pneumatic calibrator.

It and communication…

SATELLITE: -

The third transponder of INSAT-3E with a bandwidth of 18MHz has been

allocated to NTPC to cater to its communication purposes. The bandwidth was updated to

27MHZ on 19th of July, 2005.The Kahelgaon satellite Communication Plan is given below:

-

Satellite: INSAT 3E

Location : 55 DEG EAST

Transponder No. : 3

Transponder Bandwidth : 27MHz (6012.00MHz-

6039.00MHz)

Beacon Frequency : 4190.97MHz

Polarization : E/S U/L LINEAR HORIZONTAL

E/S DN/L LINEAR VERTICAL

MCPC Stn. Spt Tx. Setting

: 6025MHz

MCPC Stn. Spt Rx Setting

: 3800MHz

ANTENNA:-

The antenna is of CASs grain configuration using shaped reflector technique to

optimize receive gain to noise ratio and the corrugation technique for primary horn feed to

get better symmetry and minimum side lobe of the radiation pattern. The antenna

specification is given below:-

Type: LIMITED STEERABLE AZ-EL MOUNT

Feed And Reflector : 7.5M dia. CASSEGRAIN WITH SHAPED REFLECTOR

Tracking : MANUAL DRIVE, AUTO STEP TRACK

Sky Coverage : EL +5deg - +90deg

AZ -20deg - +20 deg

Frequency Range : TX 5.850-6.425GHz

Rx 3.700-4.200GHz

Polarization : LINEAR ORTHOGONAL AND CIRCULAR

Gain : 6GHz BAND- 51+20logf/6 dB

4GHz BAND –47.5+20logf/4dB

Power Handling Capacity

: 5KW C W

Antenna Weight : 9 TONNES (Approx)

TRANSCIEVER:-

The AAV680 C-Band single Package Transceiver ODU interfaces

with the 70MHz/140MHz Indoor Unit having bandwidths of + 18MHz.

In Single Package RF Transceiver there 70 or 140 MHz if input from the indoor

modulation to an RF signal in the C-Band, transmission via antenna and down convert the

L-Band signal (950 – 1450 MHz) to an IF signal of 70/140 MHz. for the demodulator.

The Single Package Transceiver has three parts…

Phase Locked Low Noise Block ( PLLNB)

Booster for 16W and above (optional)

Accessories

It also has two modules…

L band module comprising,

This also has some partitions. As follows..

1. synthesizers

2. up converter

3. down converter

4. monitor & control

Power Block Up Module (PBU)

This also has some partitions. As follows.

1. Power supply board

2. SSPA module ( Solid State Power Amplifier )

3. BUC module ( Block Up Converter )

It designed as a single PCB. All SSPA configurations are designed as soft boards bounded

onto aluminum base plates. A low noise amplifier is provided which receives the weak

downlink signal (3.625-4200MHz) and amplifies it to the correct level before sending it to

the SPT for frequency down conversion. It uses HEMT devices to achieve low noise, high

gain and low distortion amplification features. A Booster is provided to boost up the

transmit power of the signal from SPT.

For better reliability, two streams of outdoor C-Band and Ex C-Band

transceivers have been provided. A A low noise amplifier is provided which receives the

weak downlink signal (3.625-4.200MHz) and amplifies it to the correct level before

sending it to the SPT for frequency down conversion. It uses HEMT devices to achieve low

noise, high gain and low distortion amplification features. A Booster is provided to boost

up the transmit power of the signal from SPT.

For better reliability, two streams of outdoor C-Band and Ex C-Band transceivers

have been provided. Redundancy switching equipment performs the switchover operation

when a fault is detected at any one of the streams. Switchover can be auto/manual and

results in less than half a second of traffic interruption. Independent path switching between

the transceivers is allowed.

There is a LNA device. It receives very weak down link signal and amplifies it to the right

level before sending to SPT for the frequency down conversion. This procedure uses High

Electron Mobility Transistor (HETM) for low noise, high gain, low distortion in

amplification. It requires 12 V DC.

Another equipment is used named BOOSTER. This is used to boost the transmit

power of signal from SPT. It has two pats.

1. Power supply board

This converts AC to DC for SSPA.

2. SSPA (Solid State Power Amplifier )

This receives RF power from SPT and amplifies to correct power level.

RCU…

The redundancy control unit is supplied with +12 or +15 V DC via agile transceivers.

The unit contains..

Monitor & control (M & C ) module

Window access panel (WAP )

IF input power splitter

IF output switch

M & C…

This module comes with an intel based microcontroller to link the monitor and control

functions from the out door RCU to the indoor DTE (usually PC).

To active remote /local mode, toggle the DIP switch at the out door RCU window

access panel.

NOTE:- Users CAN NOT active remote or local mode from the indoor DTE.

MODEM:-

The word "modem" is a contraction of the words modulator-demodulator.

The digital satellite modem serves as an interface between the user’s data terminal

equipment and the IF frequency interface with the up/down converter. The UMOD

has been configured for full duplex operation. The transmit and receive paths are

independent for most applications.

In the transmit direction the UMOD accepts user data at the common interface

module (CIM) and directs it across the backplane to the transmit portion of the terrestrial

data interface which converts the users electrical format to the format used in UMOD. The

data is directed to the optional internal framing unit for processing and then to the UMOD

motherboard. On the motherboard the data is sent to the channel encoder where scrambling,

differential encoding & FEC encoding is performed. The data is then routed to the transmit

filter for digital filtering and interpolation; then passed to the modulator where the signal is

PSK modulated onto an IF carrier provided by the transmit synthesizer. This modulated

carrier is then amplified inn the IF stage, then routed for transmission across the backplane

to the IF OUT connector on the IF panel.

In the receive direction the IF signal is input at the IF IN connector on the IF Panel, passed

across the backplane and received by the receive IF processor on the UMOD motherboard

which performs low noise amplification, automatic gain control and filtering. The signal is

then routed to the receive synthesizer and demodulator where the IF carrier is removed by

either BPSK or QPSK demodulation. The resulting base band data is then directed to the

channel decoder where it is FEC decoded, differentially decoded and descrambled. The

data then passes through the optional IFU daughter card where deframing and other

processing takes place. The data is then routed to the receive portion of the terrestrial data

interface daughter card which converts the receive data and clock to appropriate formats

and directs it across the backplane to the CIM where they can be accessed by the user. The

UMOD block diagram is illustrated above:-

MULTIPLEXER:-

The Kilomux is an advanced, highly versatile user configurable modular

TDM system providing an efficient method for transmitting data, voice and fax over digital

data services. Its main functions are:-

Multiplexing/De-multiplexing operation

System management

Interfacing with the optional external system management.

The basic Kilomux system consists of two kilomux units interconnected

with each other. Two main links are there to provide for standby redundancy for the main

link. The Kilomux uses permanent on-demand allocation of main link bandwidth.

The Kilomux is designed for unattended operation. A complete collection of

parameters configuring the kilomux system and each of its modules,is determined by a

database which is stored in the non-volatile memory of the Kilomux control module.

Kilomux can store two different databases and can be configured to switchover

automatically between databases per requirement.

It is provided with alarm buffer as well as LED indicator to display in real time the main

link and power supply status. The system can be managed using either of the following:-

Front panel using push buttons and LCD

ASCII terminal connection

Telnet connection via the Ethernet LAN/SLIP

connection

RADview network management station.

ROUTER:-

Routers are specialized computers that send your messages and those of every

other Internet user speeding to their destinations along thousands of pathways.. These are

crucial devices that let messages flow between networks rather than within networks. A

router has two separate but related jobs:

The router ensures that information doesn't go where it's not needed. This is crucial for

keeping large volumes of data from clogging the connections of "innocent bystanders." The

router makes sure that information does make it to the intended destination.

It joins the two networks, passing information from one to the other and, in

some cases, performing translations of various protocols between the two networks. It also

protects the networks from one another, preventing the traffic on one from unnecessarily

spilling over to the other. As the number of networks attached to one another grows, the

configuration table for handling traffic among them grows, and the processing power of the

router is increased. A configuration table is a collection of information, including:

(i) Information on which connections lead to particular groups of addresses

(ii) Priorities for connections to be used

(iii) Rules for handling both routine and special cases of traffic.

ROUTING OF PACKETS:-

The office network connects to the router using an Ethernet

connection. There are two connections between our router and the ISP. One is via the

satcom system and the other via the BSNL ISDN line as is illustrated in figure I. This way

the ISDN line is held as an “insurance” against a problem with the other faster connection.

In addition to routing packets from one point to another, the router

has rules limiting how computers from outside the network can connect to computers inside

the network, how the network appears to the outside world, and other security functions.

While most companies also have a special piece of hardware or software called a firewall

to enforce security, the rules in a router's configuration table are important to keeping a

company's (or family's) network secure.

One of the crucial tasks for any router knows when a packet of

information stays on its local network. For this, it uses a mechanism called a subnet mask.

The subnet mask looks like an IP address and usually reads "255.255.255.0." This tells the

router that all messages with the sender and receiver having an address sharing the first

three groups of numbers are on the same network, and shouldn’t be sent to another

network.

Thus knowing where and how to send a message is the most important job of a

router. Some simple routers do this and nothing more. Other routers add additional

functions to the jobs they perform. Rules about where messages from inside a company

may be sent and from which company’s messages are accepted can be applied to some

routers. Others may have rules that help minimize the damage from "denial of service"

attacks. The one constant is that modern networks, including the Internet, could not exist

without the router.

EPABX

What is an EPABX?

A private automatic branch exchange (PABX) is an automatic telephone switching system within a private enterprise. Originally, such systems - called private branch exchanges (PBX) - required the use of a live operator. Since almost all private branch exchanges today are automatic, the abbreviation "PBX" usually implies a "PABX."

Some manufacturers of PABX (PBX) systems distinguish their products from others by creating new kinds of private branch exchanges. Rolm offers a Computerized Branch Exchange (CABX) and Usha Informatics offers an Electronic Private Automatic Branch Exchange (EPABX).

An EPABX is a telephone system within an enterprise that switches calls between enterprise users on local lines while allowing all users to share a certain number of external phone lines. The main purpose of an EPABX is to save the cost of requiring a line for each user to the telephone company's central office.

EPABX is owned and operated by the enterprise rather than the telephone company (which may be a supplier or service provider, however). Electronic Private Automatic Branch Exchanges used analog technology originally. Today,EPABXs use digital technology (digital signals are converted to analog for outside calls on the local loop using plain old telephone service).

An EPABX includes:

Telephone trunk (multiple phone) lines that terminate at the EPABX A computer with memory that manages the switching of the calls within the EPABX and

in and out of it The network of lines within the EPABX Usually a console or switchboard for a human operator In some situations, alternatives to an EPABX include centrex service (in which a pool of

lines are rented at the phone company's central office), key telephone systems, and, for very small enterprises, primary rate Integrated Services Digital Network.

PBX, EPABX and Key Telephone Systems:-A Comparison

Private Branch Exchange

Private Branch Exchange, a private telephone network used within an enterprise. Users of the PBX share a certain number of outside lines for making telephone calls external to the PBX.

Most medium-sized and larger companies use a PBX because it's much less expensive than connecting an external telephone line to every telephone in the organization. In addition, it's easier to call someone within a PBX because the number you need to dial is typically just 3 or 4 digits.

A new variation on the PBX theme is the Centrex, which is a PBX with all switching occurring at a local telephone office instead of at the company's premises.

If the telephone system in your organization is based on a legacy PBX (Private Branch Exchange) that uses old style telephone lines then PBX can offer a range of affordable products that will enhance your corporate telephony infrastructure while dramatically reducing overall capital and operating costs. PBX are at the forefront of the next generation of VoIP based PBX replacements that offer tremendous benefits by consolidating your organization's data and telephony capabilities to a single network.

Here are few of the many benefits of migrating to PBX VoIP:

Operating Cost Reductions .

Call charges are drastically reduced for IP based traffic to and from remote offices. PBX will use a reliable VoIP connection if it is available or it will fall back to a PSTN connection.

A major operating cost of legacy telephony systems is the required intervention by IT managers or technicians when staff moves around the organization. In a PBX VoIP network each user is immediately recognized by handset or log on.

Minimum Investment Risk .

Minimum investment is required to get started since the PBX VoIP technology is embedded in your existing corporate data network. Your company can invest in a PBX and just a few SIP handsets to begin with. Subsequent telephony system expansion is as simple as plugging extra handsets into your existing LAN connections.

Installation Cost Reductions .

Our PBX models are among the cheapest on the market and hardware installation is so simple that there is no need to employ technicians or specialists. Just three cable connections are needed for many installations. Software configuration is done via your corporate intranet and this is also a simple procedure that does not require specialist staff. All of this equates to significantly reduced installation, upgrade and operating costs compared to legacy PSTN-based PBXs.

Improved Employee Productivity and Control of Telephony Traffic.

The wealth of enhanced telephony user features that arrive with PBX will increase employee productivity while the superb array of built-in call recording and archiving features will facilitate control of telephony traffic.

Flexibility and Upgradeability .

To upgrade legacy PBXs a technician would need to physically visit the site and make wiring changes or even replace the unit. PBX is software based and so upgrades can be performed remotely via the internet.

Some Further Details:-

Call Cost Reduction with no Sacrifice of Reliability.

PBX lowers the cost of your long-distance calls by automatically routing each call through the cheapest available VoIP long-distance provider. This is achieved with maximum reliability through a fail-safe sequential selection of:

Automatic Recording of all Incoming and Outgoing Calls.

PBX is uniquely designed to provide non-intrusive recording of all phone calls in both directions. An option is available to record each party in a separate audio channel. If this option is enabled

the remote party will be recorded in the left channel of an audio file while the local party will be recorded in the right channel of the same file.

PBX can be configured to comply with local laws and regulations that require you to notify either party that the call is being recorded. It can be configured to play a call recording notification automatically. Several standard call recording prompts are available. Here are two examples:

'This call may be monitored and recorded.'

'To ensure the highest level of customer service, this call may be monitored and recorded.'

There is also an option to record your own prompts.

Automatic Backup to CD or DVD.

PBX can easily be configured to make a periodic back-up of all conversations onto CD or DVD media. CD-R/RW and DVD±R/RW formats are supported. You can create a CD or DVD archive of all phone conversations within your organization and with the outside world using just a few mouse clicks. Each CD or DVD is given a unique label. This allows you to easily find the conversation you need. You can access your archived data using the PBX device or any PC.

Key Telephone Systems

Often referred to as just KTS, a key telephone system is a premises telephone system that is best known by the phones that have buttons for calling inside an organization and for placing calls outside through the public telephone network. A key telephone system is in the same category as a PBX (private branch exchange), except that key systems rely on the telephone company switching equipment, while PBXs rely on a central control unit located at the customer site. In other words, with a key system, the dial tone is generated at the telephone company central office. A full PBX generates its own dial tones. Key systems also do not require dialing a number to gain an outside line since all lines are already directly connected to the telephone company central office. On a PBX system, lines are connected to the PBX, and the PBX makes connections to the central office when the outside number is dialed.Infact in a local environment, terminals and equipment that provide immediate access from all terminals to a variety of telephone services without attendant assistance.

A key system or key telephone system is a multiline telephone system typically used in small office environments.Key systems are noted for their expandability and having individual line selection buttons for each connected phone line, however some features of a private branch exchange such as dialable intercoms may also commonly be present.Key systems can be built using three principal architectures:

Electromechanical shared-control Electronic shared-control Independent keysets

Electromechanical shared-control key systems

Before the advent of large-scale integrated circuits, key systems were typically built out of the same electromechanical components (relays) as larger telephone switching systems. The system marketed in North America as 1A2 was entirely typical and sold for many decades. This system consisted of a central control unit and a number of specialized telephone sets. Each line to the telephone sets was routed using six wires:

Two wires (one pair) carried the actual telephone line Two wires (a second

pair) carried control information for that line

Two wires (a third pair) carried current to a lamp

installed at the telephone

A telephone set could contain five, 12, or many individual telephone lines. A common five-line keyphone would be connected using 25-pair cable and an Amphenol 50-position "MicroRibbon" connector. The lamps installed at the telephone sets allowed the user to instantly determine the status of all of the individual telephone lines that "appeared" at that set:

Lamp off — The line is idle Lamp steady on — The line is in use for a call Lamp flashing slowly — The line is ringing with an incoming call Lamp winking fast — A call on the line is "on hold"

A user could select any of the lines simply by pressing the appropriate line button and picking up the handset. A caller could place a call "on hold" by pressing the red "hold" button. This would place the call on hold and then mechanically release the depressed line button, allowing the user to select another line.

An individual worker or executive might have a set with one or a few lines "appearing". The system attendant (receptionist) might have a set with many lines appearing so that they could monitor the status of all incoming lines simultaneously.

These systems also supported manual buzzers, intercom lines (with or without selective ringing), music on hold, and other simple features. The features were provided on a line-by-line basis by the selection of particular Key Telephone Units (KTUs) plugged into a pre-wired backplane in the central control unit. The central control unit also provided power for the entire key system (including ringing voltage). A mechanical interrupter in the power supply provided the pulsing voltages for the various lamps, buzzers, and ringers in the system.

Compatible 1A2 equipment was manufactured by a number of vendors including Western Electric, Northern Telecom, and Automatic Electric (GTE).

Electronic shared-control systems

With the advent of LSI ICs, the same architecture could be implemented much less expensively than was possible using relays. In addition, it was possible to eliminate the many-wire cabling and replace it with much simpler cable similar to (or even identical) with that used by non-key systems.

Additionally, these more-modern systems allowed vastly more features including:

Interactive voice response systems Answering machine functions Remote supervision of the entire system Automatic call accounting Speed dialing Caller ID

Features could be added or modified simply using software, allowing easy customization of these systems.

Independent keysets

LSI also allowed smaller systems to distribute the control (and features) into individual telephone sets that don't require any single shared control unit. Generally, these systems are used with a relatively few telephone sets and it is often more difficult to keep the feature set (such as speed-dialing numbers) in synchrony between the various sets.

Major Manufacturers

The major manufacturers of Telephone Exchanges are as follows:-

1. Zentek Electronics & Communication2. Yecaud Electronics Pvt Ltd.3. Webtel Electronics4. Warahi Engineers Pvt Ltd.5. V.r. Electronics6. Ushil Electronics7. Usha Infomatics8. Unitech Microelectronics9. Pearl Telecommunication & Electronics Pvt Ltd.10. Tangent systems.

Some Unique Features of EPABX:-

1. Multiple VoIP and PSTN backups for bullet-proof reliability. 2. 911 emergency dialing over a PSTN line.3. Extensive call recording and archiving.4. Automated recorded call backup to CD or DVD media.5. Highly scalable: 2 FXO, 4 FXO, Single or Quad T1/E1 units are available.6. Secure and easy to use Firewall/NAT bypass.7. Uses SIP phones/adaptors with automatic detection and configuration.8. Voice Mail.9. Auto Attendant.

Other Features of EPABX:-

1. Voice/Data logging.2. Caller ID with call annotation to detect incoming and outgoing numbers.3. All incoming and outgoing calls are logged. 4. Long distance routing through VoIP. 5. VoIP-VoIP and VoIP-PSTN backup. 6. Easy LCD-button operation. 7. PBX is not tied to a specific VoIP provider or existing network hardware. 8. ACD Queues. 9. Music On Hold. 10. RADIUS Accounting. 11. Configuration Backup/Restore on/from CD or DVD media. 12. User's interface in English, Russian, Czech, Spanish, Norwegian, and Swedish

Brief Description of the latest Features in EPABX

Based in Micro controller technologies:

New telecom systems usually have all the basic features of simple systems, and in addition they may have (some of) the following features. These features are designed to impress those who call you on this telecom system, control possible mis-use, and improve ease and efficiency Auto Attendant - an automatic operator feature to handle the incoming calls. This can be recorded with your VOICE MESSAGES to greet the caller, announce the name of your organization, and guide the caller further in order to reach the desired extension. A MATURE auto-attendant usually has a provision to record at least the BASIC FOUR MESSAGES namely greeting message, message on busy extension, message on no-reply of extension and release message. It may also have the capability of RE-TRYING the desired extension when it is not immediately available. Older IMMATURE versions (called Voice DISA) with only greeting message are also available at lesser cost.

Keyphone Console - Provision to connect a KEYPHONE as a 'Masterphone' or 'Operator Console' to handle higher telephone traffic. Caller-ID - Capability to read CALLER-ID or Calling Party's Telephone Number as per data provided by MTNL/BSNL line and show on:- ringing extensions, on extensions where call is transferred, in call billing data. Extensions may use normal CALLER-ID phones or keyphones

Additional features may include:

CALL BILLING with buffer memory and COMPUTER INTERFACE Auto- Redial Dynamic Locking Console Never Busy Fax Homing External Call Forwarding Remote Programming DOSA Auto Disconnection after set interval Call Budgeting

INTEGRATION WITH COMPUTERS, VOICE MAIL, ETC.

VOICE MAIL : It is a computer based "answering machine", which can be used for answering your calls when you are not available. It can be set up as a "single user" or "multiple users" option with each user having a separate private mailbox with a personal password. Voice guidance messages guide a caller to the respective mailbox and the voice mail messages of that person are saved in that mailbox. These messages can be played back by the owner of the mailbox, who can access this playback function using his secret password. CALL BILLING COMPUTER: A PC can be hooked up to the telecom system if it has a RS232C serial port on the output of the call billing buffer memory. The PC needs to have an extra COM port. All the call billing data from the telecom system can be transported to the PC's hard disc so that it can be merged with the data thus obtained for the previous and the later periods. This data may be used for analysis of calls made and costs incurred, independently or along with your other applications. This helps to control your telephone bills, and is useful especially if you have many STD/ISD lines.

What is VoIP?

Voice over Internet Protocol, is a method for taking analog audio signals, like the kind you hear when you talk on the phone, and turning them into digital data that can be transmitted over the Internet. VoIP can turn a standard Internet connection into a way to place free phone calls. The practical upshot of this is that by using some of the free VoIP software that is available to make Internet phone calls, you are bypassing the phone company (and its charges) entirely. VoIP is a revolutionary technology that has the potential to completely rework the world’s phone systems. VoIP providers like Vonage have already been around for a little while and are growing steadily. Major carriers like AT&T are already setting up VoIP calling plans in several markets around the United States, and the FCC is looking seriously at the potential ramifications of VoIP service.

The interesting thing about VoIP is that there is not just one way to place a call. There are three different “types” of VoIP services used in common day to day life.

1 .ATA - The simplest and most common way is through the use of a device called an ATA (analog telephone adaptor). The ATA allows you to connect a standard phone to your computer or your Internet connection for use with VoIP. The ATA is an analog-to-digital converter. It takes the analog signal from your traditional phone and converts it into digital data for transmission over the Internet. Providers like Vonage and AT&T CallVantage are bundling ATAs free with their service. You simply crack the ATA out of the box, plug the cable from your phone that would normally go in the wall socket into the ATA, and you’re ready to make VoIP calls. Some ATAs may ship with additional software that is loaded onto the host computer to configure it; but in any case, it is a very straightforward setup.

2. IP Phones - These specialized phones look just like normal phones with a handset, cradle and buttons. But instead of having the standard RJ-11 phone connectors, IP phones have an RJ-45 Ethernet connector. IP phones connect directly to your router and have all the hardware and software necessary right onboard to handle the IP call. Soon, Wi-Fi IP phones will be available, allowing subscribing callers to make VoIP calls from any Wi-Fi hot spot.

3. Computer-to-computer - This is certainly the easiest way to use VoIP. You don’t even have to pay for long-distance calls. There are several companies offering free or very low-cost software that you can use for this type of VoIP. All you need is the

software, a microphone, speakers, a sound card and an Internet connection, preferably a fast one like you would get through a cable or DSL modem. Except for your normal monthly ISP fee, there is usually no charge for computer-to-computer calls, no matter the distance.

But chances are good you are already making VoIP calls any time you place a long-distance call. Phone companies use VoIP to streamline their networks. By routing thousands of phone calls through a circuit switch and into an IP gateway, they can seriously reduce the bandwidth they’re using for the long haul. Once the call is received by a gateway on the other side of the call, it is decompressed, reassembled and routed to a local circuit switch.

Although it will take some time, you can be sure that eventually all of the current circuit-switched networks will be replaced with packet-switching technology (more on packet switching and circuit switching later). IP telephony just makes sense, in terms of both economics and infrastructure requirements. More and more businesses are installing VoIP systems, and the technology will continue to grow in popularity as it makes its way into our homes.

The Forrester Research Group predicts that nearly 5 million U.S. households will have VoIP phone service by the end of 2006. Perhaps the biggest draws to VoIP for the home users that are making the switch are price and flexibility.

With VoIP, you can make a call from anywhere you have broadband connectivity. Since the IP phones or ATAs broadcast their info over the Internet, they can be administered by the provider anywhere there is a connection. So business travelers can take their phones or ATAs with them on trips and always have access to their home phone. Another alternative is the softphone. A softphone is client software that loads the VoIP service onto your desktop or laptop. The Vonage softphone has an interface on your screen that looks like a traditional telephone. As long as you have a headset/microphone, you can place calls from your laptop anywhere in the broadband-connected world.

Most VoIP companies are offering minute-rate plans structured like cell phone bills for as little as $30 per month. On the higher end, some offer unlimited plans for $79. With the elimination of unregulated charges and the suite of free features that are included with these plans, it can be quite a savings.

Most VoIP companies provide the features that normal phone companies charge extra for when they are added to your service plan. VoIP includes:

o Caller ID o Call waiting o Call transfer o Repeat dial o Return call o Three-way calling

There are also advanced call-filtering options available from some carriers. These features use caller ID information to allow you make a choice about how calls from a particular number are handled. You can:

Forward the call to a particular number Send the call directly to voicemail Give the caller a busy signal Play a “not-in-service” message Send the caller to a funny rejection hotline

With many VoIP services, you can also check voicemail via the Web or attach messages to an e-mail that is sent to your computer or handheld. Not all VoIP services offer all of the features above. Prices and services vary, so if you’re interested, it’s best to do a little shopping.

Now that we’ve looked at VoIP in a general sense, let’s look more closely at the components that make the system work. In order to understand how VoIP really works and why it’s an improvement over the traditional phone system, it helps to first understand how a traditional phone system works.

1. Circuit Switching

Existing phone systems are driven by a very reliable but somewhat inefficient method for connecting calls called circuit switching.

Circuit switching is a very basic concept that has been used by telephone networks for more than 100 years. When a call is made between two parties, the connection is maintained for the duration of the call. Because you are connecting two points in both directions, the connection is called a circuit. This is the foundation of the Public Switched Telephone Network (PSTN).

Here's how a typical telephone call works:

1. You pick up the receiver and listen for a dial tone. This lets you know that you have a connection to the local office of your telephone carrier.

2. You dial the number of the party you wish to talk to. 3. The call is routed through the switch at your local carrier to the party you are calling. 4. A connection is made between your telephone and the other party's line using several

interconnected switches along the way. 5. The phone at the other end rings, and someone answers the call. 6. The connection opens the circuit. 7. You talk for a period of time and then hang up the receiver. 8. When you hang up, the circuit is closed, freeing your line and all the lines in between.

Let's say that you talk for 10 minutes. During this time, the circuit is continuously open between the two phones. In the early phone system, up until 1960 or so, every call had to have a dedicated wire stretching from one end of the call to the other for the duration of the call. So if you were in New York and you wanted to call Los Angeles, the switches between New York and Los Angeles would connect pieces of copper wire all the way across the United States. You would use all those

pieces of wire just for your call for the full 10 minutes. You paid a lot for the call, because you actually owned a 3,000-mile-long copper wire for 10 minutes.

Telephone conversations over today's traditional phone network are somewhat more efficient and they cost a lot less. Your voice is digitized, and your voice along with thousands of others can be combined onto a single fiber optic cable for much of the journey (there's still a dedicated piece of copper wire going into your house, though). These calls are transmitted at a fixed rate of 64 kilobits per second (Kbps) in each direction, for a total transmission rate of 128 Kbps. Since there are 8 kilobits (Kb) in a kilobyte (KB), this translates to a transmission of 16 KB each second the circuit is open, and 960 KB every minute it's open. So in a 10-minute conversation, the total transmission is 9,600 KB, which is roughly equal to 10 megabytes.If you look at a typical phone conversation, much of this transmitted data is wasted.

While you are talking, the other party is listening, which means that only half of the connection is in use at any given time. Based on that, we can surmise that we could cut the file in half, down to about 4.7 MB, for efficiency. Plus, a significant amount of the time in most conversations is dead air -- for seconds at a time, neither party is talking. If we could remove these silent intervals, the file would be even smaller. Then, instead of sending a continuous stream of bytes (both silent and noisy), what if we sent just the packets of noisy bytes when you created them? That is the basis of a packet-switched phone network, the alternative to circuit switching.

2. Packet Switching

Data networks do not use circuit switching. Your Internet connection would be a lot slower if it maintained a constant connection to the Web page you were viewing at any given time. Instead, data networks simply send and retrieve data as you need it. And, instead of routing the data over a dedicated line, the data packets flow through a chaotic network along thousands of possible paths. This is called packet switching.

While circuit switching keeps the connection open and constant, packet switching opens a brief connection -- just long enough to send a small chunk of data, called a packet, from one system to another.

It works like this:

The sending computer chops data into small packets, with an address on each one telling the network devices where to send them.

Inside of each packet is a payload. The payload is a piece of the e-mail, a music file or whatever type of file is being transmitted inside the packet.

The sending computer sends the packet to a nearby router and forgets about it. The nearby router sends the packet to another router that is closer to the recipient computer. That router sends the packet along to another, even closer router, and so on.

When the receiving computer finally gets the packets (which may have all taken completely different paths to get there), it uses instructions contained within the packets to reassemble the data into its original state.

Packet switching is very efficient. It lets the network route the packets along the least congested and cheapest lines. It also frees up the two computers communicating with each other so that they can accept information from other computers, as well. A technique used by at least one equipment manufacturer, Adir Technologies (formerly Netspeak), to help ensure faster packet delivery is to use ping to contact all possible network gateway computers that have access to the public network and choose the fastest path before establishing a Transmission Control Protocol (TCP) sockets connection with the other end.

Using VoIP, an enterprise positions a "VoIP device" at a gateway. The gateway receives packetized voice transmissions from users within the company and then routes them to other parts of its intranet (local area or wide area network) or, using a T-carrier system or E-carrier interface, sends them over the public switched telephone network.

H.323 and SIP

The Session Initiation Protocol (SIP) is an Internet Engineering Task Force (IETF) standard protocol for initiating an interactive user session that involves multimedia elements such as video, voice, chat, gaming, and virtual reality. H.323 is a standard approved by the International Telecommunication Union (ITU) in 1996 to promote compatibility in videoconference transmissions over IP networks.

H.323 is a standard approved by the International Telecommunication Union (ITU) in 1996 to promote compatibility in videoconference transmissions over IP networks. H.323 was originally

promoted as a way to provide consistency in audio, video and data packet transmissions in the

event that a local area network (LAN) did not provide guaranteed service quality (QoS). Although it was doubtful at first whether manufacturers would adopt H.323, it is now considered to be the standard for interoperability in audio, video and data transmissions as well as Internet phone and voice-over-IP (VoIP) because it addresses call control and management for both point-to-point and multipoint conferences as well as gateway administration of media traffic, bandwidth and user participation.

H.323, which describes how multimedia communications oc/cur between terminals, network equipment and services, is part of a larger group of ITU recommendations for multi-media interoperability called H.3x. The latest of these recommendations, H.248, is a recommendation to provide a single standard for the control of gateway devices in multi-media packet transmissions to allow calls to connect from a LAN to a Public Switched Telephone Network (PSTN), as well as to other standards-based terminals. This recommendation was announced in August 2000,

by the ITU-TU Study Group 16 and the Megaco Working Group of the Internet Engineering Task Force (IETF).

The Session Initiation Protocol (SIP) is an Internet Engineering Task Force (IETF) standard protocol for initiating an interactive user session that involves multimedia elements such as video, voice, chat, gaming, and virtual reality.

Like HTTP or SMTP, SIP works in the Application layer of the Open Systems Interconnection (OSI) communications model. The Application layer is the level responsible for ensuring that communication is possible. SIP can establish multimedia sessions or Internet telephony calls, and modify, or terminate them. The protocol can also invite participants to unicast or multicast sessions that do not necessarily involve the initiator. Because the SIP supports name mapping and redirection services, it makes it possible for users to initiate and receive communications and services from any location, and for networks to identify the users whereever they are.

SIP is a request-response protocol, dealing with requests from clients and responses from servers. Participants are identified by SIP URLs. Requests can be sent through any transport protocol, such as UDP, SCTP, or TCP. SIP determines the end system to be used for the session, the communication media and media parameters, and the called party's desire to engage in the communication. Once these are assured, SIP establishes call parameters at either end of the communication, and handles call transfer and termination.

The Session Initiation Protocol is specified in IETF Request for Comments [RFC] 2543.

PBXs, IP-PBXs and hybrid systems

A PBX is a telephone system that switches calls between enterprise users on local lines while allowing all users to share a certain number of external phone lines. The main purpose of a PBX is to save the cost of requiring a line for each user to the telephone company's central office. An IP PBX is a private branch exchange that switches calls between VoIP users on local lines while allowing all users to share a certain number of external phone lines. A typical IP PBX can also switch calls between a VoIP user and a traditional telephone user.

Many major vendors are offering hybrid voice systems that combine elements of VoIP and public switched telephone network (PSTN) systems. These new hybrid systems may give enterprises an opportunity to benefit from VoIP cost savings by combining the traditional telephony hardware with new technologies.

VoIP phone

A VoIP phone is a telephone set designed specifically for use in a voice over IP (VoIP) system by converting standard telephone audio into a digital format that can be transmitted over the Internet, and by converting incoming digital phone signals from the Internet to standard telephone audio. A VoIP phone allows the user to take advantage of VoIP technology without involving a personal computer, although an Internet connection is required.

Physically, a VoIP phone set resembles a traditional hard wired or cordless telephone set. It employs the familiar ear and mouth (E&M) arrangement with an earphone (or earpiece) for listening to incoming audio, and a microphone (or mouthpiece) for transmitting audio. Some VoIP phone sets offer enhanced quality audio, comparable to that on compact disc (CD). A few VoIP phone sets allow for the transmission and reception of image data during calls, so they can be considered video telephones.

What is the difference between H.323 and SIP?

H.323 and SIP both support VoIP and multimedia communications. H.323 is an older standard developed by the ITU. A good chunk of it is based on ISDN which comes from the traditional telephony world. H.323 is a binary protocol and is fairly complex in nature. SIP was developed by the Internet Engineering Task Force (IETF) and is text based (similar to HTTP). Much of the infrastructure already in place to support HTTP has been adapted to support SIP. IT managers within businesses are generally more comfortable with SIP because they are used to handling HTTP traffic. SIP is an open standard and solutions based on SIP are highly interoperable. A lot of effort has gone into ensuring interoperability and many manufacturers work together to regularly test to ensure this. Very few manufacturers are working on new H.323 implementations. SIP has become the standard of choice and is being worked on by large companies such as Microsoft and Cisco.

Getting to know VoIP regulations

Unfortunately, the biggest challenge to the total adoption of VoIP isn't the complex technology, or the lack of experienced implementers, or even compelling business cases -- it's regulatory hoopla. VoIP is simply illegal in an astonishing number of countries, while the rest are mired in enough red tape to often negate the cost savings.

Former chairman of the Federal Communications Commission, Michael Powell, strongly argues that new communications technologies should not be subjected to federal and state regulations.

However, there is staunch opposition from several states to regulate VoIP as a traditional telephone service, a move that many fear would stifle its potential.

We're keeping track of developments and will update this report as we learn more.

Advantages

VoIP technology uses the Internet's packet-switching capabilities to provide phone service. VoIP has several advantages over circuit switching. For example, packet switching allows several telephone calls to occupy the amount of space occupied by only one in a circuit-switched network. Using PSTN, that 10-minute phone call we talked about earlier consumed 10 full minutes of transmission time at a cost of 128 Kbps. With VoIP, that same call may have occupied only 3.5 minutes of transmission time at a cost of 64 Kbps, leaving another 64 Kbps free for that 3.5 minutes, plus an additional 128 Kbps for the remaining 6.5 minutes. Based on this simple estimate, another three or four calls could easily fit into the space used by a single call under the conventional system. And this example doesn't even factor in the use of data compression, which further reduces the size of each call.

Let's say that you and your friend both have service through a VoIP provider. You both have your analog phones hooked up to the service-provided ATAs. Let's take another look at that typical telephone call, but this time using VoIP over a packet-switched network:

1. You pick up the receiver, which sends a signal to the ATA. 2. The ATA receives the signal and sends a dial tone. This lets you know that you have a

connection to the Internet. 3. You dial the phone number of the party you wish to talk to. The tones are converted by

the ATA into digital data and temporarily stored. 4. The phone number data is sent in the form of a request to your VoIP company's call

processor. The call processor checks it to ensure that it is in a valid format.

5. The call processor determines to whom to map the phone number. In mapping, the phone number is translated to an IP address (more on this later). The soft switch connects the two devices on either end of the call. On the other end, a signal is sent to your friend's ATA, telling it to ask the connected phone to ring.

6. Once your friend picks up the phone, a session is established between your computer and your friend's computer. This means that each system knows to expect packets of data from the other system. In the middle, the normal Internet infrastructure handles the call as if it were e-mail or a Web page. Each system must use the same protocol to communicate. The systems implement two channels, one for each direction, as part of the session.

7. You talk for a period of time. During the conversation, your system and your friend's system transmit packets back and forth when there is data to be sent. The ATAs at each end translate these packets as they are received and convert them to the analog audio

signal that you hear. Your ATA also keeps the circuit open between itself and your analog phone while it forwards packets to and from the IP host at the other end.

8. You finish talking and hang up the receiver. 9. When you hang up, the circuit is closed between your phone and the ATA.

The ATA sends a signal to the soft switch connecting the call, terminating the session. Probably one of the most compelling advantages of packet switching is that data networks already understand the technology. By migrating to this technology, telephone networks immediately gain the ability to communicate the way computers d Audio conferencing using Voice over IP (VoIP) works very similarly to a traditional conference call using analog telephones. In a conference call, callers connect to a conference bridge, a server that allows multiple people to talk to one another. VoIP audio conferences use the same principle -- callers connect to a conference bridge via their telephones or computers.

The main difference involves how the data moves from one point to another. Traditional telephones use circuit switching. For circuit-switched conferencing, the telephone system routes calls through a series of interconnected switches until it reaches the conference bridge. The conference bridge then connects multiple calls to one another. If all of the callers are located in the same office building, their calls connect to the bridge through the private branch exchange (PBX), which is a miniature phone network within the office. Either way, all the connections stay open as long as the call continues. Usually, only one person talks at a time, so only parts of the connection are actually in use at any given time.

VoIP networks do not use circuit switching. Instead, they use packet switching. While circuit switching keeps the connection open and constant, packet switching opens a brief connection -- just long enough to send a small chunk of data, called a packet, from one system to another. Instead of traveling from switch to switch, the data travels across the Internet, usually following the most efficient path.

Conference calling capabilities are often built into VoIP networks or available as a service upgrade. If a business uses a VoIP network for its telephones, calls from outside of the company can still typically connect the conference bridges. External lines may connect directly to the bridge server, or external calls may reach the server through the VoIP network.

Securing VoIP

Deploying effective VoIP security is challenging. Since telephony traffic must travel over the IP network between gateways, stations, servers and proxies, there are plenty of places to attack. The list of possible threats

includes; toll fraud, impersonation, hijacking of calls, session replay, media tampering, denial of service and SPIT. This section will help you avoid some security pitfalls.

QoS

A big component of VoIP infrastructure is effectively deploying a Quality of Service (QoS) model. QoS is the idea that transmission rates, error rates and other characteristics can be measured, improved, and to some extent, guaranteed in advance. QoS is of particular concern for the continuous transmission of high-bandwidth video and multimedia information. There are many ways to ensure QoS. This section will help you decide which QoS method is right for you.

Top VoIP players .

Avaya Nortel Siemens NEC Mitel Inter-Tel 3Com Alcatel Astra Technologies (EADS) Ericcson Cisco

IP Telephony Products Evaluation

If you've decided to implement an IP telephony network, but you haven't decided on a vendor yet, you're probably planning to evaluate a lot of IP telephony hardware and software in the near future. You've got a lot of questions to answer. These include concerns about the hardware and software compatibility with your existing network, and of course, you want to listen to actual phone calls to hear how it sounds for yourself.

You also want to verify interoperability between a number of vendors and product lines.

All of this may be easier said than done. You get some demo equipment from your salesmen, but if you don't already have a network set up, what good is it? If you're trying to set up all the equipment at once and you have issues, it's hard to know which product is misconfigured or at fault. These and other issues can make evaluations almost pointless.

Avaya Definity G3si Version 9 PBX with Cisco CallManager using Cisco

VG200-T1

Network Topology

As shown in the diagram above, a Lucent/Avaya Definity G3si EPABX is connected via a T1-CAS link to a Cisco VG200 Gateway, which in turn, was connected to an Ethernet switch. The interoperability testing involved basic call setup functionality between the Cisco Call Manager and the Lucent/Avaya EPABX.

Technical details/ Features of ONGC EPABX Systems:

Two TATA make G-3 Si Definity series exchanges are installed one at Infocom, Chandkheda & other at Sabaramati campus for catering need of captive inter/intra region/field communication through TDMA/ICNET as well as external communication need through 2 PRI link from local BSNL. These two exchanges are interlinked through 2 MB HDSL on copper pair. Just to have an insight,

the brief description of technical features and capacity of IS Building, Chandkheda exchange is as follows:

Model & Make of Existing Exchange : G-3 Si Definity of TATA-Avaya

License for total port : 1742 port

Installed Capacity : 1350 extensions

Analog extension : 1302

Digital extensions : 48

Analog SLIC : 54

Digital SLIC : 02

Working Extensions : 890 extension

ICNET/Telemetry : 250 extensions

EPABX extensions : 640 extensions

Expansion capacity with existing hardware : 1742 ports

Overall Expansion capacity : 2700 ports

CO Junction Lines : 16 ( 8 I/C + 8 O/G)

Two PRI link from BSNL for Level DID : 60 Channels for both I/C +

O/G

2W/4W E&M circuits for tie line to ICNET : 32 circuits

Salient Features

Single exchange works as captive EPAX and Group EPABX

Using vertical partitioning in exchange.

Level DID on PRI Link enabling CLI on each extensions

Unified Messaging System for Voice, Fax, E-mail

Access to FAX, E-Mail through LAN node

Connected on LAN through LAN switch

Remote Maintenance through LAN as well as from outside

Compatible to VOIP, DECT, ATM, Q-Signaling etc.

All features of Convectional Digital Exchange.

AVAYA Definity at Chandkheda

HDSL Modem

ICNET linesE&M Closed No. Lines for Baroda, Cambay, Ankleshwar, Mehsana

2MB Link : NEL from BSNL

Voice Mail Server

Call Billing & Maintenance PC

Satellite Network of ONGC

Microwave Network of ONGC

BSNL PRI LINKS

BSNL

CO & LD-DID

Vert

ical P

art

itio

n

Vert

ical P

art

itio

n

HDSL Modem

Network Diagram of ONGC Exchanges at Ahmedabad Asset: