ACKNOWLEDGEMENT

I would like to express our profound sense of gratitude to Mahanagar Telephone

Nigam Limited (M.T.N.L.) for giving us the opportunity to carry out this Training.

We express our feeling of thankfulness to MR. H P GUPTA(Sr. Mgr. (A)BB-o) for

giving his constant encouragement and every facility that we required during our Training

We express our deep sense of gratitude to our Training Guide Mr. Aashish Singh for
his valuable guidance. We are thankful to him for his gentle encouragement and pains he
took in while guiding us through the training.
 History of MTNL

1986

- On 28th February Mahanagar Telephone Nigam Ltd. was incorporated as a Public Limited Company under
the Companies Act, 1956. The company has been set up to take over the management, control and operation
of Delhi Telephone District (Excluding public telegraph service) and Mumbai Telephone District of the
Department of Telecommunications and to plan, establish, develop, provide, operate and maintain all types
of telecommunication services including Telephone, telex, wireless, data communication, telematic and other
like forms of communication.

- One of the important objectives of the company is to raise the necessary finance to meet its own
developmental needs and also that of the telecommunications board of the Department of
Telecommunication.

- The Main objectives and aims of Nigam are as follows: (1) To upgrade the quality of telecom services (2)
To expand telecom (3) To raise necessary financial resources (4) To provide new telecommunication
services, particularly needed by the business community and public administration such as Cellular Mobile
Radio Telephone, Radio Paging, Fascimile Videotex, Teletex, Electronic Mail etc. (5) To invest in Human
Resource Development (6) To organise and dovetail the training programmes.

- All shares issued to President of India and of which 59,99,984 shares were issued without payment in
cash.

1995

- The entire junction network in the system was fibre based on a step to provide self healing machanisms to
be built in the System through induction of SDH technology.

1997

- The Company took various steps to provide a host of value added services like datacom, inet, DIDPABX,
Voice Mail, Radio Paging and ISDN.

- In addition to phone plus facilities like dynamic locking, call waiting/call transfer, hot line etc. were
extended to valued customers. Apart from this IVRS (Interactive Voice Response System) like local
assistance changed number information, and falut booking system ensuring round the clock service, a CD-
ROM version of the telephone directory and an on-line directory enquiry through PC was introduced during
the year.

- The Company undertook to provide, wireless in the local loop in Mumbai and Delhi, GSM Mobile

- Payment of bills was made hassel free with the introduction of electronic clearing system and customer
service management system for on-line payment and adjustment of telephone bills.

1998

- As on 31st March, the waiting list of the MTNL has 1047 in Delhi and Nil in Mumbai due to its sustained
efforts and timely implementation of various projects.

- MTNL has proposed that a WiLL subscriber's bank can stand guarantee for this security, effectively
ensuring that there is no real outflow from the subscriber's pockets.

- The Mahanagar Telephone Nigam Ltd (MTNL) board has approved a proposal for setting up a joint venture
company with the Telecom Consultants India Ltd (TCIL) for operating basic and cellular services.

- The Mahanagar Telephone Nigam Ltd (MTNL) is developing a software to protect customers from
telephone tapping and reading by external sources.

- Mahanagar Telephone Nigam Ltd (MTNL) launched the country's first toll-free service in Delhi.

- MTNL will set up an integrated commercial accounting system with help from Tata Consultancy,
Services.

- Within a few weeks of listing, the MTNL (Mahanagar Telephone Nigam Ltd) GDR issue was voted as one
of the top 10 Best International Equity Issue in 1997 in a poll conducted by Euroweek's International Equity
Review, the respected international business publication.

- In the polls, MTNL secured the awards of the 10th best international equity issue of 1997; fourth best Asian
equity issue of 1997; and second best Indian equity issue of 1997.

- State-owned Mahanagar Telephone Nigam Ltd is expanding the wireless in local loop (WLL) network,
both in Mumbai and Delhi up to 50,000 and 10,000 subscribers respectively. In Delhi, they are expanding the
network to 10,000 from the existing 1,000 subscribers. In Mumbai the network would be expanded to 50,000
subscriber.

- 2000

- Mahanagar Telephone Nigam Ltd (MTNL) tied up with MasterCard International to pioneer the acceptance
of credit cards for payment of telephone bills.

- The Company will do the soft launch of its GSM-based mobile services in Delhi and Mumbai.

- Narendra Sharma has been appointed as the new chairman and managing director of the company.

- Mahanagar Telephone Nigam Ltd. has transferred its Internet Service Provider Category-A licence as well
as Internet business to its subsidiary, Millennium Telecom.
- Dolphin, the much-awaited cellular service of MTNL, was launched in Delhi on 6th Feb.

2002

-Announces its launch of Wireless in Local Loop (WLL) service in Mumbai

-Launches its pre-paid cellular card "Trump" with tariffs 50 per cent lower than that of the private players in
Delhi and Mumbai

2006

-NLD licence for MTNL

-MTNL teams up with Aksh Opticfibre for IPTV

-MTNL & Software Technology Parks signs JV agreement

2009

-MTNL to launch 3G Services in Mumbai

A Brief Introduction to MTNL

MTNL was set up on 1st April, 1986 by government of India to upgrade the quality of
telecom services, Expand the telecom network, introduce new services and to raise revenue
of telecom department needs of India’s key metros-Delhi, the political capital and Mumbai,
the business capital of India. In the past 17 year, the company has taken rapid strides to
emerge as India’s leading and Asia’s largest telecom operating companies. Besides having a
strong financial base, MTNL has achieved a market share of approximately 13% of Indian
telecommunication network with a customer base of over 4.98 million lines.
The company has been setup to provide basic, mobile, internet data, and value added services

1 Basic telephony
1. PSTN
2. ISDN

2 Garuda Mobile Service

3 Dolphin Cellular Mobil Service
4 Internet
5 Data Circuits
6 Value Added Services
The company has also been in the forefront of technology induction by converting 100% of
its telephone exchange network into the state-of-art digital mode.

The government of India currently holds 56.25% stake in the company.In the year 2003-04,
the company would not only consolidate the gains but would also focus on new areas of
enterprise viz. joint venture for project outside India, entering into national long distance
operation, widening the cellular and CDMA-based WLL customer based setting up internet
and allied services on all India basis

Packet Switching

Switching (as in switched connections) refers to finding a path for data transmission across a
number of potential links between sender and receiver. On the other hand, analog and digital
connections require a fixed connection to exist, at least for the duration of each
communication session. Switching methods include both circuit switching and packet
switching. Essentially, when data is received on an incoming line, the switching device must
find an appropriate outgoing line on which to forward it. These switching devices are usually
called routers, based on the functions they perform.

Data package is broken into packers and each package is tagged with a destination address
and other info.

Advantages of networks

1. Resource sharing: - Resource sharing makes all the programs, equipments and
especially data available to anyone on the network without regard to the physical
location of the resource and the user.
2. Scalability: - The ability to increase system performance gradually as the workload
grows just by adding more processors.
3. A computer network can provide a powerful communication medium among widely
separated employees.
4. High reliability
5. Saving money

Local Area Network (LAN)

LANs are privately owned networks within a single building or campus of few kilometers
in size.
 LANs are distinguished from other kind of networks by three characters tics: (1) their
size, (2) their transmission technology, (3) their topology.

LANs are restricted in size, which means that the worst-case transmission time is
bounded and known in advance. Traditional LANs run at speeds of 10 to 100Mbps, have
low delay and make few errors.

Metropolitan Area Network (MAN)

A MAN is basically a bigger version of a LAN and normally uses similar technology. It
might cover a group of nearby corporate offices or a city and might be either private or
public.

The main reason of even distinguishing MANs as a special category ia that standard has been
adopted for them, and this standard is now being implemented. It is called DQDB
(Distributes Queue Dual Bus).

Wide Area Network (WAN)

A WAN, spans a large geographical area, often a country or a continent. It contains a

collection of machines intended for running user programs

In most WAN, the subnet consists of two distinct components: transmission line and
switching elements. Transmission line moves bits between machines.

The switching elements are specialized computers used to connect two or more transmission
lines. When data arrive on an incoming line, the switching elements must choose an outgoing
line to forward them on.
Bus topology

With the Bus topology, all workstations are connecting directly to the main backbone that
carries the data. Traffic generated by any computer will travel across the backbone and be
received by all workstations. This works well in a small network of 2-5 computers, but as the
numbers of computers increases so will the network traffic and this can greatly decrease the
performance and available bandwidth of your network.

If

the bus is damaged anywhere in its path, then it will most certainly cause the network to stop
working or, at the very least, cause big communication problems between the workstations.

More about Bus Topology

• Bus consists of a single linear cable called a trunk.

• Data is sent to all computers on the trunk. Each computer examines EVERY packet
on the wire to determine who the packet is for and accepts only messages addressed
to them.
• Bus is a passive topology.
• Performance degrades as more computers are added to the bus.
• Signal bounce is eliminated by a terminator at each end of the bus.
• Barrel connectors can be used to lengthen cable.
• Repeaters can be used to regenerate signals.
• good for a temporary, small network
The Physical HUB or STAR Topology

The Star or Hub topology is one of the most common network topologies found in most
offices and home networks. It has become very popular in contrast to the bus type, because of
the cost and the ease of troubleshooting.

The advantage of the star topology is that if one computer on the star topology fails,
then only the failed computer is unable to send or receive data. The remainder of the
network functions normally.

The disadvantage of using this topology is that because each computer is connected to a
central hub or switch, if this device fails, the entire network fails!

More about Star Topology

• Computers are connected by cable segments to a centralized hub.

• Signal travels through the hub to all other computers.
• Requires more cable.
• If hub goes down, entire network goes down.
• most scalable and reconfigurable of all topologies
• If a computer goes down, the network functions normally.
The Physical Ring Topology

In the ring topology, computers are connected on a single circle of cable. Unlike the bus
topology, there are no terminated ends. The signals travel around the loop in one direction
and pass through each computer, which acts as a repeater to boost the signal and send it to
the next computer.

The method by which the data is transmitted around the ring is called token passing. A token
is a special series of bits that contains control information. Possession of the token allows a
network device to transmit data to the network. Each network has only one token.

More about Ring Topology

• Computers are connected on a single circle of cable.

• usually seen in a Token Ring network
• Each computer acts as a repeater and keeps the signal strong
• No termination required
• Token passing is used in Token Ring networks. The token is passed from one
computer to the next, only the computer with the token can transmit. The receiving
computer strips the data from the token and sends the token back to the sending
computer with an acknowledgment. After verification, the token is regenerated.
• relatively easy to install, requiring ;minimal hardware
The Physical Mesh Topology

In a mesh topology, each computer is connected to every other computer by a separate cable.
This configuration provides redundant paths through the new work, so if one computer blows
up, you don't lose the network

Again, the big advantage of this topology is its backup capabilities by providing multiple
paths through the network.

More about Mesh Topology

• The mesh topology connects each computer on the network to the others
• Meshes use a significantly larger amount of network cabling than do the other
network topologies, which makes it more expensive.
• The mesh topology is highly fault tolerant.
The Physical Hybrid Topology

With the hybrid topology, two or more topologies are combined to form a complete network.

Star-Bus

In a star-bus topology, several star topology networks are linked to a bus connection. In this
topology, if a computer fails, it will not affect the rest of the network. However, if the central
component, or hub, that attaches all computers in a star, fails, then you have big problems
since no computer will be able to communicate.

More about Star Bus Topology

• Several star topologies linked with a linear bus.

• No single computer can take the whole network down. If a single hub fails, only the
computers and hubs connected to that hub are affected.
 Star-Ring

In the Star-Ring topology, the computers are connected to a central component as in a star
network. These components, however, are wired to form a ring network.

Like the star-bus topology, if a single computer fails, it will not affect the rest of the network.
By using token passing, each computer in a star-ring topology has an equal chance of
communicating.

More about Star Ring Topology

• Also known as star wired ring because the hub itself is wired as a ring. This means it's
a physical star, but a logical ring.
• This topology is popular for Token Ring networks because it is easier to implement
than a physical ring, but it still provides the token passing capabilities of a physical
ring inside the hub.
• Just like in the ring topology, computers are given equal access to the network media
through
• the passing of the token.
• A single computer failure cannot stop the entire network, but if the hub fails, the ring
that the hub controls also fails.

Voice Over Internet Protocol (VOIP)

Voice over Internet Protocol (VoIP, Voice over IP) is a general term for a family of
methodologies, communication protocol, and transmission technologies for delivery of voice
communications and multimedia sessions over internet protocol (IP) networks, such as the
internet. Other terms frequently encountered and synonymous with VoIP are IP
telephony, Internet telephony, voice over broadband (VoBB),broadband telephony,
and broadband phone.

Internet telephony refers to communications services — voice, facsimile, and/or voice-

messaging applications — that are transported via the Internet, rather than the Public
Switched Transfer Protocol (PSTN). The basic steps involved in originating an Internet
telephone call are conversion of the analog voice signal to digital format and
compression/translation of the signal into internet protocol (IP) packets for transmission over
the Internet; the process is reversed at the receiving end.

VoIP systems employ session control protocols to control the set-up and tear-down of calls as
well as audio codes which encode speech allowing transmission over an IP network as digital
audio via an audio stream. Codec use is varied between different implementations of VoIP
(and often a range of codecs are used); some implementations rely on narrowband
and compressed speech, while others support high fidelity stereo codecs.
Protocols

Voice over IP has been implemented in various ways using both proprietary and open
protocols and standards. Examples of technologies used to implement Voice over IP include:

 H.323
 IP Multimedia Subsystem (IMS)
 Media Gateway Control Protocol (MGCP)
 Session Initiation Protocol (SIP)
 Real-time Transport Protocol (RTP)
 Session Description Protocol (SDP)

The H.323 protocol was the one of the first VoIP protocols that found wide-spread
implementation for long-distance traffic, as well as local area network services. However,
since the development of newer, less complex protocols, such as MGCP and SIP, H.323
deployments are increasingly limited to carrying existing long-haul network traffic. In
particular, the Session Initiation Protocol (SIP) has gained widespread VoIP market
penetration.
Adoption

Consumer market

Example of residential network including VoIP

A major development starting in 2004 has been the introduction of mass-market VoIP
services that require existing broadband Internet Access by which subscribers place and
receive telephone calls in much the same manner as they would via the Public Switched
Transfer Protocol (PSTN). Full-service VoIP phone companies provide inbound and
outbound service with Direct Inbound Dialing. Many offer unlimited domestic calling for a
flat monthly subscription fee. This sometimes includes international calls to certain countries.
Phone calls between subscribers of the same provider are usually free when flat-fee service is
not available.
PSTN and mobile network providers
It is becoming increasingly common for telecommunications providers to use VoIP
telephony over dedicated and public IP networks to connect switching stations and to
interconnect with other telephony network providers; this is often referred to as "IP”

"Dual mode" telephone sets, which allow for the seamless handover between a cellular
network and a Wi-Fi network, are expected to help VoIP become more popular.

Phones such as the NEC N900iL, many of the Nokia E-series and several other Wi-
Fi enabled mobile phones have SIP clients built into the firmware. Such clients operate
independently of the mobile phone network (however some operators choose to remove the
client from subsidised handsets). Some operators such as Vodafone actively try to block
VoIP traffic from their network. Others, like T-Mobile, have refused to interconnect with
VoIP-enabled networks as was seen in the legal case between T-Mobile and Truphone, which
ultimately was settled in the UK High Court in favour of the VoIP carrier.
Corporate use
Because of the bandwidth efficiency and low costs that VoIP technology can provide,
businesses are gradually beginning to migrate from traditional copper-wire telephone systems
to VoIP systems to reduce their monthly phone costs.

VoIP solutions aimed at businesses have evolved into "unified communications" services that
treat all communications—phone calls, faxes, voice mail, e-mail, Web conferences and more
—as discrete units that can all be delivered via any means and to any handset, including
cellphones. Two kinds of competitors are competing in this space: one set is focused on VoIP
for medium to large enterprises, while another is targeting the small-to-medium business
(SMB) market.

VoIP runs both voice and data communications over a single network, which can
significantly reduce infrastructure costs.

The prices of extensions on VoIP are lower than for PBXs and key systems. VoIP switches
run on commodity hardware, such as PCs or Linux systems. Rather than closed architectures,
these devices rely on standard interfaces.

VoIP devices have simple, intuitive user interfaces, so users can often make simple system
configuration changes. Dual-mode cellphones enable users to continue their conversations as
they move between an outside cellular service and an internal Wi-Fi network, so that it is no
longer necessary to carry both a desktop phone and a cellphone. Maintenance becomes
simpler as there are fewer devices to oversee.

Skype, which originally marketed itself as a service among friends, has begun to cater to
businesses, providing free-of-charge connection between any users on the Skype network and
connecting to and from ordinary PSTN telephones for a charge.

In the United States the Social Security Administration (SSA) is converting its field offices
of 63,000 workers from traditional phone installations to a VoIP infrastructure carried over
its existing data network.
Benefits

Operational cost
VoIP can be a benefit for reducing communication and infrastructure costs. Examples
include:

 Routing phone calls over existing data networks to avoid the need for separate voice
and data networks.
  Conference calling, IVR, call forwarding, automatic redial, and caller ID features that
traditional telecommunication companies normally charge extra for are available free
of charge from open source VoIP implementations.

Flexibility
VoIP can facilitate tasks and provide services that may be more difficult to implement using
the PSTN. Examples include:

 The ability to transmit more than one telephone call over a single broadband
connection.
 Secure calls using standardized protocols (such as Secure Real-time Transport
Protocol). Most of the difficulties of creating a secure telephone connection over
traditional phone lines, such as digitizing and digital transmission, are already in
place with VoIP. It is only necessary to encryt and authenticate the existing data
stream.
 Location independence. Only a sufficiently fast and stable Internet connection is
needed to get a connection from anywhere to a VoIP provider.
 Integration with other services available over the Internet, including video
conversation, message or data file exchange during the conversation, audio
conferencing, managing address books, and passing information about whether other
people are available to interested parties.

Challenges
Quality of service

Communication on the IP network is inherently less reliable in contrast to the circuit-

switched public telephone network, as it does not provide a network-based mechanism to
ensure that data packets are not lost, or delivered in sequential order. It is a best-effort
network without fundamental Quality of service (QoS) guarantees. Therefore, VoIP
implementations may face problems mitigating latency and jittler.

By default, IP routers handle traffic on a first-come, first-served basis. Routers on high

volume traffic links may introduce latency that exceeds permissible thresholds for VoIP.
Fixed delays cannot be controlled, as they are caused by the physical distance the packets
travel, however latency can be minimized by marking voice packets as being delay-sensitive
with methods such as DiffServ
A VoIP packet usually has to wait for the current packet to finish transmission, although it is
possible to pre-empt (abort) a less important packet in mid-transmission, this is not
commonly done, especially on high-speed links where transmission times are short even for
maximum-sized packets. An alternative to pre-emption on slower links, such as dialup and
DSL, is to reduce the maximum transmission time by reducing the maximum transmission
unit. But every packet must contain protocol headers, so this increases relative header
overhead on every link along the user's Internet paths, not just the bottleneck (usually
Internet access) link.

ADSL modems provide Ethernet (or Ethernet over USB) connections to local equipment, but
inside they are actually ATM modems. They useAAL5 to segment each Ethernet packet into
a series of 48-byte ATM cells for transmission and reassemble them back into Ethernet
packets at the receiver. A virtual circuit identifier (VCI) is part of the 5-byte header on every
ATM cell, so the transmitter can multiplex the active virtual circuits (VCs) in any arbitrary
order. Cells from the same VC are always sent sequentially.

However, the great majority of DSL providers use only one VC for each customer, even
those with bundled VoIP service. Every Ethernet packet must be completely transmitted
before another can begin. If a second PVC were established, given high priority and reserved
for VoIP, then a low priority data packet could be suspended in mid-transmission and a VoIP
packet sent right away on the high priority VC. Then the link would pick up the low priority
VC where it left off. Because ATM links are multiplexed on a cell-by-cell basis, a high
priority packet would have to wait at most 53 byte times to begin transmission. There would
be no need to reduce the interface MTU and accept the resulting increase in higher layer
protocol overhead, and no need to abort a low priority packet and resend it later.

This doesn't come for free. ATM has substantial header overhead: 5/53 = 9.4%, roughly
twice the total header overhead of a 1500 byte TCP/IP/Ethernet packet (with TCP
timestamps). This "ATM tax" is incurred by every DSL user whether or not he takes
advantage of multiple virtual circuits - and few can.
Layer-2 quality of service
A number of protocols that deal with the data link layer and physical layer include quality-of-
service mechanisms that can be used to ensure that applications like VoIP work well even in
congested scenarios. Some examples include:

 IEEE 802.11e is an approved amendment to the IEEE 802.11 standard that defines a
set of quality-of-service enhancements for wireless LAN applications through
 modifications to the Media Access Control(MAC) layer. The standard is considered
of critical importance for delay-sensitive applications, such as Voice over Wireless
IP.
 IEEE 802.1p defines 8 different classes of service (including one dedicated to voice)
for traffic on layer-2 wired Ethernet.
 The ITU-T G.hn standard, which provides a way to create a high-speed (up to 1
gigabit per second) Local area network using existing home wiring (power lines,
phone lines and coaxial cables). G.hn provides QoS by means of "Contention-Free
Transmission Opportunities" (CFTXOPs) which are allocated to flows (such as a
VoIP call) which require QoS and which have negotiated a "contract" with the
network controller.

Susceptibility to power failure

Telephones for traditional residential analog service are usually connected directly to
telephone company phone lines which provide direct current to power most basic analog
handsets independently of locally available power.

IP Phones and VoIP telephone adapters connect to routers or cable modems which typically
depend on the availability of mains electricity or locally generated power. Some VoIP service
providers use customer premise equipment (e.g., cablemodems) with battery-backed power
supplies to assure uninterrupted service for up to several hours in case of local power
failures. Such battery-backed devices typically are designed for use with analog handsets.

The susceptibility of phone service to power failures is a common problem even with
traditional analog service in areas where many customers purchase modern handset units that
operate wirelessly to a base station, or that have other modern phone features, such as built-in
voicemail or phone book features.
Emergency calls
The nature of IP makes it difficult to locate network users geographically. Emergency calls,
therefore, cannot easily be routed to a nearby call center. Sometimes, VoIP systems may
route emergency calls to a non-emergency phone line at the intended department. In the
United States, at least one major police department has strongly objected to this practice as
potentially endangering the public.

A fixed line phone has a direct relationship between a telephone number and a physical
location. A telephone number represents one pair of wires that links a location to the
telephone company's exchange. Once a line is connected, the telephone company stores the
home address that relates to the wires, and this relationship will rarely change. If an
emergency call comes from that number, then the physical location is known.

In the IP world, it is not so simple. A broadband provider may know the location where the
wires terminate, but this does not necessarily allow the mapping of an IP address to that
location. IP addresses are often dynamically assigned, so the ISP may allocate an address for
online access, or at the time a broadband router is engaged. The ISP recognizes individual IP
addresses, but does not necessarily know what physical location to which it corresponds. The
broadband service provider knows the physical location, but is not necessarily tracking the IP
addresses in use.

There are more complications, since IP allows a great deal of mobility. For example, a
broadband connection can be used to dial a virtual private network that is employer-owned.
When this is done, the IP address being used will belong to the range of the employer, rather
than the address of the ISP, so this could be many kilometres away or even in another
country. To provide another example: if mobile data is used, e.g., a 3G mobile handset or
USB wireless broadband adapter, then the IP address has no relationship with any physical
location, since a mobile user could be anywhere that there is network coverage, even roaming
via another cellular company.

In short, there is no relationship between IP address and physical location, so the address
itself reveals no useful information for the emergency services.

At the VoIP level, a phone or gateway may identify itself with a SIP registrar by using a
username and password. So in this case, the Internet Telephony Service Provider
(ITSP)knows that a particular user is online, and can relate a specific telephone number to the
user. However, it does not recognize how that IP traffic was engaged. Since the IP address
itself does not necessarily provide location information presently, today a "best efforts"
approach is to use an available database to find that user and the physical address the user
chose to associate with that telephone number—clearly an imperfect solution.

VoIP Enhanced 911 (E911) is another method by which VoIP providers in the United States
are able to support emergency services. The VoIP E911 emergency-calling system associates
a physical address with the calling party's telephone number as required by the Wireless
Communications and Public Safety Act of 1999. All "interconnected" VoIP providers (those
that provide access to the PSTN system) are required to have E911 available to their
customers. VoIP E911 service generally adds an additional monthly fee to the subscriber's
service per line, similar to analog phone service. Participation in E911 is not required and
customers can opt-out or disable E911 service on their VoIP lines, if desired. VoIP E911 has
been successfully used by many VoIP providers to provide physical address information to
emergency service operators.

One shortcoming of VoIP E911 is that the emergency system is based on a static table
lookup. Unlike in cellular phones, where the location of an E911 call can be traced
using Assisted GPS or other methods, the VoIP E911 information is only accurate so long as
subscribers are diligent in keeping their emergency address information up-to-date. In the
United States, the Wireless Communications and Public Safety Act of 1999 leaves the burden
of responsibility upon the subscribers and not the service providers to keep their emergency
information up to date.
Lack of redundancy
With the current separation of the Internet and the PSTN, a certain amount of redundancy is
provided. An Internet outage does not necessarily mean that a voice communication outage
will occur simultaneously, allowing individuals to call for emergency services and many
businesses to continue to operate normally. In situations where telephone services become
completely reliant on the Internet infrastructure, a single-point failure can isolate
communities from all communication, including Enhanced 911 and equivalent services in
other locales.
Number portability
Local number portability (LNP) and Mobile number portability (MNP) also impact VoIP
business. In November 2007, the Federal Communications Commission in the United States
released an order extending number portability obligations to interconnected VoIP providers
and carriers that support VoIP providers. Number portability is a service that allows a
subscriber to select a new telephone carrier without requiring a new number to be issued.
Typically, it is the responsibility of the former carrier to "map" the old number to the
undisclosed number assigned by the new carrier. This is achieved by maintaining a database
of numbers. A dialed number is initially received by the original carrier and quickly rerouted
to the new carrier. Multiple porting references must be maintained even if the subscriber
returns to the original carrier. The FCC mandates carrier compliance with these consumer-
protection stipulations.

A voice call originating in the VoIP environment also faces challenges to reach its
destination if the number is routed to a mobile phone number on a traditional mobile carrier.
VoIP has been identified in the past as a Least Cost Routing (LCR) system, which is based
on checking the destination of each telephone call as it is made, and then sending the call via
the network that will cost the customer the least. This rating is subject to some debate given
the complexity of call routing created by number portability. With GSM number portability
now in place, LCR providers can no longer rely on using the network root prefix to
determine how to route a call. Instead, they must now determine the actual network of every
number before routing the call.

Therefore, VoIP solutions also need to handle MNP when routing a voice call. In countries
without a central database, like the UK, it might be necessary to query the GSM network
about which home network a mobile phone number belongs to. As the popularity of VoIP
increases in the enterprise markets because of least cost routing options, it needs to provide a
certain level of reliability when handling calls.

MNP checks are important to assure that this quality of service is met. By handling MNP
lookups before routing a call and by assuring that the voice call will actually work, VoIP
service providers are able to offer business subscribers the level of reliability they require.
PSTN integration
E.164 is a global numbering standard for both the PSTN and PLMN. Most VoIP
implementations support E.164 to allow calls to be routed to and from VoIP subscribers and
the PSTN/PLMN. VoIP implementations can also allow other identification techniques to be
used. For example, Skype allows subscribers to choose "Skype names"[ (usernames) whereas
SIP implementations can use URIs similar toemail addresses. Often VoIP implementations
employ methods of translating non-E.164 identifiers to E.164 numbers and vice-versa, such
as the Skype-In service provided by Skype and the ENUMservice in IMS and SIP.

Echo can also be an issue for PSTN integration. Common causes of echo include impedance
mismatches in analog circuitry and acoustic coupling of the transmit and receive signal at the
receiving end.
Securing VoIP
To prevent the above security concerns the government and military organizations are using;
Voice over Secure IP (VoSIP), Secure Voice over IP (SVoIP), and Secure Voice over Secure
IP (SVoSIP) to protect confidential, and/or classified VoIP communications. Secure Voice
over IP is accomplished by encrypting VoIP with Type 1 encryption. Secure Voice over
Secure IP is accomplished by using Type 1 encryption on a classified network,
like SIPRNet Public Secure VoIP is also available with free GNU programs.
Caller ID
Caller ID support among VoIP providers varies, although the majority of VoIP providers
now offer full Caller ID with name on outgoing calls.
In a few cases, VoIP providers may allow a caller to spoof the Caller ID information,
potentially making calls appear as though they are from a number that does not belong to the
caller Business grade VoIP equipment and software often makes it easy to modify caller ID
information. Although this can provide many businesses great flexibility, it is also open to
abuse.

The "Truth in Caller ID Act" has been in preparation in the US Congress since 2006, but as
of January 2009 still has not been enacted. This bill proposes to make it a crime in the United
States to "knowingly transmit misleading or inaccurate caller identification information with
the intent to defraud, cause harm, or wrongfully obtain anything of value ..."
Compatibility with traditional analog telephone sets
Some analog telephone adapters do not decode pulse dialing from older phones. They may
only work with push-button telephones using thetouch-tone system. The VoIP user may use a
pulse-to-tone converter, if needed.
Support for other telephony devices
Another challenge for VoIP implementations is the proper handling of outgoing calls from
other telephony devices such as Digital Video RecordersDVR boxes, satellite
television receivers, alarm systems, conventional modems and other similar devices that
depend on access to a PSTN telephone line for some or all of their functionality.

Internet protocol Television(IPTV)

Internet Protocol television (IPTV) is a system through which Internet television services are
delivered using the architecture and networking methods of the internet protocol suite over a
packet-switched network infrastructure, e.g., the internet and broadband Internet access
networks, instead of being delivered through traditional radio
frequency broadcast, satellite signal, and cable television (CATV) formats.

IPTV services may be classified into three main groups: live television, time-shifted
programming, and video on demand. It is distinguished from general Internet-based or web-
based multimedia services by its on-going standardization process and preferential
deployment scenarios in subscriber-based telecommunications networks with high-speed
access channels into end-user premises via set-top boxes or other customer premises
equipment
Future
In the past, this technology has been restricted by low broadband penetration and by the
relatively high cost of installing wiring capable of transporting IPTV content reliably in the
customer's home. In the coming years, however, residential IPTV is expected to grow at a
brisk pace as broadband was available to more than 200 million households worldwide in the
year 2005, projected to grow to 400 million by the year 2010. Many of the world's major
telecommunications providers are exploring IPTV as a new revenue opportunity from their
existing markets and as a defensive measure against encroachment from more conventional
Cable Television services.

Also, there is a growing number of IPTV installations within schools, universities,

corporations and local institutions.
Markets

Map of IPTV countries of the world. Countries where IPTV is available in at least some
parts of the country
The number of global IPTV subscribers is expected to grow from 28 million in 2009 to 83
million in 2013. Europe and Asia are the leading territories in terms of the over-all number of
subscriber. But in terms of service revenues, Europe and North America generate a larger
share of global revenue, due to very low average revenue per user (ARPU) in China and
India, the fastest growing (and ultimately, the biggest markets) in Asia. The global IPTV
market revenues are forecasted to grow from US$12 billion in 2009 to US$38 billion in
2013.
Discussion of IPTV

A telco IPTV service is usually delivered over a investment heavy walled garden network,
which is carefully engineered to ensure bandwidth efficient delivery of vast amounts of
multicast video traffic. The higher network quality also enables easy delivery of high quality
SD or HD TV content to subscribers’ homes. This makes IPTV by default the preferred
delivery platform for premium content. However, the investment for a telco to build an end-
to-end telco IPTV service can be substantial.

Local IPTV, as used by businesses for audio visual AV distribution on their company
networks is typically based on a mixture of:

1. Conventional TV reception equipment and IPTV encoders

2. IPTV Gateways that take broadcast MPEG channels and IP wrap them to
create multicast streams.

IPTV via satellite

Although IPTV and conventional satellite TV distribution have been seen as complementary
technologies, they are likely to be increasingly used together in hybrid IPTV networks that
deliver the highest levels of performance and reliability. IPTV is largely neutral to the
transmission medium, and IP traffic is already routinely carried by satellite for internet
backbone trunking and corporate VSAT networks. The use of satellite to carry IP is
fundamental to overcoming the greatest shortcoming of IPTV over terrestrial cables – the
speed/bandwidth of the connection.

The copper twisted pair cabling that forms the last mile of the telephone/broadband network
in many countries is not able to provide a sizeable proportion of the population with an IPTV
service that matches even existing terrestrial or satellite digital TV distribution. For a
competitive multi-channel TV service, a connection speed of 20Mbit/s is likely to be
required, but unavailable to most potential customers. The increasing popularity of high
definition TV (with twice the data of SD video) increases connection speed requirements, or
limits IPTV service quality and connection eligibility even further.

However, satellites are capable of delivering in excess of 100Gbit/s via multi-spot beam
technologies, making satellite a clear emerging technology for implementing IPTV networks.
Satellite distribution can be included in an IPTV network architecture in several ways.
Simplest to implement is an IPTV-DTH architecture, in which hybrid DVB/broadband set-
top boxes in subscriber homes integrate satellite and IP reception to give near-infinite
bandwidth with return channel capabilities. In such a system, many live TV channels may be
multicast via satellite (IP-encapsulated or as conventional DVB digital TV) with stored
video-on-demand transmission via the broadband connection.Argiva’s Satellite Media
Solutions Division suggests “IPTV works best in a hybrid format. For example, you would
use broadband to receive some content and satellite to receive other, such as live channels”.
Advantages

The IP-based platform offers significant advantages, including the ability to integrate
television with other IP-based services like high speed Internet access and VoIP.

A switched IP network also allows for the delivery of significantly more content and
functionality. In a typical TV or satellite network, using broadcast video technology, all the
content constantly flows downstream to each customer, and the customer switches the
content at the set-top box. The customer can select from as many choices as the telecomms,
cable or satellite company can stuff into the “pipe” flowing into the home. A switched IP
network works differently. Content remains in the network, and only the content the
customer selects is sent into the customer’s home. That frees up bandwidth, and the
customer’s choice is less restricted by the size of the “pipe” into the home. This also implies
that the customer's privacy could be compromised to a greater extent than is possible with
traditional TV or satellite networks. It may also provide a means to hack into, or at least
disrupt the private network.
IPTV charges by MTNL.
 Commercial Instructions on IPTV Services (Internet Protocol TV) service in Delhi

MTNL Delhi announces commercial launch of IPTV service. IPTV was soft launched on 17-10-
2006. Details of Services and tariff are given below:

(a) Registration charges: Rs. 500/- (To be adjusted in first telephone bill after (Non-
refundable) provision of service.)

(b) For security and monthly service charges for the set Top Box (STB), three options are
available:

Life time Yearly Quarterly

Rental(Rs.) Security (Rs.) Rental (Rs.) Security (Rs.) Rental (Rs.) Security (Rs.)
3,000/-
NIL 860/- 1,000 250/- 2000

(c) Monthly charges for TV channels: Four options are available:

Rs 60 p.m. Rs 130 p.m. Rs 200 p.m. Rs 250 p.m.

30 FTA 30 FTA30 FTA30 FTA Channels +all
Channels Channels +any 1Channels +any 23 bouquets of Sony,
only bouquet of Sony, Starbouquets of Sony, StarStar & Zee
or Zee or Zee

(d) Monthly charges for TSTV channels: Two options are available:

Rs 30 p.m. for 30 FTA TSTV Rs 45 p.m. additional charges for TSTV per
channels only bouquet of SONY, STAR & ZEE

(e) Monthly charges for VoD:

1. Rs.30 p.m. as subscription charge + 10 all time hit movies free p.m.

2. Other VOD content will be divided in 5 categories. Charge per movie is as follows:
Category A: Rs10; Category B: Rs20; Category C: Rs30 & Category D: Rs100 and premium of
movie : Rs200/-

Provision of IPTV for subscribers having high-speed Internet connection

While providing IPTV connection on existing high-speed broadband line, Internet speed is
reduced from 2mbps to 256 kbps. If a subscriber is not satisfied with 256 kbps speed at any time of
service and demands for higher Internet speed, his request may be accepted to by providing a
separate IPTV connection without any extra charges on him. To implement this, after taking written
consent from the subs, his existing IPTV plan may cancelled and IPTV is to be booked in Entry
Level Plan without any registration and installation charges.

Other features are furnished below:

1. Service will be provided to broadband customers of MTNL Delhi.

2. Booking will be done at Call Centre1500 and at Sanchar Haats.

3. Security deposit for STB shall be collected by M/s Aksh Optifiber Limited & will bear
 Asymmetric Digital Subscriber Line (ADSL)

Asymmetric Digital Subscriber Line (ADSL) is one form of the Digital Subscriber
Line technology, a data communications technology that enables faster data transmission
over copper telephone lines than a conventional voice band modem can provide. It does this
by utilizing frequencies that are not used by a voice telephone call. A splitter, or DSL filter,
allows a single telephone connection to be used for both ADSL service and voice calls at the
same time. ADSL can generally only be distributed over short distances from the central
office, typically less than 4 kilometres (2 mi), but has been known to exceed 8 kilometres
(5 mi) if the originally laid wire gauge allows for farther distribution.

At the telephone exchange the line generally terminates at a Digital Subscriber Line Access
Multiplexer (DSLAM) where another frequency splitter separates the voice band signal for
the conventional phone network. Data carried by the ADSL are typically routed over the
telephone company's data network and eventually reach a conventional internet
protocol network.
Operation

Currently, most ADSL communication is full duplex. Full-duplex ADSL communication is

usually achieved on a wire pair by either frequency-division duplex (FDD), echo-cancelling
duplex (ECD), or time divison duplex (TDD). FDD uses two separate frequency bands,
referred to as the upstream and downstream bands. The upstream band is used for
communication from the end user to the telephone central office. The downstream band is
used for communicating from the central office to the end user.

Frequency plan for ADSL. Red area is the frequency range used by normal voice telephony,
the green (upstream) and blue (downstream) areas are used for ADSL.
With standard ADSL (annex A), the band from 26.000 kHz to 137.825 kHz is used for
upstream communication, while 138 kHz – 1104 kHz is used for downstream
communication. Each of these is further divided into smaller frequency channels of
4.3125 kHz. These frequency channels are sometimes termed bins. During initial training,
the ADSL modem tests each of the bins to establish the signal to noise ratio at each bin's
frequency. The distance from the telephone exchange and the characteristics of the cable
mean that some frequencies may not propagate well, and noise on the copper wire,
interference from AM radio station and local interference and electrical noise at the customer
end mean that relatively high levels of noise are present at some frequencies both effects
the signal to noise ratio in some bins (at some frequencies) may be good or completely
inadequate. A bad signal-to-noise ratio measured at certain frequencies will mean that those
bins will not be used, resulting in a reduced maximum link capacity, but with an otherwise
functional ADSL connection.

The DSL modem will make a plan on how to exploit each of the bins, sometimes termed
"bits per bin" allocation. Those bins that have a good signal-to-noise ratio (SNR) will be
chosen to transmit signals chosen from a greater number of possible encoded values (this
range of possibilities equating to more bits of data sent) in each main clock cycle. The
number of possibilities must not be so large that the receiver might incorrectly decode which
one was intended in the presence of noise. Noisy bins may only be required to carry as few as
two bits, a choice from only one of four possible patterns, or only one bit per bin in the case
of ADSL2+, and very noisy bins are not used at all. If the pattern of noise versus frequencies
heard in the bins changes, the DSL modem can alter the bits-per-bin allocations, in a process
called "bitswap", where bins that have become more noisy are only required to carry fewer
bits and other channels will be chosen to be given a higher burden. The data transfer capacity
the DSL modem therefore reports is determined by the total of the bits-per-bin allocations of
all the bins combined. Higher signal-to-noise ratios and more bins being in use gives a higher
total link capacity, while lower signal-to-noise ratios or fewer bins being used gives a low
link capacity.

The total maximum capacity derived from summing the bits-per-bins is reported by DSL
modems and is sometimes termed sync rate. This will always be rather misleading, as the
true maximum link capacity for user data transfer rate will be significantly lower; because
extra data are transmitted that are termed protocol overhead, reduced figures
for PPPoA connections of around 84-87 percent, at most, being common. In addition, some
ISPs will have traffic policies that limit maximum transfer rates further in the networks
beyond the exchange, and traffic congestion on the Internet, heavy loading on servers and
slowness or inefficiency in customers' computers may all contribute to reductions below the
maximum attainable.

The choices the DSL modem make can also be either conservative, where the modem
chooses to allocate fewer bits per bin than it possibly could, a choice which makes for a
slower connection, or less conservative in which more bits per bin are chosen in which case
there is a greater risk case of error should future signal-to-noise ratios deteriorate to the point
where the bits-per-bin allocations chosen are too high to cope with the greater noise present.
This conservatism involving a choice to using fewer bits per bin as a safeguard against future
noise increases is reported as the signal-to-noise ratio margin or SNR margin. The telephone
exchange can indicate a suggested SNR margin to the customer's DSL modem when it
initially connects, and the modem may make its bits-per-bin allocation plan accordingly. A
high SNR margin will mean a reduced maximum throughput, but greater reliability and
stability of the connection. A low SNR margin will mean high speeds, provided the noise
level does not increase too much; otherwise, the connection will have to be dropped and
renegotiated (resynced). ADSL2+ can better accommodate such circumstances, offering a
feature termed seamless rate adaptation (SRA), which can accommodate changes in total
link capacity with less disruption to communications.

Frequency spectrum of modem on ADSL line

Vendors may support usage of higher frequencies as a proprietary extension to the standard.
However, this requires matching vendor-supplied equipment on both ends of the line, and
will likely result in crosstalk problems that affect other lines in the same bundle.

There is a direct relationship between the number of channels available and the throughput
capacity of the ADSL connection. The exact data capacity per channel depends on the
modulation method used.

ADSL initially existed in two versions (similar to VDSL), namely CAP and DMT. CAP was
the de facto standard for ADSL deployments up until 1996, deployed in 90 percent of ADSL
installs at the time. However, DMT was chosen for the first ITU-T ADSL standards, G.992.1
and G.992.2 (also called G.dmt and G.lite respectively). Therefore all modern installations of
ADSL are based on the DMT modulation scheme.
Interleaving and fastpath
Some ADSL connections use interleaving of packets to counter the effects of noise bursts on
the telephone line. Each packet to be sent (usually an Ethernet packet) is split into segments,
that are sent over a longer period of time interleaved with data from previous and following
packets. This allows error correction algorithms to recover the packets even if all data is lost
during the burst. A negative side effect of interleaving is an increase of latency by tens of
milliseconds. An ADSL profile with interleaving turned off is referred to asfastpath.
Installation issues

Due to the way it uses the frequency spectrum, ADSL deployment presents some issues. It is
necessary to install appropriate frequency filters at the customer's premises, to avoid
interference with the voice service, while at the same time taking care to keep a clean signal
level for the ADSL connection.

In the early days of DSL, installation required a technician to visit the premises.
A splitter or microfilter was installed near the demarcation poin, from which a dedicated data
line was installed. This way, the DSL signal is separated earlier and is not attenuated inside
the customer premises. However, this procedure is costly, and also caused problems with
customers complaining about having to wait for the technician to perform the installation. As
a result, many DSL vendors started offering a self-install option, in which they ship
equipment and instructions to the customer. Instead of separating the DSL signal at the
demarcation point, the opposite is done: the DSL signal is filtered at each phone outlet by use
of a low-pass filter for voice and a high-pass filter for data, usually enclosed in what is
known as a microfilter. This microfilter can be plugged directly into any phone jack, and
does not require any rewiring at the customer's premises.

A side effect of the move to the self-install model is that the DSL signal can be degraded,
especially if more than 5 voiceband devices are connected to the line. The DSL signal is now
present on all telephone wiring in the building, causing attenuation and echo. A way to
circumvent this is to go back to the original model, and install one filter upstream from all
telephone jacks in the building, except for the jack to which the DSL modem will be
connected. Since this requires wiring changes by the customer and may not work on some
household telephone wiring, it is rarely done. It is usually much easier to install filters at each
telephone jack that is in use.

DSL signals may be degraded by older telephone lines, surge protectors, poorly designed
microfilters, radio frequency interference, electrical noise, and by long telephone extension
cords. Telephone extension cords are typically made with small-gauge multi-strand copper
conductors which do not maintain a noise-reducing pair twist. Such cable is more susceptible
to electromagnetic interference and has more attenuation than solid twisted-pair copper wires
typically wired to telephone jacks. These effects are especially significant where the
customer's phone line is more than 4 km from the DSLAM in the telephone exchange, which
causes the signal levels to be lower relative to any local noise and attenuation. This will have
the effect of reducing speeds or causing connection failures.

FIREWALL

A firewall is a part of a computer system or network that is designed to block unauthorized

access while permitting authorized communications. It is a device or set of devices which is
configured to permit or deny computer applications based upon a set of rules and other
criteria.

Firewalls can be implemented in either hardware or software, or a combination of both.

Firewalls are frequently used to prevent unauthorized Internet users from accessing private
networks connected to the Internet, especially intranets. All messages entering or leaving the
intranet pass through the firewall, which examines each message and blocks those that do not
meet the specified security criteria.

There are several types of firewall techniques:

1. Packet filter: Packet filtering inspects each packet passing through the network and
accepts or rejects it based on user-defined rules. Although difficult to configure, it is
fairly effective and mostly transparent to its users. It is susceptible to IP spoofing.
2. Application gateway: Applies security mechanisms to specific applications, such as
FTP and Telnet servers. This is very effective, but can impose a performance
degradation.
3. Circuit-level gateway: Applies security mechanisms when a TCP or UDP connection
is established. Once the connection has been made, packets can flow between the
hosts without further checking.
4. Proxy server: Intercepts all messages entering and leaving the network. The proxy
server effectively hides the true network addresses.

Function

=> A firewall's basic task is to regulate some of the flow of traffic between computer
networks of different trust levels. Typical examples are the Internet which is a zone with no
trust and an internal network which is a zone of higher trust. A zone with an intermediate
trust level, situated between the Internet and a trusted internal network, is often referred to as
a "perimeter network" or Demilitarized zone (DMZ).

=> A firewall's function within a network is similar to physical firewalls with fire doors in
building construction.

History

(i)First generation: packet filters:

=> This type of packet filtering pays no attention to whether a packet is part of an existing
stream of traffic (it stores no information on connection "state"). Instead, it filters each packet
based only on information contained in the packet itself.
(ii) Second generation: application layer:

 The key benefit of application layer filtering is that it can "understand" certain
applications and protocols (such as File Transfer Protocol, DNS, or web browsing),
and it can detect if an unwanted protocol is sneaking through on a non-standard port
or if a protocol is being abused in any harmful way.

(iii) Third generation: "stateful" filters:

 Third-generation firewalls, in addition to what first- and second-generation look for,

regard placement of each individual packet within the packet series. This technology
is generally referred to as a stateful packet inspection as it maintains records of all
connections passing through the firewall and is able to determine whether a packet is
the start of a new connection, a part of an existing connection, or is an invalid packet.
Though there is still a set of static rules in such a firewall, the state of a connection
can itself be one of the criteria which trigger specific rules.

Types

There are several classifications of firewalls depending on where the communication is

taking place, where the communication is intercepted and the state that is being traced.

1.Network layer and packet filters:

Network layer firewalls, also called packet filters, operate at a relatively low level of the
TCP/IP protocol stack, not allowing packets to pass through the firewall unless they match
the established rule set. The firewall administrator may define the rules; or default rules may
apply. The term "packet filter" originated in the context of BSD operating systems.

2. Proxies :

A proxy device (running either on dedicated hardware or as software on a general-

purpose machine) may act as a firewall by responding to input packets (connection requests,
for example) in the manner of an application, whilst blocking other packets.

ROUTERS
Router is a device that forwards data packets along networks. A router is connected to at least
two networks, commonly two LANs orWANs or a LAN and its ISPs network. Routers are
located at gateways, the places where two or more networks connect, and are the critical
device that keeps data flowing between networks and keeps the networks connected to the
Internet.
When data is sent between locations on one network or from one network to a second
network the data is always seen and directed to the correct location by the router. The router
accomplishes this by using headers and forwarding tables to determine the best path for
forwarding the data packets, and they also use protocols such as ICMP to communicate with
each other and configure the best route between any two hosts.

The Internet itself is a global network connecting millions of computers and smaller
networks — so you can see how crucial the role of a router is to our way of communicating
and computing.

Types of routers

A demonstration of a router forwarding information to many clients.

Routers may provide connectivity inside enterprises, between enterprises and the Internet,
and inside internet service providers (ISPs). The largest routers (for example the Cisco CRS-
1 or Juniper T1600) interconnect ISPs, are used inside ISPs, or may be used in very large
enterprise networks. The smallest routers provide connectivity for small and home offices.

Routers for Internet connectivity and internal use

Routers intended for ISP and major enterprise connectivity almost invariably exchange
routing information using the Border Gateway Protocol (BGP). RFC 4098 defines several
types of BGP-speaking routers according to the routers' functions:

• Edge router (ER): An ER is placed at the edge of an ISP network. The router speaks
external BGP (EBGP) to a BGP speaker in another provider or large enterprise
Autonomous System(AS). This type of router is also called PE (Provider Edge)
routers.
• Subscriber edge router (SER): An SER is located at the edge of the subscriber's
network, it speaks EBGP to its provider's AS(s). It belongs to an end user (enterprise)
organization. This type of router is also called CE (Customer Edge) routers.
• Inter-provider border router: Interconnecting ISPs, this is a BGP-speaking router that
maintains BGP sessions with other BGP speaking routers in other providers' ASes.
 • Core router: A core router is one that resides within an AS as back bone to carry
traffic between edge routers.

Within an ISP: Internal to the provider's AS, such a router speaks internal BGP (IBGP) to
that provider's edge routers, other intra-provider core routers, or the provider's inter-provider
border routers.

History

Avaya ERS 8600 (2010)

The very first device that had fundamentally the same functionality as a router does today, i.e
a packet switch, was the Interface Message Processor (IMP); IMPs were the devices that
made up the ARPANET, the first packet switching network. The idea for a router (although
they were called "gateways" at the time) initially came about through an international group
of computer networking researchers called the International Network Working Group
(INWG). Set up in 1972 as an informal group to consider the technical issues involved in
connecting different networks, later that year it became a subcommittee of the International
Federation for Information Processing.

Router Manufacturers

The major router manufacturers include:

• Alcatel-Lucent
• Asus
• Avaya
• Brocade
• Buffalo
 • Cisco Systems
• D-link.

Reliance-
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Diag of BB NOC

PREFACE
B-Tech is the one of the reputed course.There is a industrial training scheme as an integral
part of b-tech in fifth semester.As a complementary to that every one has to submit a report
on the work conducted in the industry.
This report is thus prepared for the industrial training done at Mahanagar
Telephone Nigam Limited (MTNL) The organization shall make all the possible effort to
have a secure and safe domain

Nalin Madhav
E0829