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Semester: VII Branch: Information Technology Seminar title: Voice Over Internet Protocol

TABLE OF CONTENTS

ABSTRACT…………………………………………………………………...1

1. INTRODUCTION………………………………………………………….3

2. METHODOLOGY………………………………………………………...5
2.1 Telephone system v/s VoIP………………………………….6
2.2 VOIP gateway…………………………………………………. 7
2.3 VOIP network…………………………………………………. 8
10
2.4 Requirements
2.4.1 Software Requirements…………………………….9
2.4.2 Hardware Requirements …………………………..10

2.5 Types of communication…………………………………....11

3. WORKING
3.1 How VoIP works?...............................................................14

3.2 Codecs and switches ………………………………………..17


3.3 Protocols………………………………………………………19
3.4 Applications……………………………………………………22

4. ADVANTAGES AND DISADVANTAGES……………………………..25

5. FUTURE SCOPE…………………………………………………………28

6. CONCLUSION…………………………………………………………….30

7. BIBLIOGRAPHY………………………………………………………….31

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ABSTRACT

Voice over Internet Protocol (VoIP) is a general term for a family of transmission
Technologies for the delivery of voice communications over IP networks such as the
Internet or other packet switched networks. VoIP was born back in the stone age of the
internet, that is, 1995, when Israeli computer enthusiasts made the first voice
connection between two computers. This same year this technology was developed into
a software package called “Internet Phone Software”. Like many new technologies, it
wasn’t very pretty at first. Sound quality was poor and vastly inferior to the audio quality
of a standard phone network, which, by the way, isn’t really that good either. The
technology continued to be developed and by 1998 gateways had been established
allowing PC-to-phone connections. Later that year, phone-to-phone connections were
possible using the internet to transmit the audio. The phone-to-phone connections still
required a computer to initiate the call, but once the call was established, callers could
use a regular phone set.VoIP is fast becoming a big business, with the major telecom’s
getting on board offering VoIP service. Service is available for both commercial and
residential use, ranging from PC-to-PC service, all the way up to phone-to-phone.

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1.0 INTRODUCTION

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. In general, this means sending voice information in digital
form in discrete packets rather than in the traditional circuit-committed protocols of the
public switched telephone network (PSTN).Other terms for VoIP also include IP
Telephony, Internet telephony, Broadband Telephony, Broadband Phone, Voice over
Broadband.

How is this useful? Internet Telephony 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 Internet Telephony software that is available to make Internet phone calls, you are
bypassing the phone company (and its charges) entirely. Many industry experts see
Voice over IP as a leading-edge technology for the future in telecommunication. The
main users of VoIP service are Residential home users and Small Business or Home
Office.

Internet Telephony is a revolutionary technology that has the potential to completely


rework the world's phone systems. Internet Telephony providers like Vonage have
already been around for a little while and are growing steadily. Major carriers like AT&T
are already setting up Internet Telephony calling plans in several markets around the
United States, and the FCC is looking seriously at the potential ramifications of Internet
Telephony service.

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The interesting thing about Internet Telephony is that there is not just one way to place a
call. There are three different "flavors" of Internet Telephony service in common use
today:

ATA –ANALOG TELEPHONE ADAPTER

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 Internet Telephony.

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. Wi-Fi phones
allow subscribing callers to make Internet Telephony calls from any Wi-Fi hot spot.

Computer-to-computer

This is certainly the easiest way to use VoIP.You don’t have to pay for long distance
calls. All you need is the software, microphone, speakers, sound card and an internet
connection preferably fast as you get through 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.

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2.0 METHODOLOGY

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Voice over IP (VoIP) is a blanket description for any service that delivers standard voice
telephone services over Internet Protocol (IP). Internet protocol is used to transfer data
and files between computers.

"Voice over IP is the technology of digitizing sound, compressing it, breaking it up into
data packets, and sending it over an IP (Internet protocol) network where it is
reassembled, decompressed, and converted back into an analog wave form.” Protocols
used to carry voice signals over the IP network are commonly referred to as Voice over
IP or VoIP protocols.

2.1 Comparison: Telephone system vs. VoIP

Traditional telephone phone system

In a traditional telephone phone system (POTS, Plain Old Telephone System) an analog
voice signal is switched to make a single direct connection to each point. This is known
as 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're connecting two
points in both directions, the connection is called a circuit. This is the foundation of the
Public Switched Telephone Network (PSTN).This system works by setting up a
dedicated channel (or circuit) between two points for the duration of the call. These
telephony systems are based on copper wires carrying analog voice data over the
dedicated circuits.

VoIP

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VoIP, in contrast to PSTN, uses what is called packet-switched technology. Using this
system the voice information travel to destination in countless no of individual packets
across the internet. 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.

Packet switching is very efficient. It lets the network route the packets along the least
congested and cheapest lines. It also frees two computers communicating with each
other so that they can accept information from other computer as well. This type of
communication can accommodate many transmissions at the same time because
each packet only takes up what bandwidth that is necessary.

2.2 VOICE GATEWAY


The VoIP network acts as a gateway to the existing PSTN network. This gateway forms
the interface for transportation of the voice content over the IP network. This is
responsible for call origination, call detection, analog-to-digital conversion of voice
packets. Voice (analogue and/or digital) compression, echo cancellation, silence
suppression, and statistics gathering are their optional features. The gateways must
also perform some of the database services, such as phone number translations, host
lookup and signaling. The extent of gateway functionalities is based on Internet
Telephony or VoIP products used. Fig. 1 shows the architecture of a typical gateway.

FIG 1 ARCHITECTURE OF A TYPICAL GATEWAY

The DSP in a gateway is responsible for signal processing functions such as analogue-
to-digital conversion of voice signals, voice compression, echo cancellation, and voice
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activity detection. The functions like call origination, call detection, signaling, and phone
number translations are performed by the microprocessor.

2.3 A TYPICAL VoIP NETWORK

Fig. 2 shows a typical Internet Telephony or VoIP network. The IP network should
ensure smooth delivery of voice and signaling information to the Internet Telephony
elements. Since the IP network is to carry both voice and data, it must be able to
prioritize the voice traffic. This prioritization is required for real-time Internet Telephony
applications to ensure that voice traffic is unaffected by other network traffic. Without
prioritization, the voice packets may be bogged down by heavy data traffic like large file
transfers using file transfer protocol (FTP).The voice packets are encapsulated with
Real-time protocol (RTP) and Real-time control protocol (RTCP) for real-time
transfer. The Resource reservation protocol (RSVP) is used at the networking
gateways (such as the routers) to reserve a particular amount of bandwidth for real-time
applications (Internet Telephony, video multicasting, etc).

FIG 2: A TYPICAL FULL SERVICE VOIP NETWORK

Unlike the PCM data streams in circuit switched telephony, in VoIP data travels over the
networks in packets. In this digitized voice is bundled into IP packets and sent out into
the network for delivery. Routers, switches, and other network equipment direct the

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packets to their destination IP address. This mode is called packet switched telephony.
The transport of voice packets is affected by several factors, such as the amount of
bandwidth available in the network connection, the delay that the packet experiences,
and any packet loss or corruption that occurs. The ability of the network to deliver the
voice packets quickly and consistently is referred to as Quality of Service (QoS).

2.4 Requirements of a VoIP


The requirement for implementing an IP Telephony solution to support Voice Over IP
varies from organization to organization, and depends on the vendor and product
chosen. The following section aims to identify the fundamental requirements in the
general case and is split into 3 sections:

• Software Requirements
• Hardware Requirements

2.4.1 Software Requirements


The software package chosen will reflect the organizational needs, but should contain
the following modules as defined in the Technology Guide Series - Voice Over IP
Publication, and other sources.

Voice Processing Module


This aspect of the software is required to prepare voice samples for transmission. The
functionality provided by the voice processing module should support:

A PCM Interface is required to receive samples from the telephony interface (e.g. a
voice card) and forward them to the Voice over IP software for further processing.

Echo Cancellation is required to reduce or eliminate the echo introduced as a result of


the round trip exceeding 50 milliseconds.

Idle Noise Detection is required to suppress packet transmission on the network when
there are no voice signals to be sent. This helps to reduce network traffic as up to 60%
of voice calls are silence and there is no point in sending silence.

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A Tone Detector is required to discriminate between voice and fax signals by detecting
DTMF (Dial Tone Multi frequency) signals.

A Voice Playback Module is required at the destination to buffer the incoming


packets before they are sent to the Codec for decompression.

Call Signaling Module is required to serve as a signaling gateway which allows calls
to be established over a packet switched network as opposed to a circuit switched
network (PSTN for example).

Packet Processing Module This module is required to process the voice and
signaling packets ready for transmission on the IP based network.

Network Management Protocol Allows for fault accounting, configuration


management.

2.4.2 Hardware Requirements


The exact hardware, which would be required, again, depends on organizational needs
and budget. The most obvious requirement is the existence (or installation) of an IP
based network within the branch office, gateway is required to bridge the differences
between the protocols used on an IP based network and the protocols used on the
PSTN.The gateway takes a standard telephone signal and digitizes it before
compressing it using a Codec. The compressed data is put into IP packets and these
packets are routed over the network to the intended destination. The PC's attached to
the IP based network require the voice/fax software. They also require Full Duplex
Voice Cards which allow both communicating parties to speak at the same time - as
often happens in reality.

As an alternative to installing Voice Cards, IP Telephones can be attached to the


network to facilitate Voice Over IP. A secondary gateway should be considered as a
backup in the event of the failure of the primary gateway.

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2.5 TYPES OF COMMUNICATION

Mainly the different types of communications that exist in an Internet Telephony are:
• PC to PC communication.
• PC to PHONE communication.
• PHONE to PHONE communication.

PC to PC Communication:

• Need a PC with sound card


• IP Telephony software: Cuseeme, Internet Phone...
• Video optional

PC to Phone Communication:

• Need a gateway that connects IP network to phone network (Router to


PBX)

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Figure: PC TO PHONE COMMUNICATION

Phone to Phone communication:

• Need more gateways that connect IP network to phone Networks.


• The IP network could be dedicated intra-net or the Internet.
• The phone networks could be intra-company PBXs or the carrier switches.

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3.1 How VoIP works?

VoIP is a collection of digitally encrypted voice transmissions that are carried over a
network based on a single common language, or protocol — in this case, the Internet
Protocol TCP/IP. VoIP converts the voice signal from your telephone into a digital
signal that travels over the Internet and is then converted back at the other end, so you
can speak to anyone with a regular phone number. When placing a VoIP call using a
phone with an adapter, you'll hear a dial tone and dial just as you always have. VoIP
may also allow you to make a call directly from a computer using a conventional
telephone or a microphone.

Let's say that you and your friend both have service through a VoIP service provider.
You both have your analog phones hooked up to the service-provider ATA’s.

• You pick up the receiver, which sends a signal to the ATA.

• The ATA receives the signal and sends a dial tone. This lets you know that you
have a connection to the Internet.

• 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.

• The phone number data is sent in the form of a request to your Internet
Telephony company's call processor. The call processor checks it to ensure that
it is in a valid format.

• The call processor determines to whom to map the phone number. In mapping,
the phone number is translated to an IP address. 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.

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• 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.

• 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.

• You finish talking and hang up the receiver.

• 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 do.

VoIP works as a peer-to-peer application, entailing handshaking and direct media


exchange between two IP devices. To call someone, the user dials the telephone
number, the handset translates that number into IP address format (e.g.,
123.456.11.22), and the device sends encrypted data packets whose payloads contain
messages conforming to a particular call-setup protocol between the two devices. They
then establish a common connection for voice exchange. Their device rings, they pick
up, and media packets flow in both directions.

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VoIP user packet-switched technology, in this the voice information travels to its
destination in countless individual network packets across the Internet. This type of
communication presents special TCP/IP challenges because the Internet wasn't really
designed for the kind of real-time communication a phone call represents. Individual
packets may — and almost always do — take different paths to the same place. It's not
enough to simply get VoIP packets to their destination. The packets must arrive in a
fairly narrow time window and be assembled in the correct order to be intelligible to the
recipient.

To improve performance, VoIP employs encoding schemes and compression


technology to reduce the size of the voice packets so they can be transmitted more
efficiently. Audio signals are also digitally processed in order to accentuate the voice
information and suppress background noise. To conserve bandwidth, VoIP systems stop
transmitting during lulls in a conversation and even generate some "comfort noise" to
forestall the eerie silence that might make you think the call was disconnected.

VoIP uses a number of compression standards that offer different balances between
packet size and audio quality. Generally speaking, the higher the compression the more
simultaneous calls you can have, but the lower voice quality will be.

Despite all of the advantages of a VoIP system, it does have its drawbacks. For
instance, some VoIP services will not work during power outages and the service
provider may not offer any type of back-up power solution. Many VoIP providers may
not offer directory assistance or white pages listings which is essential to the small
business.

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3.2 CODECS, SOFT SWITCHES


A codec, which stands for coder-decoder, converts an audio signal into a compressed
digital form for transmission and then back into an uncompressed audio signal for
replay. This is the essence of VoIP. Digital-to-analog conversion is seen in everything
from CD players to cell phones to video game consoles.

Codecs accomplish the conversion by sampling the audio signal several thousand
times per second. For instance, a G.711 codec samples the audio 64,000 times a
second. It converts each tiny sample into digitized data and compresses it for
transmission. When the 64,000 samples are reassembled, the pieces of audio missing
between each sample are so small that to the human ear, it sounds like one continuous
second of audio signal. There are different sampling rates in VoIP depending on the
codec being used:

• 64,000 times per second


• 32,000 times per second
• 8,000 times per second
A G.729A codec has a sampling rate of 8,000 times per second and is the most
commonly used codec in VoIP. It is a compromised balance between sound quality and
efficiency of bandwidth.

Codecs operate by using advanced algorithms that help them sample, sort, compress
and packetize audio data. The CS-ACELP algorithm (CS-ACELP = conjugate-structure
algebraic-code-excited linear prediction) is one of the most prevalent algorithms in
Internet Telephony. CS-ACELP helps to organize and streamline the available
bandwidth. Annex B is an aspect of CS-ACELP that creates the transmission rule,
which basically states "if no one is talking, don't send any data." As discussed
before, the efficiency created by this rule is one of the greatest ways in which packet
switching is superior to circuit switching. It is Annex B in the CS-ACELP algorithm that is
responsible for that aspect of the Internet Telephony call.

So the codec works with the algorithm to convert and sort everything out, but none of
that is any good without knowing where to send the data. In Internet Telephony, that
task is handled by soft switches.

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E.164 is the name given to the standard for the North American Numbering Plan
(NANP). Simply stated, this is the numbering system that phone networks use to know
where to route a call based on the numbers entered into the phone keypad. In that way,
a phone number is like an address.

IP addresses correspond to a particular device on the network. It can be a computer, a


router, a switch, a gateway or, in this case, a telephone. To make matters worse, IP
addresses are not always static. They are assigned by a DHCP server on the network
and generally change with each new connection. So the challenge with Internet
Telephony is figuring out a way to translate NANP phone numbers to IP addresses and
then finding out the current IP address of the requested number. This is the mapping
process referred to earlier and is handled by a central call processor running a soft
switch.

The central call processor is a piece of hardware running a specialized


database/mapping program called a soft switch. Think of the user and the phone or
computer associated with that user as one package -- man and machine. That package
is called the endpoint. The soft switch connects endpoints.

Soft switches know:

• Where the endpoint is on the network


• What phone number is associated with that endpoint
• The current IP address assigned to that endpoint

So when a call is placed using Internet Telephony/VoIP, a request is sent to the soft
switch asking which endpoint is associated with the dialed phone number and what that
endpoint's current IP address is. The soft switch contains a database of users and
phone numbers. If it doesn't have the information it needs, it hands off the request
downstream to other soft switches until it finds one that can answer the request. Once it
finds the user, it locates the current IP address of the device associated with that user in
a similar series of requests. It sends back all the relevant information to the softphone or
IP phone, allowing the exchange of data between the two endpoints.

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Soft switches work in tandem with the devices on the network to make VoIP possible. In
order for all of these devices to work together, they must communicate in the same way.
This communication is one of the most important aspects that will have to be refined in
order for Internet Telephony to really take off. Currently, there are three protocols used
for this communication.

3.3 PROTOCOLS
A VoIP phone call occurs in two stages:
1. Call setup. This stage is required to set up everything needed to make the
telephone connection between the person making the call (the caller) and the
person receiving the call (the called party).
2. The call itself. The audio component of the conversation must be encoded and
transmitted across the network.

The call setup stage of the call requires protocols that enable dial tone, number lookup,
ringing, and busy signals before the call even occurs. In addition, the call setup
protocols handle things that happen after the call -- any resource cleanup and statistical
reporting.

Call setup protocols use the Transmission Control Protocol (TCP) or User Datagram
Protocol (UDP) to transfer data during the setup and takedown phases of a telephone
call. Each protocol uses a well-known port or ports to communicate with a call server,
which functions like a PBX to enable IP phone calls. The required setup messages are
sent back and forth between the caller, called party, and call server. For calls that travel
between the VoIP network and the Public Switched Telephone Network (PSTN), the call
server converses with a voice gateway using the same call setup protocol.

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The setup messages, which vary in size and number, handle functions like the mapping
of phone numbers to IP addresses, generating dial tones and busy signals, ringing the
called party, and hanging up. Many different call setup protocols are in current use for
VoIP deployments; some are standardized and some proprietary. The major call setup
protocols are described below.

H.323

The call setup protocol H.323 is standardized by the International Telecommunications


Union (ITU). H.323 is widely deployed among the call setup protocols, and has been
around the longest. In a VoIP environment, H.323 is a common protocol running on
voice gateways to connect the VoIP network to the PSTN.

H.323 is actually a family of telephony-based standards for multimedia, including voice


and videoconferencing. This set of interrelated protocols has been refined over many
years. As a result, it is robust and flexible, but one downside of its robust capabilities is
high overhead: a calling session includes lots of handshakes and data exchanges for
each function performed. Because H.323 uses TCP for communication, setting up a call
with H.323 requires many back-and-forth TCP flows. H.323 requires additional
configuration on the voice gateway, which maintains information about how calls are
routed.

MGCP

The Media Gateway Control Protocol (MGCP) is another commonly used call setup
protocol. It is covered in the informational RFC 2705. MGCP differs from some other
call setup protocols in that the endpoints, or phones, do not use MGCP to control the
phone call itself. More commonly, MGCP is used so that a call server can control a
voice gateway connection to the PSTN.MGCP sends messages between the gateway
and call server over UDP port 2427. Because the call server controls the gateway, the
bulk of the call control intelligence resides there. Likewise, call routing information is
configured in the call server instead of in the gateway.

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SIP

SIP (Session Initiation Protocol) is a lightweight protocol developed by the IETF in


RFC 3261 (with Proposed Standard status). SIP represents typical data-networking
logic, which asks: Why use a heavyweight protocol (such as H.323) when a lightweight
protocol (such as SIP) gets the job done most of the time? SIP represents the future for
call setup as more vendors, including Cisco and Avaya, offer SIP phone/endpoint
support. In addition, Microsoft recently announced the availability of their Office
Communications Server, which uses SIP for call setup. Although SIP can use either
TCP or UDP for transport, most implementations use TCP and port 5060. SIP
messages are similar to HTTP in that they are text-based and generally follow a
request-response structure.

Proprietary

In addition to the standardized call setup protocols discussed above, certain vendors
have provided their own proprietary protocols. One popular example is the Cisco
Skinny Client Control Protocol (SCCP). SCCP or "Skinny" provides a simple,
lightweight call setup protocol for Cisco devices. Skinny passes messages using TCP
and port 2000.There is no single, dominant call setup protocol in use today. The
protocols discussed here (H.323, MGCP, SIP, and SCCP) are all commonly used in
VoIP equipment. However, the trend is moving toward SIP as the call setup protocol of
choice.

RTP

Unlike call setup protocols, where no one protocol dominates, the single protocol that is
used almost exclusively for transfer of VoIP conversations is RTP. Widely used for
streaming audio and video, RTP is designed for applications that need real-time
performance to send data in one direction with no acknowledgments Since a VoIP call is
bidirectional; two RTP streams carry the conversation, one in each direction. The path
that these RTP streams take through the network and the impairments encountered
along the way are important factors in determining the quality of voice conversations
carried over data networks. RTP is an application protocol that uses UDP for transport.

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RSVP
Resource Reservation Protocol (RSVP) is the protocol which supports the reservation
of resources across an IP network. RSVP can be used to indicate the nature of the
packet streams that a node is prepared to receive.

3.4 APPLICATIONS

A wide variety of applications are available. The first application, shown in Figure 1, is a
network configuration of an organization with many branch offices (e.g., a bank) that
wants to reduce costs and combine traffic to provide voice and data access to the main
office. This is accomplished by using a packet network to provide standard data
transmission while at the same time enhancing it to carry voice traffic along with the
data. Typically, this network configuration will benefit if the voice traffic is compressed.

Voice over packet provides the Interworking function (IWF), which is the physical
implementation of the hardware and software that allows the transmission of combined
voice and data over the packet network. The interfaces the IWF must support in this
case are analog interfaces, which directly connect to telephones or key systems. The
IWF must emulate the functions of both a private branch exchange (PBX) for the
telephony terminals at the branches, as well as the functions of the telephony terminals
for the PBX at the home office. A traditional Private Branch Exchange (PBX) connects
all the phones within an organization to the public telephone network.

FIGURE 1. BRANCH OFFICE APPLICATION

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A second Internet Telephony application, shown in Figure 2, is a trunking application. In


this scenario, an organization wishes to send voice traffic between two locations over
the packet network and replace the tie trunks used to connect the PBXs at the
locations. This application usually requires the IWF to support a higher-capacity digital
channel than the branch application, such as a T1/E1 interface of 1.544 or 2.048 Mbps.
The IWF emulates the signaling functions of a PBX, resulting in significant savings to
companies' communications costs.

FIGURE 2. INTEROFFICE TRUNKING APPLICATION

A third application of Internet Telephony software is interworking with cellular networks,


as shown in Figure 3. The voice data in a digital cellular network is already compressed
and packetized for transmission over the air by the cellular phone. Packet networks can
then transmit the compressed cellular voice packet, saving a tremendous amount of
bandwidth. The IWF provides the transcoding function required to convert the cellular
voice data to the format required by the public switched telephone network (PSTN).

FIGURE 3. INTEROFFICE TRUNKING APPLICATION

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4.0 ADVANTAGES OF USING VoIP

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• VoIP technology uses the Internet's packet-switching capabilities to provide


phone service. Internet Telephony 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.

• Provides security through encryption.

• You can use the service almost anywhere in the world, as long as there is a high
speed internet connection.

• Provide features such as voicemail, caller ID, call forwarding and more. Normally
you would pay extra for these features with the phone or cellular companies.

• Allows you to save money on your long distance calling and decreases costs
of calls and phone communication.

4.1 DISADVANTAGES OF USING VoIP

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• First of all, VoIP is dependant on wall power. Your current phone runs on
phantom power that is provided over the line from the central office. Even if
your power goes out, your phone (unless it is a cordless) still works. With
Internet Telephony, no power means no phone. A stable power source must be
created for VoIP.

• Another consideration is that many other systems in your home may be


integrated into the phone line. Digital video recorders, digital subscription TV
services and home security systems all use a standard phone line. There is
currently no way to integrate these products with VoIP.

• Because VoIP uses an Internet connection, it is susceptible to all the


hiccups normally associated with home broadband services. All of these
factors will affect call quality:
 Latency
 Jitter
 Packet loss

Phone conversations can become distorted, garbled or lost because of


transmission errors. Some kind of stability in Internet data transfer needs to
be guaranteed before Internet Telephony could truly replace traditional
phones.

• VoIP is susceptible to worms, viruses and hacking, although this is very


rare and Internet Telephony developers are working on Internet Telephony
encryption to counter this.

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5.0 FUTURE SCOPE


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Voice over Internet Protocol (VoIP) is one of the hottest and most hyped technologies in
the communications industry. Businesses and consumers are already taking advantage
of the cost savings and new features of making calls over a converged voice-data
network, and the logical next step is to take those advantages to the wireless world.
Wireless VoIP theoretically has many advantages, including reduced cost for calls and
higher-bandwidth data transfers versus a traditional cellular connection. WiFi networks
cost a fraction of what traditional cell tower technology costs to deploy, and can be
rolled out quickly without the detailed site reviews required to install radio towers. More
importantly, wireless VoIP can actually dramatically improve call quality — especially in
residential areas or office towers where traditional mobile network coverage is spotty.
What does this mean for the average user? As the workforce moves to a flexible, non-
static environment, wireless VoIP will allow employees to roam from mobile networks to
WiFi-based home and office networks — using a single device to manage
communications that currently traverses mobile, home, and office handsets.

Wireless VoIP offers potential savings by allowing companies to change the way they
manage their phone systems. For example, instead of having voicemail, caller ID and e-
mail separately, wireless VoIP will allow customers to retrieve all of their messages in
one place, alleviating the pain of having different operators for different services and
ultimately dealing with several bills at a time. Employees can also download software
applications, enabling them to turn their phones into “mini-computers” and track
inventory, or log onto the company’s intranet. The biggest obstacles to making WiFi
telephony a success are not that different from the early days of cell phones. Three
main areas need addressing are cost of infrastructure to support calls; and Security.
Once reliable, roaming-friendly networks are built out, WiFi enabled handsets are
broadly available, and the connections are as easy to make as with our standard cell
phones, wireless VoIP will become a reality.

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6.0 CONCLUSION

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VoIP has grown in recent years because small business customers and consumers are
clamoring for this technology because of its easy-to-use and sophisticated features that
surpass those of traditional phones, its software upgrade potential, and its bandwidth
efficiency. The business world has already recognized VoIP as “unified communication”
as it integrates the phone calls, faxes, voice mail, email, Web conferences and more-as
discrete units that can all be delivered through any means and any handset, including
cellphones.VoIP also offers the advantage of running both voice and data
communication over a single network which can represent a significant saving in
technology costs.VoIP can facilitate tasks and provide services that may be more
difficult to implement using the PSTN like the ability to transmit more than one
telephone calls over the same broadband connection, Secure calls using standardized
protocols, location independence and integration with other services available over the
internet. Examples of some cost-efficient residential IP Telephony services include
Vonage, Packet8 and Skype.

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7.0 BIBLIOGRAPHY

1. Computer Networks by Andrew S.Tanenbaum


2. Internetworking with TCP/IP by Douglas E.comer
3. www.iptelephony.org/
4. www.tmcnet.com
5. www.ietf.org
6. www.avaya.com
7. www,HowStuffsWork.com

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