RESEARCH REPORT

SUBMITTED TO

HEAD OF DEPARTMENT OF COMPUTER SCIENCE

AND ENGG.
KIIT COLLEGE OF ENGG.
MARUTI KUNJ, GURGAON
IN PARTIAL FULFILMENT OF THE

REQUIREMENT FOR THE AWARD OF THE DEGREE

OF

Bachelor of Engineering
IN

Computer Science & Engg.

BY
MANJEET RANA

DEPARTMENT OF COMPUTER SCIENCE &

ENGG.
KIIT COLLEGE OF ENGG.
MARUTI KUNJ, GURGAON

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 MAR 2010

Table of Contents

PREFACE......................................................................................3

ACKNOWLEDGMENT......................................................................4

INTRODUCTION TO VOIP.............................................................5-6

REQUIREMENT OF VOIP.............................................................7-10

MAIN TYPES OF VOIP..............................................................11-12

WORKING OF VOIP..................................................................13-16

PROS AND CONS.....................................................................17-18

OPPORTUNITIES..........................................................................19

CONCLUSION..............................................................................20

BIBLIOGRAPHY............................................................................21

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 PREFACE

In our B.E. ( CSE ) course it is obligatory on part of every student to

carry out a Individual Research in partial fulfillment requirement of the
degree course . Students by doing the research , experience the latest
technologies in the real world and have a deep view of their
Applications in practical work .

I have made my report on VOIP . What I have done on this research , a

report of that is being presented in this report .

MANJEET RANA

B.E. CSE VII SEM

ROLL-1125

HARYANA.

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 ACKNOWLEDGEMENT

The report titled VOIP has been conducted by me during my VIII SEM at
KIIT COLLEGE OF ENGG. I have completed this report, based on the
primary research and study, under the guidance of Ms. KIRTI (MENTOR) .

I owe enormous intellectual debt towards my guide Ms. KIRTI who have
augmented my knowledge in the field of computer network technologies.
She has helped me learn about same and giving me valuable insight into the
analysis of the such technology in the industry.

Last but not the least, I feel indebted to all those persons and organizations
which have provided help directly or indirectly in successful completion of
this study.

Dated: 23, Mar, 2010

MANJEET RANA

B.E. CSE VIII SEM

ROLL NO. -1125

KIIT COLLEGE OF ENGG .

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 INTRODUCTION

VoIP (Voice Over IP - that is, voice delivered using the Internet Protocol) is a
term used in IP telephony for a set of facilities for managing the delivery of
voice information using the Internet Protocol (IP). 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). A major advantage of VoIP and Internet telephony is that it
avoids the tolls charged by ordinary telephone service

VoIP is therefore telephony using a packet based network instead of the PSTN
(circuit switched).

The first VoIP application was introduced in 1995 - an "Internet Phone". An

Israeli company by the name of "VocalTec" was the one developing this
application. The application was designed to run on a basic PC. The idea was
to compress the voice signal and translate it into IP packets for transmission
over the Internet. This "first generation" VoIP application suffered from
delays (due to congestion), disconnection, low quality (both due to lost and
out of order packets) and incompatibility.
VocalTec's Internet phone was a significant breakthrough, although the
application's many problems prevented it from becoming a popular product.
Since this step IP telephony has developed rapidly. The most significant
development is gateways that act as an interface between IP and PSTN
networks.

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

"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.." The transmission of sound over a packet switched network in
this manner is an order of magnitude more efficient than the transmission of
sound over a circuit switched network.

As mentioned before, VoIP saves bandwidth also by sending only the

conversation data and not sending the silence periods. This is a considerable
saving because generally only one person talks at a time while the other is
listening. By removing the VoIP packets containing silence from the overall
VoIP traffic we can reach up to 50% saving. In a circuit switched network,
one call consumes the entire circuit. That circuit can only carry one call at a
time.

In a packet switched network, digital data is chopped up into packets, sent

across the network, and reassembled at the destination. This type of circuit
can accommodate many transmissions at the same time because each
packet only takes up what bandwidth that is necessary.. Internet Telephony
simply takes advantage of the efficiencies of packet switched networks.

Gateways are the key component required to facilitate IP Telephony. A

gateway is used to bridge the traditional circuit switched PSTN with the
packet switched Internet.

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 Requirements of a VoIP
The requirements 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:

Hardware Requirements

Protocol Requirements

Software Requirements

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:

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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.

A Tone Detector is required to discriminate between voice and fax

signals by detecting DTMF (Dial Tone Multi frequency) signals.

The Packet Voice Protocol is required to encapsulate

compressed voice and fax data for transmission over the network.

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 This 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

and configuration management to be performed.

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Hardware Requirements
The exact hardware, which would be required, again, depends on
organizational needs and budget. The list below highlights the most general
hardware required.

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
outlined above. 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.

Protocol Requirements
There are many protocols in existence but the main ones are considered to
be the following:

H.323 is an ITU (International Telecommunications Union) approved

standard which defines how audio/visual conferencing data is transmitted
across a network. H.323 relies on the RTP (Real-Time Transport Protocol) and
RTCP (Real Time Control Protocol) on top of UDP (User Datagram Protocol) to
deliver audio streams across packet based networks.

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G.723.1 defines how an audio signal with a bandwidth of 3.4KHz should
be encoded for transmission at data rates of 5.3Kbps and 6.4Kbps. G.723.1
requires a very low transmission rate and delivers near carrier class quality.
The VoIP Forum as the baseline Codec for low bit rate IP Telephony has
chosen this encoding technique.

G.711. The ITU standardized PCM (Pulse Code Modulation) as G.711. This
allows carrier class quality audio signals to be encoded for transmission at
data rates of 56Kbps or 64Kbps. G.711 uses A-Law or Mu-Law for amplitude
compression and is the baseline requirement for most ITU multimedia
communications standards.

Real-Time Transport Protocol (RTP) is the standard protocol

for streaming applications developed within the IETF (Internet Engineering
Task Force).

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.

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 MAIN TYPES OF VoIP
VoIP has broadly three main branches, which can and do overlap.

VoIP over the Internet This is probably the best known and most
publicized, talking PC to PC. Basically free telephone calls. The call is only
free if both parties to the call have access to the public Internet at zero cost..

Advantage. free calls regardless of distance or length of call.

Disadvantage. often the voice quality is bad due to the lack of

bandwidth available for the call.

Other factors Have to use a PC or other computer running VoIP software.

Office to Office A large multinational company will have offices

across the whole country. They have a fixed data network connecting all the
offices together. This allows every computer access to every other computer
in the company. By installing a VoIP Gateway in each office and connecting it
to the office legacy PBX and to the data network, employees use the data
network for voice calls between offices.

Advantages. Interoffice calls are free, since the company already has
the bandwidth between offices. The technology is transparent to the user,
and requires minimum training. The only new equipment required is a
gateway at each office. Voice quality is good, because the company has

control over the bandwidth.

Disadvantage. Extra bandwidth may be required between offices,

which offset the savings.

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Other factors... The carrier providing the interoffice bandwidth will almost
certainly offer an alternative solution including management of the internal
telephone traffic.

IP PBX A traditional Private Branch Exchange (PBX) connects all the

phones within an organization to the public telephone network. Essentially IP
PBX replaces all the internal phones with VoIP telephones. The IP PBX has
standard telephone trunk connections to the public telephone network. The
IP PBX is a PBX with VoIP, but it also has the ability to support VoIP over the
Internet and Office to Office VoIP.

Advantages. Single cable infrastructure. The technology is transparent

to the user, and requires minimum training. Future proof technology.

Disadvantages. Primarily useful for Greenfield sites, but can be

adapted to work with existing technology.

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 WORKING OF VOIP

How VoIP works : Part 1

Let us look at very simple VoIP call. Consider two VoIP telephones connected
via an IP network .In this example both VoIP telephones are connected to a
local LAN. Sally’s phone has an IP address of 192.168.1.1, Bill’s phone is
192.168.1.2, and the IP addresses uniquely identify the telephones. Both our
phones are configured to use a widely used VoIP standard called H.323.

Bill wants to talk to Sally and his phone knows the IP address of Sally’s
phone. Bill lifts the handset and 'dials' Sally, the phone sends a call setup
request packet to Sally's phone, Sally’s phone starts to ring, and responds to
Bill's phone with a call proceeding message. When Sally lifts the handset the
phone sends a connect message to Bill's phone. The two phones will now
exchange the data packets containing the speech. At the end of the call Bill
replaces his handset and phone stops sending voice data sends a disconnect

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message and Sally's phone responds with a release message. The call is now
complete. all the messages contain the Q931 ISDN protocol.

Having introduced VoIP I will now talk about three main 'types' of VoIP
installed in the market place today.

How VoIP works part 2 : The Protocols.

I have made an assumption that both ends of a VoIP telephone conversation

are compatible. This compatibility only happens if both ends agree to use the
same protocol. All manufacturers who claim to be producing industry

standard voice over IP either support SIP or H.323 protocol.

H.323

Over the next few years, the industry will address the bandwidth limitations
by upgrading the Internet backbone to asynchronous transfer mode (ATM),
the switching fabric designed to handle voice, data, and video traffic. Such
network optimization will go a long way toward eliminating network
congestion and the associated packet loss. The Internet industry also is
tackling the problems of network reliability and sound quality on the Internet
through the gradual adoption of standards. Standards-setting efforts are
focusing on the three central elements of Internet telephony: the audio
codec format; transport protocols; and directory services.

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 H.323 Call Sequence :

As such, H.323 addresses the core Internet-telephony applications by

defining how delay-sensitive traffic, (i.e., voice and video), gets priority
transport to ensure real-time communications service over the Internet. (The
H.324 specification defines the transport of voice, data, and video over
regular telephony networks, while H.320 defines the protocols for
transporting voice, data, and video over integrated services digital network
(ISDN).

How VoIP works part 3: Encoding

The call control part of H.323 sets up the parameters for the full duplex voice
path between source telephone and destination telephone. I will continue
with my analogies to explain how your voice gets transported across the
Internet.

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In terms of H.323 there is a trade off between call quality and bandwidth, in
general the higher the quality the greater the bandwidth required

During the call setup portion of H.323 the phones have to decide which
speech encoder/decoder to use when they send the speech to the other
phone, Bill and Sally both have phones that support G.723.1, G.711 and
G729.

The main difference between each of these encoders is the amount of

bandwidth they use, G.711 uses 64kbit/s and G.723.1 can use as little as
5.3kbit/s. Although it would seem obvious to use the encoder with the lowest
bandwidth, there is a loss of quality with a lower bandwidth.. At the same
time a stream of G723.1 encoded voice data starts being sent from each
phone to the other phone.

How VoIP works part 4 :Hear the Quality.

The performance of the speech encoders at each end, the number of

packets lost on route, Latency and Jitter.

In the early days of voice calls via satellite there would be an annoying echo.
As the technology improved the echo disappeared. Echo suppression is very
key to good quality VoIP calls. I do not dwell on the subject since the
mathematics is beyond my comprehension. Good echo suppression makes
for quality calls.

Be warned that because a manufacturer has a G.723.1 encoder it may not

sound the same as another manufacturer who claims to have G.723.1,
quality does vary. As a general rule the occasional lost packet will not affect
too drastically the quality of a call, but lose 5 in a row and an entire word is
lost and this will be a problem. So if you are going to have lost packets make
sure they are only lost in a regular distributed manner. 5% lost packets
distributed evenly will not result in the loss of words lose 5% of the words by
clustering the packets and the effect is bad.

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PROS AND CONS :

Advantages of VoIP

There are many advantages to be gained from implementing an IP

Telephony solution within the organization. The following list aims to
highlight some of the advantages of such a strategy:

Single network infrastructure. When installing VoIP in the

office only a single cable is required to the desk, for both telephone and
data. Eliminating separate telephone wiring.

VoIP uses "soft" switching which eliminates most of the legacy

PBX equipment. Reducing the cost of installing a communications infra-
structure and the maintenance cost once installed.

Simple upgrade path. The VoIP PBX technology is software based.

It is easier to expand, upgrade and maintain than its traditional telephony
counterparts.

Bandwidth efficiency. VoIP can compress more voice calls into

available bandwidth than legacy telephony.. IP Telephony helps to eliminate
wasted bandwidth by not transporting the 60% of normal speech which is
silence

Only one physical network is required to deal with

both voice/fax and data traffic instead of two physical networks.
Having only one physical network has the following advantages:

lower physical equipment cost ,lower maintenance costs.

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Weaknesses:

While there are many aspects of VoIP which provide considerable benefits,
the technology is still very young and problems remain. The following section
looks at some of the weaknesses of this technology and their consequences.

The Internet is not the best medium for real time communications.
Individual packets can take different routes and varying delays can be
encountered and packets lost in transit. Waiting for delayed packets or
retransmission of lost packets can result in considerable degradation of
quality. Long delays in transit can affect quality so much that the technology
can become unusable, though many vendors do have solutions which aim
to negate the degradation suffered due to transit delays.

While some standards have been set by the ITU, the technology is
not fully standardized and there is no guarantee that products from different
vendors will be interoperable. Some vendors are trying to resolve this
problem by forming groups and making guarantees about the products in the
group but this is only a partial solution - vendors outwits the group cannot
guarantee interoperability.

Heavy congestion on the network can result in considerable

degradation of service as IP is not good at providing QoS (Quality of
Service) guarantees. Feedbacks to Lucent Technologies customers reflect
this worry. Major companies are planning to install IP Telephony capabilities
at some point and have carried out initial investigations.

Since only one physical network for both data and voice/fax transmissions is
required, failure of the network could be catastrophic, as all communications
capabilities are lost.

Opportunities

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Many vendors offer the ability to incorporate Virtual Private Networking
(VPN) with relative ease into the IP Telephony solutions they provide. This
allows any transmission to be encrypted using a number of cryptographic
techniques and providing security by transmitting the communications
through a 'tunnel' which is set up using PPTP (Point-to-Point Tunneling
Protocol) before commencing communications.

IP Telephony allows companies to exploit Computer Telephony Integration to

its full extent.

The convergence of communications technologies allows greater control over

communications; most vendors provide logging and accounting facilities
whereby all usage can be monitored.

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Conclusion :
Without a doubt, the data revolution will only gain momentum in the coming
years, with more and more voice traffic moving onto data networks. Vendors
of voice equipment will continue to develop integrated voice and data
devices based on packetized technology. Users with ubiquitous voice and
data service integrated over one universal infrastructure will benefit from
true, seamless, transparent interworking between voice and all types of
data.

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 BIBLIOGRAPHY

1. Computer Networks by Andrew S.Tanenbaum

2. Internetworking with TCP/IP by Douglas

E.comer

3. WWW.WIKIPEDIA.COM

3. www.iec.org.com

4. www.telogy.com

5. www.rad.com

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