SEMINAR REPORT

ON
“VOICE OVER INTERNET PROTOCOL”

CONTENTS

1) INTRODUCTION

2) WHAT IS VoIP ?

3) REQUIREMENT OF VoIP

4) HOW VoIP WORKS ?

a) How VoIP works part 1

b) How VoIP works part 2
c) How VoIP works part 3
d) How VoIP works part 4

5) PROS AND CONS OF VoIP

6) CONCLUSION
 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).

During the early 90's the Internet was beginning its commercial spread.
The Internet Protocol (IP), part of the TCP/IP suite (developed by the
U.S. Department of Defense to link dissimilar computers across many
kinds of data networks) seemed to have the necessary qualities to
become the successor of the PSTN.
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.

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.
The gateway allows the calls to transfer from one network to the other
by converting the incoming signal into the type of signal required by the
network it is required to send it on. For example, A PC user wishes to
call someone using a conventional phone. The PC sends the IP packets
containing digitized voice to the gateway.

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:

Software Requirements
Hardware Requirements
Protocol 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:

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.

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.

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 standardised 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.
How VoIP works
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, 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 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.
Main ‘types’ 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.
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.
.

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.

So what is 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.

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.
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.
I have already talked about the encoders in the previous
section. I also bundle into the encoding process echo suppression. 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.
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

IP - the underlying protocol - is supported by most platforms and is

independent of the transport protocol used.
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.

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
outwith 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. Feedback 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,
however:
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

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.

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.

REFRENCES:
1. Computer Networks by Andrew S.Tanenbaum
2. Internetworking with TCP/IP by Douglas E.comer
3. www.iec.org.com
4. www.telogy.com
5. www.rad.com
6. www.mailto:blazer@gslis.utexas.edu