SEMINAR TOPIC

ON
VOICE OVER INTERNET PROTOCOL
PRESENTED BY:
PRAKASH RANJAN
MECE-170-2K10

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 OUTLINE
INTRODUCTION
WHAT IS VOIP?
HISTORY
HOW VOIP WORKS?
RQUAREMENTS OF VoIP
CALL SET UP PROCESS
VoIP OVERALL TECHNIQUE
PROTOCLOS H323 AND SIP
ADVANTAGES
LIMITATIONS
APPLICATIONS
CONCLUSION

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 INTRODUCTION
 The traditional method used for transmitting
voice is PSTN.

 Now more and more communication is done in

digital format and transported via data networks
such as internet.

 As data traffic is much faster than telephone

traffic, so we prefer to send voice over data
networks.
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 WHAT IS VOIP ?
 Voice Over Internet protocol (VoIP) is a
protocol used for the transmission of voice
and fax phone calls through the Internet
or other packet-switched networks.

It is also called IP telephony, Internet

telephony, voice over broadband,
broadband telephony.
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 VoIP
 It is a bunch of protocols and standards; and years of
research done by organizations all over the world that would
bring about this revolution. They call it ‘VOICE OVER IP’
 In the near future, if you make a telephone call, it is more
than likely that it would be over the Internet or some other
packet network.
 With the passes of time VoIP replace the PSTN and people
will acquaint with the new technology .
 VoIP Internet Telephony simply takes advantage of the
efficiencies of packet switched networks.

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 HISTORY OF VOIP

• The first Voip software was

introduced in 1995 by Vocaltec.
Designed for home PC
Uses H.323 Protocol
Uses Packet Switching

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CIRCUIT SWITCHING PACKET SWITCHING

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 How VoIP WORKS
• Part 1:- Technical over view

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 Protocols and Encoding
• Part 2 : The Protocols
 H 323 Protocol
 SIP Protocol
• Part 3 : Encoding
 G.723.1 audio signal with a bandwidth of 3.4KHz should
be encoded for transmission at data rates of 5.3Kbps and
6.4Kbps
 G.711 ITU standardised PCM (Pulse Code Modulation) as
G.711. audio signals to be encoded for transmission at
data rates of 56Kbps or 64Kbps

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 Jitter and packet loss
• Part 4 : Hear the Quality
 The performance or quality of the speech depends up on
encoders at each end, the number of packets lost on route,
Latency and Jitter.
 5% lost packets distributed evenly will not result in the loss of
words lose.
 but lose in 5 a row the entire word is lost and this will afect
significantly.
 Quality also vary with manufacturer .
 Overall quality is good if we have good quality of network.
 there is a trade off between call quality and bandwidth.

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 Requirements of a VoIP

• Software Requirements:- User interface, Voice

Processing Module, Echo Cancellation, Idle Noise Detection
• Hardware Requirements:- existence (or installation)
of an IP based network.
 gateway is required to bridge the differences between packet
switching and other networks.
• Protocol Requirements :- H.232, Q.391,
H.245,SIP, RTP, RTCP, with TCP & UDP DLL
protocols.

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Need of Gateway for other n/w

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 CALL SET UP PROCESS

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 VoIP OVERALL Technique
Audio Codecs (G.723.1, G711)
Data Transport (RTP, RTCP)

 Addressing:-
 Here phone no. is converted to an IP address.
• Simple format addresing:
<user | phone number>@<domain | hostname | IP
address>

 Signaling (SIP, H.323)

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 H.323
• Recommendation published by ITU in 1996.
• It is designed to act above transport layer and is mainly used for
transmission of voice, data and video conferencing over packet
networks.

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 Terminals
• User End point.
• either be a personal computer (PC) or a stand-alone device,
running an H.323 & the multimedia applications.

• Function of the Terminal:

 Provides real-time 2 way communication.
 Optional : Video & data streaming.
 Mandatory voice streaming

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 Gateways
• A gateway connects two dissimilar networks.
• An H.323 gateway provides connectivity between an H.323
network and a non-H.323 network.

• The Functions of the gateway can be stated as follows:

 Task-translation.
 Audio Codec
 Video codec
 H.245  H.221 (ISDN Conference)
 H.245  H.242 (Audio-Visual terminals)

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 Gatekeepers
• A gatekeeper can be considered the brain of the H.323
network.
• It is the focal point for all calls within the H.323 network .
• Although they are not required, gatekeepers provide
important services such as
addressing, authorization and authentication of
terminals and gateways; bandwidth management;
accounting; billing; and charging.
• Gatekeepers may also provide call-routing service.

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 Multipoint Control Unit
• MCU’s provide support for conferences of three or more
H.323 terminals.
• All terminals participating in the conference establish a
connection with the MCU.
• The functions of the MCU is to maintain all audio, video
data & control streams mandatory for conferences.

 The gatekeepers, gateways and MCU’s are logically separate

components of the H.323 standard but can be implemented
as a single physical device.

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 SIP
• SIP stands for Session Initiation Protocol.
• Developed by IETF since 1999.
• It is an text-based application-layer control protocol for creating,
modifying and terminating sessions with one or more participants.
• Sessions include Internet Multimedia conferences or Internet
Telephone calls.

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 SIP LOGICAL ENTITIES
• User Agent (UA):-
• In SIP, a User Agent (UA) is the endpoint entity.
• Proxy Server
• A Proxy Server is an intermediary entity that acts as both a
server and a client for the purpose of making requests on
behalf of other clients.
• Redirect Server
• A Redirect Server is a server that accepts a SIP request.
• Registrar
• A Registrar is a server that accepts REGISTER requests for the
purpose of updating a location database .

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 ADVANTAGES
• Cheaper call rates, not depend on distance.
• Simplification
• High efficiency
• Free calls within organization .
• Better Voice Quality Using Wideband Codecs
• Adding new features and applications over time
is easy.
• Integration of voice, data, fax, video is possible.

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12/8/21 08:40:22 PM 23
 LIMITATIONS
 Packet Delay
Packet Loss (no guarantee of delivering packets)
Jitter (variable delay)

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 Security- The concern (Flaws)
• An interesting example of an identified Cisco VoIP
vulnerability  is shown below: -

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 Security- The counter measure
• These are some what analogues to general PC “Anti Virus”.
• One of the popular vender provides following solutions :-
» VoIP Vulnerability Assessment (VVA) and
penetration testing .
» VoIP Intrusion Prevention System (VIPS).

» VoIP PBX configuration management tool

enables PBX.
» VoIP PBX monitoring tool provides around the
clock monitoring  for hardware failures.

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 APPLICATONS
• Real time applications:
Telephony ( two-way)
 Radio-TV Broadcast (one-way)
• Non-real time applications:
 e_mail
OTHERS:
• Integration of data,voice and fax
• Video telephony
• Enhanced teleconferencing
• Used in Amateur Radio
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 Can we make VoIP through mobile?

 Yes . By two ways:-

 1. One of the ways to do that, offered by 
Rebtel - VoIP connection is established via a
local access number.
 2. Another way is to use special Internet
phone applications for mobile phones. The
most well known of such applications is fring.

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 CONCLUSION
● Like everything else, as the technology changes so at first
only a few companies like Cisco and Lucent offered VoIP
services, but the large telecommunications carriers – such
as AT&T and Sprint -- are catching on .
● This technology has enormous growth potential and it is
expected that it will be very popular in future with
advancement in network technology.
● The availability of high-quality audio using wideband
codecs, video conferencing, and document sharing
enables more effective and pleasant communication.
● “It is not Voice over IP, it is Everything over IP…”

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THANK YOU

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