Presentation on

Voice over IP
Telecommunication and Computer Networks

Presenters:
Subash Chandra Pakhrin (072MSI616)
Sujiv Shrestha (072MSI617)
Sumir Maharjan (072MSI618)
MSC in Infromation and Communication Engineering
Pulchowk Campus 1
 Contents
Introduction
What is VoIP
How VoIP works
Major Components
VoIP Architecture
Signaling Protocols
Challenges
Security Issues

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 Introduction
Voice over Internet Protocol (VoIP) is a technology
that enables one to make and receive phone calls
through the internet instead of using the traditional
PSTN (Public Switched Telephone Network) lines.

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 What is VOIP
VoIP is a packetisation and transport of classic public
switched telephone system audio voice over an IP
network.
It allows 2-way voice transmission over broadband
connection.
It is also called IP telephony, internet telephony,
voice over broadband, broadband telephony.
It was developed in February of 1995 by a small
company in Israel called VocalTec.

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 PSTN vs. Internet
PSTN Internet
1. Voice network use circuit 1. Data network use packet
switching switching
2. Dedicated path between 2. No dedicated path between
calling and called party Sender and receiver
3. Bandwidth reserved in 3. It acquires and releases
advance bandwidth, as needed.
4. Cost is based on distance 4. Cost is based on time and
and time bandwidth

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 VoIP How does it work ?
Continuously sample audio
Converts each sample to digital form
Send digitized stream across Internet in packet
Converts the stream back to analog for playback

Analog to Depacketizer and

Voice Digital Voice
Internet Digital to Analog
(Source) Conversion and (Destination)
Conversion
Packetizer

VoIP Basic Principle

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 VoIP How does it work ?
1. Compression voice is compressed typically with
the codecs such as PCM, ADPCM, ACELP, etc
2. Encapsulation the digitized voice is wrapped in an
IP packet
3. Routing the voice packet is routed through the IP
network to its final destination.

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 Components

VoIP Codecs
VoIP Gateway
VoIP Protocols

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 VoIP Codecs
Codecs (COder DECoder) are used to convert between analog voice
signal and digitally encoded version.
Voice Bit Rate
Coding Algorithm Type of Codec
(kbits/s)
G.711 8-bit PCM 64
Waveform Coder
G.726 ADPCM 32
G.728 LD-CELP 16
Model based
G.729a CS-ACELP 8
Vocoding /
G.723.1 MPMLQ 6.3
Vocoders
G.723.1 ACELP 5.3
Trade-off between various attributes of speech coders such as bit rate,
algorithms processing delay, complexity and quality and depending
upon the applications, bandwidth available, one can have the choice of
a speech coders in a particular VoIP context.
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What Kind of Transport Protocol Used?
UDP (User Datagram Protocol), But Why not TCP
TCP Reliable Transport Mechanism
UDP Unreliable Transport Mechanism.
In real time communication like voice retransmission of packet
is not possible.
UDP has no control over the order in which packets arrive at
the destination or how long it takes them to get there.
Real-time Transport Protocol (RTP) solves the problem
enabling the receiver to put the packets back into the correct
order and not wait too long for packets that have either lost
their way or are taking too long to arrive.

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 VoIP Network Model
Layer Name of Layer Protocol
7 Application VoIP data, H.323, SIP
6 Presentation or MGCP
5 Session RTP
4 Transport UDP
3 Network IP
2 Link Frame (Ethernet, ATM..)
1 Physical Medium

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 VoIP Data Units

Fig. VoIP: Speech payload nested in a packet, with headers added by different
protocols (example for VoIP in an Ethernet-based LAN)
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VoIP Architecture

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 Signaling Protocols
Main complexity of VoIP : Call setup and call
Management
The process of establishing and terminating a call is
called Signaling.
In PSTN, signaling protocol is SS7(Signaling System 7)
In VoIP, most implemented signaling protocols are:
1. H.323 by ITU-T
2. SIP (Session Initiation Protocol) by IETF (RFC 2443)
3. MGCP (Media Gateway Controller Protocol) by Cisco
VoIP signaling protocols should be able to interact
with SS7.

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 Signaling Protocols
H. 323
Most widely used protocol
Provides specifications for real-time, interactive
videoconferencing, data sharing and audio applications (VoIP)
Logical components: Terminals, Gateways, Gatekeepers and
Multipoint control units (MCU)
Terminals: IP phone
Gatekeeper: provides location and signaling functions; coordinates
operation of Gateways.
Gateways: used to interconnect IP telephone system with PSTN,
handling both signaling and media translation.
MCU : provides services as multipoint conferencing.

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 Signaling Protocols / SIP
Session Initiation Protocol. Developed by IETF.
Three main elements that comprise a signaling system:
1. User Agent: IP phone or applications.
2. Location servers: stores information about users location
or IP address.
3. Support servers:
a. Proxy Server: forward requests from user agent to
another location.
b. Registrar Server: receives users registration request
and updates the database that location server consults.
c. Redirect Server: provides an alternate called partys location
for the user agent to contact.

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Schematic View of Different Server in
SIP

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VoIP SIP

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 SIP Messages
Messages are used for communication between the client and
the SIP server.
These messages are:

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 Comparison of H.323 with SIP
H .323 SIP
Complex Protocol Comparatively Simpler
Binary representation for its messages Textual representation
Requires full backward compatibility Doesnt require full backward
compatibility
Not very modular Very modular
Not very scalable Highly scalable
Complex signaling Simple Signaling
Large share of market Backed by IETF
Hundreds of elements Only 37 headers
Loop detection is difficult Loop detection is comparatively easy

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 VoIP Gateway
Also known as Media Gateway
This provides conversion between the audio signals carried
on telephone circuits and data packets carried over the
Internet or over other packet networks.
Perform the conversion between Time-division
multiplexing (TDM) voice to a media streaming protocol,
such as the Real-time Transport Protocol (RTP), as well as
signaling protocol used in the VoIP system.

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 Delay Jitter / How to Address It
Delay Smaller than 150ms are not perceived by human.
150 400 ms delay acceptable.
Delay Jitter : Due to random queuing delays.
Delay Jitter : Different packet will experience different delays.
If all packets are experiencing a same 150 ms delay than thats
not a problem.
Delay Jitter actually degrades the speech quality.
How to address delay Jitter in VoIP?
Answer: Use the play-out buffer in the Receiver (Rx).
Play-out Buffer (In Receiver):
In this buffer arriving packets are stored in Rx and these
packets are played out at an appropriate time.

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Fixed Play-out Buffer Algorithm

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 Security Risks
As VoIP uses the Internet, for example, it is vulnerable to the
same type as security risks
Hacking
Denial of Service
Eavesdropping
Most VoIP services do not support encryption
Further more : Lost or Delayed packets cause drop out in voice
(Addressed by Fixed Play-out buffer algorithm in receiver)

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VoIP Applications

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 References:
Freeman, Roger L. "Voice-Over IP." Telecommunication
System Engineering. New York: Wiley, 1996
Voice over IP Wikipedia :
https://en.wikipedia.org/wiki/Voice_over_IP
YouTube Video by Prof. Karandikar, IIT Bombay
VoIP Overview -
http://users.ecs.soton.ac.uk/dt302/guides/VOIP-Overview.pdf
IP Telephony (VoIP):
http://www.site.uottawa.ca/~bob/csi4118/notes/VoIP.ppt
Chapter 1: Working with VoIP - www.networkworld.com

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Thank You !

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