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Introduction of VOIP

Internet Telephony:" In the beginning, Internet telephony simply meant the
technology and techniques to let you make voice phone calls local, long
distance, and international over the Internet using your PCthe definition
of Internet telephony is broadening day by day to include all forms of media
(voice, video, image), and all forms of messaging and all variations of speed
from real-time to time-delayed.

IP Telephony: As defined by Microsoft) IP Telephony is an emerging set of
technologies that enables voice, data, and video collaboration over existing
IP-based LANs, WANs and the Internet. Specifically, IP Telephony uses open
IETF and ITU standards to move multimedia traffic over any network that
uses IP (the Internet Protocol).

Voice over IP (VoIP): The technology used to transmit voice conversations
over a data network using the Internet Protocol. Such data network may be
the Internet or a corporate Intranet, or managed networks typically used by
long and local service traditional providers and ISPs that use VoIP.
What is VOIP?
VoIP is a term used in IP telephony to describe a set of facilities for
managing the delivery of voice information using the Internet

This means sending voice information in digital form in discrete
packets rather than in the circuit committed protocols of the
Public Switched Telephone Network (PSTN).
Understanding VOIP
VoIP stands for Voice over the Internet Protocol and is also referred to as IP
telephony. This technology enables users to :

Make calls by moving packets of information over the Internet using broadband
network connection to make phone calls to other VoIP and regular phones users
rather than traditional public switched telephony circuits also known as Public
Switched Telephone Network (PSTN) or analog telephones

VOIP Overview
IP Telephony is a process that enables the
transfer of voice data over a packet switched
network as opposed to the traditional circuit
switched network.

The transmission of voice packets over the
Internet Protocol (IP) is known as VoIP and is
implemented properly holds the promise of
converged networks and unified
VOIP Network
Major components of the VOIP network include:

Signaling Protocols and Standards: Initiate and control communications of
voice, video, and data.
- H.323: Network, platform, and application independent standard that
allows interoperability between H.323-compliant devices.
- SIP: Protocol that is based upon request-response or INVITE model. SIP is
used to establish conferencing, telephony, multimedia, and other Internet
communication sessions. SIP uses a media-description language, such as
Hypertext Transfer Protocol (HTTP).
Media Gateway:
- Converts one media stream to another; for example, converts voice
packets to analog.
- Can interact with call controllers, proxies, and soft switches via
proprietary or standard protocols such as SIP.
H.323 Gateway:
- Transforms audio received from a telephone device or telecommunications
system into a format that the data network can use.
- Acts as a bridge to the IP network.
- Generally has built-in intelligence to select the voice compression Codec's and
adjust the protocols and timing between two dissimilar computer systems or
voice over data networks.

Call Server:
- Receives call setup request messages.
- Determines the status of destination devices and checks the authorization of
users to originate or receive calls.
- Creates and sends the necessary messages to process the call requests.
- Provides call control, media access, and bandwidth management
between endpoints.
- Performs address translation, admissions control, bandwidth
management, and zone control.
- Coordinates access to other servers and manages call routing.
- Receives requests from clients, determines the destination server that it
needs to communicate with, and coordinates access with that server.
- Maps destination telephone number to destination endpoint IP address.
IP Terminals and Clients:
- Include end points on the network, such as hard or soft telephones
(portable or stationary) and wireless devices (802.11a and 802.11b).
- Bring voice and data communications to the end user. A Call Server
completes the call processing.
IP Backbone Network:
- Provides the universal communication language and foundation to
allow dissimilar networks and equipment from a variety of vendors to
Recommendation published by ITU
Ties together a number of protocols to allow multimedia transmission
through an unreliable packet-based network
1996: approved by ITU
2003: Version 5
H.323 Architecture

H.323 Protocol Stack for VoIP
H.323 Protocol Suite
Video Audio Data Transport
G.7xx Speech (De)Coding
H.323 systems must support G.711: PCM, 64kbps
Other codecs: G.729, G.726,

Realtime Transport Protocol
(RFC 3550, July 2003)
Application layer protocol for transmitting realtime
data (audio, video, ...)
Includes payload type identification, sequence
numbering, timestamping, delivery monitoring
Mostly over UDP
Supports multicast & unicast

Control Protocol - RTCP
RTP Control Protocol (RFC 3550, July 2003)
Periodic transmission of control packets to all
participants in the session
Main functions:
- provide feedback on quality of data distribution
- carries a persistent transport-level identifier for an
RTP source (CNAME)
- each participant sends control packets to all others
which independently observe the number of
More Control Protocols in H.323
H.225 (RAS)
- protocol between terminal and gatekeeper (if
- allows terminals to join/leave zone, request/return
bandwidth, provide status updates,
H.245 (Call Control)
- Media Control Protocol
- Allows terminals to negotiate connection
parameters (codec, bit rate, ..)
Q.931 (Call Signalling)
- Manages call setup and termination
SIP Session Initiation Protocol
Developed by IETF since 1999
RFC 2543, March 1999 (obsolete)
RFC 3261, June 2002
Target: develop simpler and more modular protocol for
VoIP than the large and complex H.323 by ITU
SIP is a text-based protocol similar to HTTP and SMTP,
for initiating interactive communication sessions between
SIP is an application-layer control (signalling) protocol
for creating, modifying and terminating sessions with one
or more participants
Sessions include Internet Multimedia conferences,
Internet Telephone calls and Multimedia distribution

SIP (2)
SIP can be used with different transport protocols, it
doesn't even require reliable transport protocols
A simple SIP client can be implemented using only UDP
SIP (3)
Comparison of H.323 and SIP
Comparison of H.323 and SIP (2)
Selecting the appropriate speech CODEC is essential. CODEC performance
includes the baseline quality (that is, without impairments) and the performance
with impairments present, such as background noise and lost or late packets.
The table below shows some CODECs that are used for voice traffic. Bandwidth
requirements are estimates.
CODEC Selection
It is important to select a CODEC that meets the bandwidth and voice quality
G.711 is the preferred choice when bandwidth and cost are not an issue and is
generally the default CODEC for Local Area Networks (LANs) because G.711 does
not compress the audio.

G.729 A/B is generally the default CODEC for Wide Area Networks (WANs)
because it requires less bandwidth than G.711 and delivers near-toll voice
quality; for example, a G.729 A/B with a 30 ms sample size is an effective
technique to reduce bandwidth limitations, while delivering acceptable voice

G.723.1 is used when bandwidth, not voice quality, is the customers objective.
Speech CODECs (compression algorithms), such as G.729 A/B and G.723.1, are
designed to reduce the bandwidth required; however, when using these CODECs,
consider parameters such as end-to-end delay (latency) and distortion in voice
quality. Although the G.726 CODEC has less processing delay than the G.729 A/B
CODEC, G.729 A/B is generally used for VoIP because it delivers better voice quality.

Tip: Remember, when selecting a CODEC, consider the customers
voice quality targets and bandwidth requirements. For example, the
G.723.1 CODEC might be appropriate if bandwidth consumption, not
voice quality, is the primary consideration. However, if voice quality,
not bandwidth consumption, is the primary consideration, another
CODEC, such as G.711 is appropriate. If bandwidth consumption AND
voice quality are important, consider G.729 A/B.
VoIP & QoS

Voice quality characteristics
- Clarity: fidelity, clearness, and intelligibility of signal
- Delay: effect on interactivity
- Echo: distracting and confusing

- Components: Encoding, Packetisation, Network delay,
Receiver buffering, Decoding
- ITU-TG.114 recommends 150ms
One-way Delay Effect on perceived Quality
<100 -150ms Delay not detectable
150 -200ms Acceptible quality; slight delay or hestitation noticeable

Over 200 -300ms Unacceptible delay; normal conversation impossible
- Smoothed by playback buffers
- Receivers adapt the depth of these
-Sudden changes in jitter may cause loss

Figure: Playback buffer
- Generally modest (64 kbps or less)
- Depends on codec and use of silence
Codec Rate (kbps)
G.729 (A/B) 8
G.722 48-64
G.711 64
Packet loss
- Should be less then 5%