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VOIP 4 Mobile

This document compares VoIP software and protocols for mobile phones. It discusses general softphone clients, mobile phone frameworks and libraries, audio encoding and transmission using RTP over UDP, and signaling systems including H.323 using gatekeepers and SIP using proxy servers. An example SIP call session is provided to illustrate the signaling process.

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Ahmed Hesham
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51 views22 pages

VOIP 4 Mobile

This document compares VoIP software and protocols for mobile phones. It discusses general softphone clients, mobile phone frameworks and libraries, audio encoding and transmission using RTP over UDP, and signaling systems including H.323 using gatekeepers and SIP using proxy servers. An example SIP call session is provided to illustrate the signaling process.

Uploaded by

Ahmed Hesham
Copyright
© Attribution Non-Commercial (BY-NC)
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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VOIP 4 Mobile

1
Comparison of VoIP software

2
General softphone clients

3
General softphone clients

4
General softphone clients

5
General softphone clients

6
General softphone clients

7
General softphone clients

8
Mobile phones

9
Frameworks and libraries

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Encoding and Transmission
protocols

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Encoding and Transmission

 Audio is encoded using a well-known standard


such as Pulse Code Modulation (PCM).

 Audio is transferred using the Real-time


Transport Protocol (RTP).

 RTP message is encapsulated in a UDP


datagram that is further encapsulated in an IP
datagram for transmission.

12
Encoding and Transmission
 UDP is used for transport because
 lower overhead: audio must be played as it
arrives.
 Playback cannot be stopped to wait for a
retransmitted packet.

 Two independent RTP sessions exist,


because an IP phone call involves transfer
in two directions
 IP phone acts as sender for outgoing data, and
 IP phone acts as receiver for incoming data.

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Signaling Systems & Protocols

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Signaling Systems & Protocols
 Main complexity of VoIP: Call setup and call
management.

 The process of establishing and terminating a call is


called Signaling.

 In traditional telephone system, signaling protocol is SS7


(signaling System 7).

 In VoIP, signaling protocols are:


○ SIP (Session Initiation Protocol), by IETF
○ H.323, by ITU
○ Megaco & MGCP, jointly by IETF and IUT.
 VoIP signaling protocols should be able to interact with SS7 if we
will use traditional telephone system.

15
Signaling Protocols
 Two major protocols: H.323, SIP

 H.323, invented by ITU, defines four elements


that comprising a signaling system:
 Terminal: IP phone
 Gatekeeper: provides location and signaling
functions; coordinates operation of Gateway.
 Gateway: used to interconnect IP telephone system
with PSTN, handling both signaling and media
translation.
 Multipoint Control Unit: provides services such as
multipoint conferencing.

16
Signaling Protocols
 SIP: Session Initiation Protocol. Invented
by IETF.
 SIP defines three main elements that
comprise a signaling system:
 User Agent: IP phone or applications
 Location servers: stores information about
user’s location or IP address
 Support servers:
○ Proxy Server: forwards requests from user agents to
another location.
○ Redirect Server: provides an alternate called party’s
location for the user agent to contact.
○ Registrar Server: receives user’s registration requests
and updates the database that location server consults.

17
H.323 Characteristics
 H.323 consists of a set of protocols that work
together to handle all aspects of
communication, including:
 Transmission of a digital audio phone call
 Signaling to set up and manage phone call
 Allows transmission of video and data while a phone
call is in progress
 Sends binary message
 Incorporates protocols for security
 Uses a special hardware Multipoint Control Unit for
conferencing calls
 Defines servers for address resolution, authentication,
accounting, features, etc.

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H.323 Layering
• H.323 uses both UDP and TCP over IP.
– Audio travels over UDP
– Data travels over TCP

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SIP Characteristics
 Operates at the application layer.

 Encompasses all aspects of signaling, e.g. location of


called party, ringing a phone, accepting a call, and
terminating a call.

 Provides services such as call forwarding.

 Relies on multicast for conference calls.

 Allows two sides to negotiate capabilities and choose


the media and parameters to be used.

 SIP URI is similar to email address. (with prefix “sip:”)


E.g. sip:[email protected]

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SIP Methods
 Six basic message types, known as
methods:

21
An Example SIP Session
 User agent A contacts DNS
server to map domain name in
SIP request to IP address.

 User agent A sends a INVITE


message to proxy server that
uses location server to find
the location of user agent B.

 Call is established between A


and B. Then media session
begins.

 Finally, B terminates the call


by sending a BYE request.

22

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