VOIP 4 Mobile

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Comparison of VoIP software

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General softphone clients

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General softphone clients

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General softphone clients

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General softphone clients

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General softphone clients

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General softphone clients

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Mobile phones

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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.

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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.

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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.

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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.

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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:bob@somewhere.com

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

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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.

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