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.
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.
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.
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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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
10 Encoding and Transmission protocols
11 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.
13 Signaling Systems & Protocols
14 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.
18 H.323 Layering • H.323 uses both UDP and TCP over IP. – Audio travels over UDP – Data travels over TCP
19 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
20 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.
