Difference between revisions of "SIP"
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RTP (Real-time Transport Protocol) is the most usual method for encapsulating audio into IP packets. Further reading here, http://en.wikipedia.org/wiki/Real-time_Transport_Protocol. And the standards document is http://tools.ietf.org/html/rfc3550. In AlphaCom, all voice over IP uses RTP, both in internal connections with proprietary call protocols, as well as for SIP calls. RTP mainly deals with handling the sequencing and timing of packets, not so much about the actual audio or video contents. However, a number of very common audio codes formats are listed and enumerated in http://tools.ietf.org/html/rfc3551. The number behind the "RTP/AVP" often found in SDP messages refer to [http://tools.ietf.org/html/rfc3551#page-33 table 4] in this document. These are called "static payload types". When they are used, it is possible to playback the stream of RTP packets without having any other information. On the other hand, when using "dynamic payload types", the meaning of the RTP packets has to be agreed upon using an other protocol, like SIP/SDP. | RTP (Real-time Transport Protocol) is the most usual method for encapsulating audio into IP packets. Further reading here, http://en.wikipedia.org/wiki/Real-time_Transport_Protocol. And the standards document is http://tools.ietf.org/html/rfc3550. In AlphaCom, all voice over IP uses RTP, both in internal connections with proprietary call protocols, as well as for SIP calls. RTP mainly deals with handling the sequencing and timing of packets, not so much about the actual audio or video contents. However, a number of very common audio codes formats are listed and enumerated in http://tools.ietf.org/html/rfc3551. The number behind the "RTP/AVP" often found in SDP messages refer to [http://tools.ietf.org/html/rfc3551#page-33 table 4] in this document. These are called "static payload types". When they are used, it is possible to playback the stream of RTP packets without having any other information. On the other hand, when using "dynamic payload types", the meaning of the RTP packets has to be agreed upon using an other protocol, like SIP/SDP. | ||
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| − | [[Category:AMC Software]] | + | [[Category: AMC Software]] |
| − | [[Category:AlphaCom - SIP Integration]] | + | [[Category: AlphaCom - SIP Integration]] |
Latest revision as of 14:09, 24 October 2024
SIP - Session Initiation Protocol is the call control protocol used for non-proprietary internet telephony. http://en.wikipedia.org/wiki/Session_Initiation_Protocol gives a brief overview of the protocol. http://tools.ietf.org/html/rfc3261 is the standards document. SIP is a human readable text protocol, which makes it a bit easer to debug.
Contents
Introduction
AlphaCom supports a subset of the SIP (Session Initiation Protocol). The AlphaCom SIP interface is implemented as a daemon service, SIPd. SIPd runs on the AMC-IP board alongside the main AlphaCom daemon, AMCd. The SIPd process is always started on AMC-IP, but the actual use require purchase and installing proper SW license. SIPd is act as a gateway between the AlphaNet protocol and SIP. The SIPd enables calls between AlphaCom stations and SIP phones.
AlphaCom - SIP integration
The AlphaCom has successfully been integrated with the following products (the list is not complete):
SIP Gateways
- AudioCodes MP114/118 (analog lines)
- Mediatrix 1204 (analog lines)
- AudioCodes Mediant 600
- AudioCodes Mediant 1000
- Mobile VoIP MV-370 (GSM)
iPBX (SIP Trunk)
- Simens HighPath 4000
- Open Scape 4000
- Simens HighPath 8000
- Alcatel OmniPCX OXO
- Alcatel-Lucent OmniPCX OXE
- Nortel CS1000
- Asterisk Phone System
- Aastra MX-ONE
- 3CX Phone System
- Cisco Call Manager 4.1
- Cisco Unified Call Manager 6
- Cisco Unified Call Manager 7 & 8
- Cisco Call Manager 10, 11 and 11.5
SIP Phones
SIP WiFi Phones
SIP Analog Telephone Adapters
Softphones
IP-DECT systems
SIP Stations with iPBX
The STENTOFON SIP Stations have successfully been integrated with the following products (the list is not complete):
- Alcatel-Lucent OmniPCX OXE
- Cisco Call Manager
- Avaya iPBX
- Avaya-Nortel CS1000 v. 7.0
- Asterisk Phone System
Related standards
SIP is a 270 page document which mainly deals with locating the destination "user" when making a call, and provides a framework for negotiating the media setup. Describing the media (audio/video) formats and protocols is left to another protocol: SDP (Session Description Protocol). SDP messages are attached to some of the SIP messages. See http://en.wikipedia.org/wiki/Session_Description_Protocol for a introduction. http://tools.ietf.org/html/rfc4566 is the standards document. By current practice, SIP and SDP is almost always used together.
RTP (Real-time Transport Protocol) is the most usual method for encapsulating audio into IP packets. Further reading here, http://en.wikipedia.org/wiki/Real-time_Transport_Protocol. And the standards document is http://tools.ietf.org/html/rfc3550. In AlphaCom, all voice over IP uses RTP, both in internal connections with proprietary call protocols, as well as for SIP calls. RTP mainly deals with handling the sequencing and timing of packets, not so much about the actual audio or video contents. However, a number of very common audio codes formats are listed and enumerated in http://tools.ietf.org/html/rfc3551. The number behind the "RTP/AVP" often found in SDP messages refer to table 4 in this document. These are called "static payload types". When they are used, it is possible to playback the stream of RTP packets without having any other information. On the other hand, when using "dynamic payload types", the meaning of the RTP packets has to be agreed upon using an other protocol, like SIP/SDP.