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== Session Initiation Protocol ==
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'''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.
  
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SIP 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.
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== Introduction ==
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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 [[licenses|license]].
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SIPd is act as a gateway between the AlphaNet protocol and SIP. The SIPd enables calls between AlphaCom stations and SIP phones.
  
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== Configuration Guides ==
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== AlphaCom - SIP integration ==
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The AlphaCom has successfully been integrated with the following products (the list is not complete):
  
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=== SIP Gateways to analog FXO lines ===
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=== SIP Gateways ===
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* [[Configuration guide for AudioCodes MP114/118, v5.4 and higher|AudioCodes MP114/118 (analog lines)]]
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* [[Configuration guide for Mediatrix 1204|Mediatrix 1204 (analog lines)]]
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* [[SIP2SIP Transcoding using Mediant 600|AudioCodes Mediant 600]]
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* [[Configuration guide for AudioCodes Mediant 1000|AudioCodes Mediant 1000]]
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* [[Configuration guide for Mobile VoIP|Mobile VoIP MV-370 (GSM)]]
  
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[[AudioCodes MP114/118]]
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=== iPBX (SIP Trunk) ===
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[[Mediatrix 1204]]
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* Simens HighPath 4000
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* Open Scape 4000
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* Simens HighPath 8000
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* Alcatel OmniPCX OXO
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* Alcatel-Lucent OmniPCX OXE
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* Nortel CS1000
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* Asterisk Phone System
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* Aastra MX-ONE
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* 3CX Phone System
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* [[Cisco Call Manager 4.1 configuration guide|Cisco Call Manager 4.1]]
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* [[Cisco Call Manager 6 configuration guide|Cisco Unified Call Manager 6]]
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* [[Cisco Call Manager 8 configuration guide|Cisco Unified Call Manager 7 & 8]]
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* [[Cisco Call Manager 10, 11 and 11.5 configuration guide|Cisco Call Manager 10, 11 and 11.5]]
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=== SIP Phones ===
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* [[Configuration guide for Grandstream GXP2000|Grandstream GXP2000]]
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* [[Configuration guide for Grandstream BudgeTone 100|Grandstream BudgeTone 100]]
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=== SIP WiFi Phones ===
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* [[Configuration guide for UTStarcom F1000G WiFi phone|UTStarcom F1000G]]
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=== SIP Analog Telephone Adapters ===
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* [[Configuration guide for Grandstream HandyTone-488|Grandstream HandyTone-488]]
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=== Softphones ===
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* [[Configuration guide for X-Lite|X-Lite]]
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* [[Configuration guide for SJphone|SJphone]]
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=== IP-DECT systems ===
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* [[Configuration guide for STENTOFON IP-DECT|STENTOFON IP-DECT]]
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* [[Configuration guide for Ascom IP-DECT|Ascom IP-DECT]]
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== SIP Stations with iPBX ==
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The STENTOFON SIP Stations have successfully been integrated with the following products (the list is not complete):
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* Alcatel-Lucent OmniPCX OXE
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* Cisco Call Manager
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* Avaya iPBX
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* [[Avaya-Nortel CS1000 v. 7.0]]
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* Asterisk Phone System
  
 
== Related standards ==
 
== Related standards ==
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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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== Enable SIP INFO in X-Lite 3.0 ==
 
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AlphaCom use SIP INFO type signaling during connection. The X-Lite has DTMF support for SIP INFO messages, in addition to RFC 2833 and inband DTMF. But in order to use SIP INFO you need to disable the other two options.
 
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Follow these steps on the X-Lite:
 
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Dial ***7469 (send)
 
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Once you get the advanced options menu up, filter first for DTMF.
 
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Find the Force Inband DTMF entry, double click and change the value to 0.
 
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Then filter for 2833. Find the RTP 2833 enabled line, double click and change this value to 0.
 
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Setting both to 0 will result in your sending SIP INFO DTMF.
 
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Now * and # (and door opening) will work from X-Lite.
 
  
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[[Category:AlphaCom E Software]]
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[[Category: AMC Software]]
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[[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.

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

iPBX (SIP Trunk)

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):

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.

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