Searching for Asterisk Sip Protocol Support information? Find all needed info by using official links provided below.
https://issues.asterisk.org/jira/secure/attachment/28201/AsteriskSipSessionTimers.pdf
Asterisk to decide whether it wants to run Session-Timer for the session or not. If the Asterisk is configured in the “originate ” mode then Asterisk will run the Session-Timer even if the UAC does not support it. The Asterisk will act as a refresher and will refresh the session at the interval
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-UnderstandingVoIP-SECT-2.html
Asterisk does not currently have a TCP implementation for transporting SIP messages, but it is possible that future versions may support it (and patches to the code base are gladly accepted). SIP is used to “establish, modify, and terminate multimedia sessions such as Internet telephony calls.” [ 206 ]
https://wiki.asterisk.org/wiki/display/AST/Simple+Network+Management+Protocol+%28SNMP%29+Support
Aug 27, 2010 · SNMP support comes in two varieties – as a sub-agent to a running SNMP daemon using the AgentX protocol, or as a full standalone agent. If you wish to run a full standalone agent, Asterisk must run as root in order to bind to port 161. Configuring access when running as a full agent is something that is left as an exercise to the reader.
https://wiki.zenitel.com/wiki/Asterisk_configuration
Asterisk Configuration(CHAN_SIP) Configuration with UDP/TCP transport protocol and video support [general] context=default bindaddr=0.0.0.0 videosupport=yes port=5060 //Extension [5001] type=friend host=dynamic secret=password disallow=all allow=ulaw,alaw,g722,g729
https://community.cisco.com/t5/unified-communications/support-sip-cisco-ip-phone/td-p/3528937
Hi, The Cisco IP Phone 7911G/ 7942G / 7965G / 9951 / 9971 support SIP protocol. I can signaling phones with Asterisk solution???....What considerations should I have for the licensing of the Cisco IP Phone?? The price is only IP Phone???? Please, I need a confirmation.. Greeting
https://wiki.asterisk.org/wiki/display/AST/Interactive+Connectivity+Establishment+%28ICE%29+in+Asterisk
Feb 04, 2014 · Asterisk ICE support is enabled globally by default throughout Asterisk, but is disabled by default for chan_sip specifically, and can be enabled inside chan_sip both globally or on a SIP peer basis in sip.conf.
https://www.asterisk.org/get-started/features
VoIP Protocols. Google Talk H.323 IAX™ (Inter-Asterisk eXchange) Jingle/XMPP MGCP (Media Gateway Control Protocol SCCP (Cisco® Skinny®) SIP (Session Initiation Protocol) UNIStim. Traditional Telephony Protocols. E&M E&M Wink Feature Group D FXS FXO GR-303 Loopstart Groundstart Kewlstart MF and DTMF support Robbed-bit Signaling (RBS) Types ...
http://forums.asterisk.org/viewtopic.php?t=66972
I'm looking for Asterisk's RFC support for SIP and other (like RTP, RTCP, ..) but can not find a reference. The list of SIP RFCs is quite long, I assume that the following are fully supported: ... RFC 3311 The Session Initiation Protocol UPDATE Method (RFC 3311)
https://stackoverflow.com/questions/23193035/whats-the-supported-sip-encryption-protocols-by-asterisk
Asterisk supports encryption of the media in one of two ways. The first, supported in Asterisk 1.8 and later, is SDES-SRTP, via the libsrtp library. libsrtp uses AES as the default cipher. As SDES-SRTP has to exchange keys in plain text in the signalling, another method of encrypting the media is available in Asterisk 11 and later, DTLS-SRTP.
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