Re: [Freeswitch-users] Codec negotiation
You have a few options: Early Negotiation (default behaviour): *) When leg A calls FS the codecs will be compared against the configured codecs in the profile config. *) When B sends the invite, the chosen codec in leg A will be pushed to the top of the list regardless of if it's actually chosen in the config. (the other codecs offered to leg A will not appear) Late Negotation (requires param): <param name="inbound-late-negotiation" value="true"/> *) The call will hit the dialplan without looking at the codecs at all. *) The negotiation will take place when leg A is answered. *) This allows you to route the call to a script and examine the sdp and rewrite the acceptable codecs with the "codec_string" channel variable. Early Negotiation + Disable Transcoding: <param name="disable-transcoding" value="true"/> If this sofia profile param is set, all B legs that you invite will contain *only* the codec negotiated by the A leg. Also in either case the variable "codec_string" on the A leg controls what codecs will be offered to the B leg. The variable "absolute_codec_string" is similar except it implies the implicit list of codecs and will disable the addition of the A leg codec to the list as described in the defualt behaviour. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: James H Thompson <jht@lj.net> To: freeswitch-users@lists.freeswitch.org Sent: Monday, April 2, 2007 6:20:50 PM Subject: [Freeswitch-users] Codec negotiation I wondering how FreeSwitch does codec negotiation in the following call flow: SIP Endpoint A ----> FreeSwitch ----> SIP Endpoint B For example, if the SIP Endpoint A offers Codecs: G.711u, G.711a, G.729 when FreeSwitch sets up the Call to SIP Endpoint B, how does it determine what list of Codecs to offer to SIP Endpoint B? What I'm looking for is for Freeswitch to always offer the same list of Codecs to SIP Endpoint B as it received from SIP Endpoint A. In this application, having the RTP packets bypass FreeSwitch and go directly between SIP Endpoint A and SIP Endpoint B is not an option since the two SIP Endpoints cannot talk to each other directly. SIP Endpoint A is in public IP space, and SIP Endpoint B is in private IP space. _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Looking for earth-friendly autos? Browse Top Cars by "Green Rating" at Yahoo! Autos' Green Center. http://autos.yahoo.com/green_center/
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Anthony Minessale