On Mon, Jan 4, 2010 at 1:46 PM, Kevin P. Fleming <kpflem...@digium.com>wrote:
> hadi motamedi wrote: > > > Sorry . I didn't get the point clearly . In the SIP Invite message , it > > says "my audio endpoint is IP x.x.x.x port x, and I can use codecs > > A,B,C". The remote endpoint responds with a 200 OK, saying "my audio > > stream is at IP y.y.y.y port y, and I choose codec B". Can you please do > > me favor and let me know if my understanding is right or not ? > > Thank you > > No, you are not understanding the SDP offer/answer model properly. If > one endpoint offers codecs A, B and C in its SDP, it is willing to > *receive* media in those formats. The receiver of that offer can choose > to send media to the offerer in any of those formats, at any time. If > the answering endpoint includes only codec B in its SDP, then it is > willing to *receive* only codec B. In that scenario, it is possible for > media to flow from endpoint 1 to endpoint 2 using codec B, and from > endpoint 2 to endpoint 1 using codec A (or C), but this will not happen > if Asterisk is an endpoint in this scenario. > > When Asterisk receives a media frame, if the format of that frame is not > the format that it is currently sending to the other endpoint, it will > switch to that format automatically. If it cannot do so because the > other endpoint did not offer to receive that format, then the call's > audio will probably fail. This is the reason why I responded before that > Asterisk does not support asymmetric formats in a media session. > > In reality, it is extremely uncommon for a SIP endpoint to want to send > media in a format that it is not also willing to receive; in fact, I > can't say I've ever seen this situation arise in any testing I've done > or in any issues reported in our issue tracker. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kpflem...@digium.com > Check us out at www.digium.com & www.asterisk.org > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > Thank you very much for correcting me .
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