I too agree to Attila, who knows RFC 3264 very well. But I also know, that many UA implementations do not follow the recommended behavior. So the UAC often ends up with multiple choices in the SDP answer even if no dynamic switching will be used. > and I can't see how the UA knows which one the other end will start > sending!
The only way then to see, what codec is actually used is to look into the RTP-header of the packets. Franz -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Attila Sipos Sent: Mittwoch, 17. September 2003 11:53 To: 'Alex Zeffertt'; SIP-LIST Subject: RE: [Sip-implementors] which codec will peer send? In the SDP response, one should only list multiple formats if you can support dynamic switching between them. I get this from top of page 22 RFC3264. > Bob can support dynamic switching between PCMU and G.723. So, he > sends the following answer: Can the second UA really support dynamic switching between all its listed media formats? If so, it doesn't matter what you transmit to the second UA. I'm guessing but it is likely that, since both lists have PCMU first (payload 0), then PCMU will be chosen. > -----Original Message----- > From: Alex Zeffertt [mailto:[EMAIL PROTECTED] > Sent: 17 September 2003 10:46 > To: SIP-LIST > Subject: [Sip-implementors] which codec will peer send? > > > Hi all, > > I have a SIP UA implementation which seems to choose the codec to use > for rx and tx at the point at which the INVITE is accepted. This > confuses me because the SDP offer and response contain > multiple codecs, > and I can't see how the UA knows which one the other end will start > sending! > > Can anybody explain this to me? > > Here's the detail: > > The first SIP UA sends the second the following SDP offer in a SIP > INVITE: > > v=0 > o=vssip 123456 654321 IN IP4 10.0.0.228 > s=A conversation > c=IN IP4 10.0.0.228 > t=0 0 > m=audio 7078 RTP/AVP 0 3 8 101 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:3 GSM/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > Then the second sends the first the following SDP response in it's SIP > 200 OK message: > > v=0 > o=vssip 123456 654321 IN IP4 10.0.0.107 > s=A conversation > c=IN IP4 10.0.0.107 > t=0 0 > m=audio 7078 RTP/AVP 0 3 8 101 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:3 GSM/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-11 > > So both know that they both support ulaw (0), gsm (3), and > alaw(8). But > how do they know which codec the other UA will start sending? > > Regards, > > Alex > > _______________________________________________ > Sip-implementors mailing list > [EMAIL PROTECTED] > http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors > _______________________________________________ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors _______________________________________________ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
