Attilla, Thanks for your reply. It makes sense to only put one codec in the 200 OK message, since you already know what the other end supports. But if I restrict myself to putting only one codec in the INVITE message, then there's a big chance the other end won't support it.
Ideally, what I want is to be able to advertise ALL the codecs I support in my INVITE, and somehow force the other end to respond with just one codec. Is this possible? Regards, Alex On Wed, 2003-09-17 at 10:53, Attila Sipos wrote: > 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
