Attila, Thanks again. Is it also the case that SIP UAs will start off by sending the first codec listed in the 200 OK? If so, then that, together with the RFC3264 RECOMMENDation you mention should be enough to guarrantee that my SIP UA gets away with it. (Because - between you and me - it can't do dynamic switching!)
Regards, Alex On Wed, 2003-09-17 at 11:21, Attila Sipos wrote: > No, there is no way to make the answerer select just one. > > However, I don't think this is an issue becaause this paragraph > in RFC3264 is adhered to: > > Although the answerer MAY list the formats in their desired order of > preference, it is RECOMMENDED that unless there is a specific reason, > the answerer list formats in the same relative order they were > present in the offer. In other words, if a stream in the offer lists > audio codecs 8, 22 and 48, in that order, and the answerer only > supports codecs 8 and 48, it is RECOMMENDED that, if the answerer has > no reason to change it, the ordering of codecs in the answer be 8, > 48, and not 48, 8. This helps assure that the same codec is used in > both directions. > > > So, I think you will be OK and both will select to use PCMU. > > Of course, if the answerer couldn't do PCMU then it would answer: > > > > 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 3 8 101 > > > > a=rtpmap:3 GSM/8000/1 > > > > a=rtpmap:8 PCMA/8000/1 > > > > a=rtpmap:101 telephone-event/8000 > > > > a=fmtp:101 0-11 > > This answer keeps the media formats in the same order > as the offer. Both ends would transmit GSM. > > Regards, > > Attila > > > > -----Original Message----- > > From: Alex Zeffertt [mailto:[EMAIL PROTECTED] > > Sent: 17 September 2003 11:12 > > To: Attila Sipos > > Cc: SIP-LIST > > Subject: RE: [Sip-implementors] which codec will peer send? > > > > > > 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
