Attila, Me again. I've just tested this RECOMMENDation against a couple of SIP UAs. linphone adheres to it, but - unfortunately - Asterisk PBX does not.
I wonder if there is any way to find out if Asterisk is a one off, or if there are other UAs that don't adhere to the RECOMMENDation as well. 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
