Hi ,

My observation is that in some implementation, the
negotiated codec is the first one listed in the 200 OK
message SDP.

Other than that, the RTP layer should be ready to
handle all codecs it announced as supported. In this
case, it would be the set of codecs common in both the
messages (INVITE & 200 OK). This allows the RTP layer
switch between the codecs types, at any point of time
during a session.

Regards,
Prithi.

--- Alex Zeffertt <[EMAIL PROTECTED]> wrote:
> 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
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>
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