If you are a B2BUA, couldnt we be able to send a RE-INVITE with out SDP
towards the callee? The callee should provide all capabilities at that time.

Venkatesh


On 4/14/06, Paul Kyzivat <[EMAIL PROTECTED]> wrote:
>
> Artem,
>
> [your call flow was mangled almost to the point of complete
> incomprehensibilty by the time it reached my mail reader (Thunderbird).
> I *think* I figured it out.]
>
> You need to say more about the players here. Is the SIP server an agent
> for the callee? Is it a B2BUA? Or a "proxy"?
>
> If it is a "proxy" then you are out of luck.
>
> If it is a B2BUA, you have options, though they may start to get
> complicated. It can answer the invite from the caller itself. Then it
> can reinvite the callee, offering at least one of the codecs that had
> been previously agreed and offering one that it does support as well,
> listing the supported one first as preferred. If the one it supports is
> accepted all is well. If not, it won't be able to do MOH, and may want
> to reinvite again specifying a=inactive, or c=0.
>
>        Paul
>
> Artem Naluzhny wrote:
> > hi
> > Here is the call scenario:
> > Caller side                   SIP server                    Callee
> side|                             |                             |@-> INVITE
> ------------------>|                             ||   m=audio 16428 RTP/AVP
> 4 0 2 8 100 101                   ||
> |                             ||                             @-> INVITE
> ------------------>||                             |   m=audio 16428 RTP/AVP
> 4 0 2 8 100 101|                             |
> ||                             |<- 200 OK
> -------------------@|                             |   m=audio 16460 RTP/AVP
> 4 100 101|                             |                             ||<-
> 200 OK -------------------@                             ||   m=audio 16460
> RTP/AVP 4 100 101                         ||
> |                             ||
> |                             |@-> re-INVITE --------------->|
>            ||   c=IN IP4 0.0.0.0          |                             |
> > On-hold event has been detected and now the SIP server wants to playmy
> own music-on-hold to callee side. But the problem is that SIPserver has no
> the MOH in G723 codec but has it in G711 and G729 forexample. I know that
> the callee side supports both G711 and G729codecs. The question is how to
> ask callee side of established dialogfor full list of supported codecs?
> > (Actually the same codec related issue exists for attended
> transferhandling on SIP server side.)
> > --tut
> > _______________________________________________
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> >
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