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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