On 4/14/06, Paul Kyzivat <[EMAIL PROTECTED]> wrote:

> [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.]

Gmail ;) OK, once again:

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          |                             |


> 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"?

b2bua


> 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.

I know but the original callee replied in '200 OK' on the original
INVITE with only best codec not with all supported codecs. My SIP
server can not keep my own MOH in all possible codecs. Actually the
only solution I have at the moment is to send MOH in one of G711
codecs in the case (if we belive that all user agents support G711).

Any options to learn the supported codec list?

--
tut

_______________________________________________
Sip-implementors mailing list
[email protected]
https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors

Reply via email to