[OpenSIPS-Users] Asterisk reinvite confuses Mediaproxy

[Jeff Pyle]

Hi Ruud,

Sorry for any confusion.  I've attached fresh traces, including a full ngrep
and mediaproxy relay and dispatcher logs.

This is an inbound call from PSTN gateway to Asterisk (with reinvites) to
Opensips with Mediaproxy to the callee endpoint.  I have a single
engage_media_proxy() at the initial invite.


- Jeff




On 7/16/09 4:15 AM, "Ruud Klaver" <ruud at ag-projects.com> wrote:

> Hi Jeff,
>
> I've just been scrutinizing your SIP trace, as you still haven't
> provided me with mediaproxy-relay debug output. What happens when the
> SDP offerer comes with a new ip/port combination for a particular
> stream is that mediaproxy allocates a new set of ports for this
> internally. You can see that this happens by the fact that for the re-
> invite, the RTP port in the modified SDP is different. This means that
> both endpoints actually should start sending to a new destination as a
> result of the re-INVITE exchange. If they do, the previous RTP
> exchange and the next one can never actually "cross wires".
> 
> Now I'm not exactly sure what your problem is, as you said before it's
> PSTN -> SIP phone that is giving you trouble, yet you've included a
> trace which seems to be in the opposite direction. Again, please
> include a media-relay log and describe what you are (not) hearing at
> either endpoint.
> 
> Ruud Klaver
> AG Projects