[OpenSIPS-Users] Asterisk reinvite confuses Mediaproxy

[Jeff Pyle]

Let's try it with the attachment this time...


On 7/17/09 4:48 PM, "Jeff Pyle" <jpyle at fidelityvoice.com> wrote:

> 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

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