[OpenSIPS-Users] Asterisk reinvite confuses Mediaproxy

[Ruud Klaver]

Hi Jeff,

On 16 Jul 2009, at 00:37, Jeff Pyle wrote:

> We're interfacing this Opensips + Mediaproxy with an existing  
> system.  We
> cannot change the existing system.  Eventually this system will  
> contain
> NAT'ed clients, hence the Mediaproxy.  I didn't want to introduce  
> NAT until
> everything was working properly without it.
>
> I'll try the use_media_proxy() method and see what happens.
>
>
> - Jeff
>
>
> On 7/12/09 5:28 AM, "Olle E. Johansson" <oej at edvina.net> wrote:
>
>> On a different note, why keep asterisk re-invites turned on if you  
>> use
>> a media proxy in the call?
>> Re-invites are best used when no NAT or firewall support is needed,
>> and since you're using media proxy, it indicates to me that you might
>> not want to have re-invites turned on at all.
>>
>> /O

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
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