[OpenSIPS-Users] nat problems with one-way audio and forwarding

[Bogdan-Andrei Iancu]

Hi Kelly,

There are 2 approaches:

1) if you enabled rtpproxy in request route for the INVITE, then, 
whatever branches you keep forking, take care and force rtpproxy in all 
200 OK (whatever branch - nated or not).

2) enable rtproxy individually, per branch - instead of using the 
request route for forcing RTPP in the INVITE, do use branch_route[] as 
changes in branch route do apply only for that branch and not to all 
branches (so for the second not nated branch, you can simply avoid the 
usage of RTPP).

Regards,
Bogdan

prescott at wcoil.com wrote:
> I need some advice:
> I have a test case that looks like this:
> outside customer calls a phone number, number is busy.
> opensips looks up the customer preference and forwards the busy call 
> to another phone.
> the first (busy) number is behind  a nat.
> the second is not.
> I am using rtpproxy for my media relaying on the nat side.
> The problem is that when the 200 OK response ggets sent from the 
> second phone picking up the call, opensips does not fix the sdp in the 
> message.
> This results in one-way audio on the call.
> I am using opensips-1.5.3
> I am attaching the sip trace for reference.
> The way I do call forwarding is: look for the 486 busy response and 
> then append_branch to the forwarded destination.
> as you can see in the invite, the sdp information is retained, but the 
> system doesn't seem to recognise the ok response as part of that sip 
> transaction.
> Any help or suggestions of where to look would be appreciated.
> Thanks.
>
> -- Kelly Prescott
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