Hi Daniel,

sorry, I probably didn't explain the problem correctly. The SIP part is okay, the ports on the rtpproxy are allocated, but bridging the audio doesn't work until both parties actually send at least one RTP packet. Since both streams (the inbound and the outbound call) end up at one of the rtpproxies, there will never be an audio stream from the B2BUA to the rtpproxy.

Best Regards,
Sebastian

On Mon, Jul 8, 2013 at 1:57 PM, Daniel-Constantin Mierla <miconda@gmail.com> wrote:
>
> Hello,
>
> iirc the flag, for the requests/replies coming from b2bua, use the 'r' as part of parameters to rtpproxy functions -- check the readme of the module.
>
> Cheers,
> Daniel
>
>
> On 7/8/13 1:52 PM, Sebastian Damm wrote:
>
> Hi,
>
> we are building a setup where we use an rtpproxy in all cases. This works fine except for one scenario.
>
> Caller -> SIP(+rtpproxy) -> B2BUA -> SIP(+rtpproxy) -> Called
>
> In this case, the B2BUA implements forwarding and sends the call back through our setup. The B2BUA does not send out a 183 reponse by itself.
>
> Now, when the caller sends the INVITE, the rtpproxy gets enabled in both cases. The caller sends his RTP to the rtpproxy, after a 183 or 200 OK response, the called sends RTP to the rtpproxy, too, But since the B2BUA doesn't send any audio, both rtpproxies don't know where to pass on the RTP.
>
> Does anybody know how to circumvent this issue? I searched for an option to tell rtpproxy to send the RTP to the address advertised in the SDP as long as it hasn't received any packets on the port, but couldn't find it.
>
> Any hints?
> Thanks in advance,
> Sebastian
>

-- 
Daniel-Constantin Mierla - http://www.asipto.com
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