[SR-Users] RTPproxy issue in forwarding scenario

[Sebastian Damm]

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 at 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
>
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