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(a)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