Hi Sebastian,
I understood, the 'r' flag (iirc being the right one) tells to rtpproxy
to trust the ip in sdp and send the rtp to it as soon as received from
the other side. It is the same case as chaining many rtpproxies.
Cheers,
Daniel
On 7/8/13 2:25 PM, Sebastian Damm wrote:
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
<mico...@gmail.com <mailto:mico...@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
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
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