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Hello,<br>
<br>
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.<br>
<br>
Cheers,<br>
Daniel<br>
<br>
<div class="moz-cite-prefix">On 7/8/13 1:52 PM, Sebastian Damm
wrote:<br>
</div>
<blockquote
cite="mid:CABkWSFzZJgU7YhS4Lz_OFXGXV-wQxsphK5oXLnVEH_F3zuSrFg@mail.gmail.com"
type="cite">
<div dir="ltr">
<div>
<div>Hi,<br>
<br>
</div>
we are building a setup where we use an rtpproxy in all cases.
This works fine except for one scenario.<br>
<br>
</div>
Caller -> SIP(+rtpproxy) -> B2BUA -> SIP(+rtpproxy)
-> Called<br clear="all">
<div>
<div>
<div><br>
</div>
<div>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.<br>
<br>
</div>
<div>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. <br>
<br>
</div>
<div>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.<br>
<br>
Any hints?<br>
</div>
<div>Thanks in advance,<br>
</div>
<div>Sebastian<br>
</div>
</div>
</div>
</div>
<br>
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</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla - <a class="moz-txt-link-freetext" href="http://www.asipto.com">http://www.asipto.com</a>
<a class="moz-txt-link-freetext" href="http://twitter.com/#!/miconda">http://twitter.com/#!/miconda</a> - <a class="moz-txt-link-freetext" href="http://www.linkedin.com/in/miconda">http://www.linkedin.com/in/miconda</a>
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