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<div class="moz-cite-prefix"><tt>Hi, Sebastian!<br>
<br>
The problem is that engage_rtp_proxy function does not support
media bridging functionality. In order to implement the scenario
described successfully, you will have to configure rtpproxy
manually, using rtpproxy_offer/rtpproxy_answer.<br>
<br>
Best regards,<br>
</tt>
<pre class="moz-signature" cols="72">Razvan Crainea
OpenSIPS Core Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com">http://www.opensips-solutions.com</a>
</pre>
On 07/25/2012 08:26 PM, Sebastian Sastre wrote:<br>
</div>
<blockquote
cite="mid:CAB=UX=CDMNGBNBv4O3tzWL5RqZ+7398ytkf4ifZm69TcFrwD_A@mail.gmail.com"
type="cite">Hello, <br>
<div class="gmail_quote">
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><br>
</div>
<div>Im trying to configure RTP proxy for the following
scenario. </div>
<div><br>
</div>
<div>I have opensips in the border server between my public
ipaddress and my internal. So that ETH0 has public IP and
eth1 has private. </div>
<div><br>
</div>
<div>I set mhomed=1 so that packets are forwarded correctly
and the call does connect. </div>
<div><br>
</div>
<div>First of all, is this possible with rtpproxy and
opensips? I know Mediaproxy is not able to bind like that. </div>
<div><br>
</div>
<div>The RTP proxy command I'm using is </div>
<div><br>
</div>
<div>rtpproxy -f -l PUBLIC-IP/<a moz-do-not-send="true"
href="http://192.168.3.18" target="_blank">192.168.3.18</a>
-s udp:<a moz-do-not-send="true"
href="http://127.0.0.1:12221" target="_blank">127.0.0.1:12221</a>
-F</div>
<div><br>
</div>
<div>Opensips Config looks like this. </div>
<div><br>
</div>
<div>
<div>
<div> if (is_method("INVITE")){</div>
<div> ##xlog("-> Route(0) - New
Incomming Invite request to [$ru]\n");</div>
<div> if
(check_source_address("1")) {</div>
<div> record_route();</div>
<div> create_dialog();</div>
<div><br>
</div>
<div> if
(load_balance("1","channels")){</div>
<div><br>
</div>
<div> # dst URI
points to the new destination</div>
<div> #ahora le
sacamos el prefijo</div>
<div>
dp_translate("1","$ruri.user/$ruri.user");</div>
<div><br>
</div>
<div> #xlog("->
Route(0) - $ruri.user going to call to $du\n");</div>
<div> $ru = "sip:"
+ $rU + "@" + $dd + ":" + $dp;</div>
<div> $avp(dst) =
$dd;</div>
<div><br>
</div>
<div>
engage_rtp_proxy("ie");</div>
<div> route(1);</div>
<div> }</div>
<div> else{</div>
<div> <span
style="white-space:pre-wrap"> </span>xlog("->
Route(0) - Did not find available GWs\n");</div>
<div>
sl_send_reply("500", "All is full");</div>
<div> }</div>
<div> }</div>
<div> else{</div>
<div> <span
style="white-space:pre-wrap"> </span>xlog("-> IP
not in address table \n");</div>
<div>
sl_send_reply("503","IP not in address table");</div>
<div> }</div>
<div> }</div>
</div>
</div>
<div><br>
</div>
<div>When is tart open sips, it connects to RTP Proxy with no
problems. </div>
<div><br>
</div>
<div>When the invite is sent out to the internal ip it
correctly sets c= on the body but i don't see rtpproxy doing
anything nor i get audio in any direction. </div>
<div><br>
</div>
<div>rtpproxy leaves no logs, or errors or anything. </div>
<div><br>
</div>
<div>am i doing something wrong here? </div>
<div><br>
</div>
<div>thanks </div>
<div><br>
</div>
<div><br>
</div>
<div><br>
</div>
</blockquote>
</div>
<br>
<br>
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<br>
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</pre>
</blockquote>
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