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 href="http://192.168.3.18" target="_blank">192.168.3.18</a> -s udp:<a 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>
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