Razvan, <div><br></div><div>thanks, i was actually able to make it work, it was all a matter of putting the right flags on the engage_rtp_proxy, since the ftp proxy command was fine. </div><div><br></div><div>engage_rtp_proxy("rcie");</div>
<div><br></div><div>Now i have a different problem. Iooking around i came across a long email exchange you had with someone else regarding the set_rtp_proxy_set() function. </div><div>For some reason its able to create on side of the socket, but when creating the other side to bridge it too, it comes back saying that a set could not be found. </div>
<div><br></div><div><div>This is the config for those sets. </div><div><div>modparam("rtpproxy", "rtpproxy_sock", "1==udp:localhost:7001")</div><div>modparam("rtpproxy", "rtpproxy_sock", "2==udp:localhost:7002")</div>
<div>modparam("rtpproxy", "rtpproxy_sock", "3==udp:localhost:7003")</div><div>modparam("rtpproxy", "rtpproxy_sock", "4==udp:localhost:7004")</div></div></div><div>
<br></div><div>Whats interesting is that on load opensips complaints that </div><div>/sbin/opensips[28262]: ERROR:rtpproxy:select_rtpp_set: script error-invalid id_set to be selected </div><div><br></div><div>However it does connect to each socket and enables it. </div>
<div>INFO:rtpproxy:rtpp_test: rtp proxy <udp:localhost:7001> found, support for it enabled</div><div>INFO:rtpproxy:rtpp_test: rtp proxy <udp:localhost:7002> found, support for it enabled</div><div>INFO:rtpproxy:rtpp_test: rtp proxy <udp:localhost:7003> found, support for it enabled</div>
<div>INFO:rtpproxy:rtpp_test: rtp proxy <udp:localhost:7004> found, support for it enabled</div><div><br></div><div>The open sips script is a simple .</div><div><div>set_rtp_proxy_set("1");</div><div>engage_rtp_proxy("rcie");</div>
<div><br></div><div>But when placing a call i get </div><div><span class="Apple-tab-span" style="white-space:pre"> </span></div></div><div><div>Jul 26 08:32:49 gw1 /sbin/opensips[28287]: -> Route(0) - New Incomming Invite request to [sip:xxxxx@PUBLICIP] </div>
<div>Jul 26 08:32:49 gw1 /sbin/opensips[28287]: -> Route(0) - Group selected 1 </div><div><span class="Apple-style-span">Jul 26 08:32:49 gw1 /sbin/opensips[28287]: -> Route(0) - </span>xxxxx<span class="Apple-style-span"> going to call to sip:<a href="http://192.168.3.8:5060">192.168.3.8:5060</a> </span></div>
<div>Jul 26 08:32:49 gw1 /sbin/opensips[28287]: -> Route(0) - RT Proxy Set 1</div><div><span class="Apple-style-span">Jul 26 08:32:49 gw1 rtpproxy[26135]: INFO:handle_command: new session </span>xxxxx<span class="Apple-style-span">@<a href="http://domain.ip-com.com">domain.ip-com.com</a>, tag </span>xxxxx<span class="Apple-style-span">;1 requested, type strong</span></div>
<div>Jul 26 08:32:49 gw1 rtpproxy[26135]: INFO:handle_command: new session on a port 14690 created, tag 3552298369-227332;1</div><div><span class="Apple-style-span">Jul 26 08:32:49 gw1 rtpproxy[26135]: INFO:handle_command: pre-filling caller's address with </span>xxxxx<span class="Apple-style-span">:52810</span></div>
<div>Jul 26 08:32:49 gw1 /sbin/opensips[28287]: -> Route(1) - Forwarding Call to [<a href="http://sip:xxxxxxx@192.168.3.8:5060">sip:xxxxxxx@192.168.3.8:5060</a>] </div><div><b>Jul 26 08:32:56 gw1 /sbin/opensips[28338]: ERROR:rtpproxy:select_rtpp_node: script error -no valid set selected </b></div>
<div><b>Jul 26 08:32:56 gw1 /sbin/opensips[28338]: ERROR:rtpproxy:force_rtp_proxy: no available proxies </b></div><div>Jul 26 08:32:59 gw1 /sbin/opensips[28336]: ERROR:rtpproxy:select_rtpp_node: script error -no valid set selected </div>
<div>Jul 26 08:32:59 gw1 /sbin/opensips[28336]: ERROR:rtpproxy:force_rtp_proxy: no available proxies </div><div>Jul 26 08:33:08 gw1 /sbin/opensips[28287]: ERROR:rtpproxy:select_rtpp_node: script error -no valid set selected </div>
<div>Jul 26 08:33:08 gw1 /sbin/opensips[28287]: ERROR:rtpproxy:unforce_rtp_proxy_f: no available proxies </div><div>Jul 26 08:33:08 gw1 /sbin/opensips[28287]: ERROR:rtpproxy:engage_close_callback: cannot unforce rtp proxy </div>
<div><br></div></div><div><br></div><div>Has anyone found a solution to this? (same thing happens with using rtpproxy_offer)</div><div><br></div><div>Thanks</div><div><br><br><div class="gmail_quote">On Thu, Jul 26, 2012 at 3:42 AM, Răzvan Crainea <span dir="ltr"><<a href="mailto:razvan@opensips.org" target="_blank">razvan@opensips.org</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div><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 cols="72">Razvan Crainea
OpenSIPS Core Developer
<a href="http://www.opensips-solutions.com" target="_blank">http://www.opensips-solutions.com</a>
</pre><div><div class="h5">
On 07/25/2012 08:26 PM, Sebastian Sastre wrote:<br>
</div></div></div>
<blockquote type="cite"><div><div class="h5">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>
</blockquote>
</div>
<br>
<br>
<fieldset></fieldset>
<br>
</div></div><pre>_______________________________________________
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</pre>
</blockquote>
<br>
</div>
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<br></blockquote></div><br></div>