I think the time when you were having full audio w/o any RTP proxy was due to the fact that you've everything on the same subnet.<div>In my opinion you'll need RTPproxy eventually whenever you deploy it in real environments.</div>
<div><br></div><div>I cant tell the exact reason for destination unreachable, maybe the virtual IP has something to do with it.</div><div><br></div><div>To configure RTP proxy you need to do following.</div><div>1- Start RTPproxy in bridged mode i.e #<i>/usr/sbin/rtpproxy -l <externalipofproxy>/<internalipofproxy> blah blah switches</i><br>
2- Set module params in opensips.cfg file, and find out the point in main route where call is forwarded to Asterisk Server VIP, just before that write the function "<i>force_rtp_proxy("ei")</i>"<br>3- hmmm..on BYE or CANCEL you need to unforce rtp proxy as well.</div>
<div><br></div><div>Thats all I could think about it so far. </div><div><br></div><div>Regards,</div><div>Sammy</div><div><br></div><div><div class="gmail_quote">On Mon, Dec 19, 2011 at 10:36 PM, Nick Khamis <span dir="ltr"><<a href="mailto:symack@gmail.com">symack@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">What happened to my nice diagram? Argh.... Sorry guys!<br>
<br>
Router -> OpenSIPS -> Asterisk -> ITPS<br>
<div class="HOEnZb"><div class="h5"><br>
On Mon, Dec 19, 2011 at 12:20 PM, Nick Khamis <<a href="mailto:symack@gmail.com">symack@gmail.com</a>> wrote:<br>
> Hello Sammy,<br>
><br>
> Thank you for your response. I now have outgoing audio again which is<br>
> half the battle.<br>
> The second half (incoming audio), has proven to be a challenge. Maybe<br>
> if I start with<br>
> a description of the setup:<br>
><br>
> * This is a test environment done on virtual machines<br>
><br>
><br>
> Network:<br>
><br>
> RouterL (192.168.2.1)<br>
> Polycom Phone (192.168.2.11)<br>
> OpenSIPS (192.168.2.102)<br>
> Asterisk Virtual IP for AST1 and AST2 (192.168.2.6)<br>
> Asterisk1 (192.168.2.110)<br>
> Asterisk2 (192.168.2.111)<br>
><br>
><br>
> ------------- Port FWD (1) --------------------------------<br>
> ---------------------------<br>
> | Router |----------------------> |OpenSIPS/RTPProxy|----------> |<br>
> Asterisk GTWY | ----------- Internet/ITSP<br>
> -------------<br>
> ---------------------------------<br>
> ---------------------------<br>
><br>
><br>
> 1) The port forwarding range is:<br>
> SIP: 5060<br>
> RTP: 10,000-50,000<br>
> RTP Proxy: 7789<br>
><br>
><br>
> I just want to clear some things up. I had outgoing audio the whole<br>
> time without RTPProxy.<br>
> All the test UC (Polycom Phones) are within the same network. Do I<br>
> need to use RTPProxy<br>
> to get incomming audio working? As you can see in the diagram, I did<br>
> try using RTP Proxy<br>
> but never succeeded.<br>
><br>
> Doing a raw UDP trace from ports (10000-50000) I found this:<br>
> <a href="http://pastebin.com/yzgBZQ9S" target="_blank">http://pastebin.com/yzgBZQ9S</a><br>
> There is a "Destination unreachable" at first attempt being returned<br>
> by opensips server,<br>
> and then it dissapears, the it comes back again. Not sure if this is<br>
> related to the no<br>
> outgoing audio, but I will need to resolve it nevertheless.<br>
><br>
> As for a SIP trace without RTP Proxy proxy running:<br>
> <a href="http://pastebin.com/PUXJ3wpK" target="_blank">http://pastebin.com/PUXJ3wpK</a>.<br>
> Wanted to turn your attention to:<br>
><br>
> * The network architecture consists of OpenSIPS sending requests to<br>
> the Asterisk virtual IP (192.168.2.6),<br>
> which is connected to the Asterisk physical machines (192.168.2.110,<br>
> 192.168.2.111). The responding<br>
> asterisk box, in this particular eaxample, was 192.168.2.111. I hope<br>
> this would not be the problem?<br>
><br>
> * A summary of the SDP trace is as follows:<br>
><br>
> INVITE from UC: m=audio 10006 RTP/AVP 0 8 18 101<br>
> OK from Asterisk to OpenSIPS: m=audio 31576 RTP/AVP 8 0 101.<br>
> OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.<br>
> OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101.<br>
> OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.<br>
> OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101.<br>
> OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101.<br>
> OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101.<br>
> OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.<br>
> OK From OpenSIPS to UC: m=audio 34030 RTP/AVP 8 0 101.<br>
> OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.<br>
> OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.<br>
><br>
> Is taht my problem right there? My system is unable to connect<br>
> the initial request from the UC on port 10006, to the followup response<br>
> of the ITSP on port 34030? There is no OK from the UC to OpenSIPS.<br>
><br>
> I've been struggling with this for a week now. Any help would be greatly<br>
> appreciated!<br>
><br>
> Kind Regards,<br>
><br>
> Nick.<br>
<br>
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</div></div></blockquote></div><br></div>