My friend has this good walkthrough's for Opensips configuration and RTP. and example is <br><br><a href="http://saevolgo.blogspot.com/2012/03/making-rtpproxy-work.html">http://saevolgo.blogspot.com/2012/03/making-rtpproxy-work.html</a><br>
<br>You can also find other posts there. Just go through them and you will be good to go.<br><br>PS: I also learned using rtpproxy using above mentioned page. <br><br>Regards,<br>Qasim<br><br><div class="gmail_quote">On Wed, Aug 29, 2012 at 7:58 AM, Nick Khamis <span dir="ltr"><<a href="mailto:symack@gmail.com" target="_blank">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">Hello Everyone,<br>
<br>
After a week of tinkering with opensips rtpproxy functions, I have a<br>
quite messy config file. Was wondering if anyone would<br>
be kind enough to share or walkthrough a configuration that will get<br>
two way audio working. Presently I have single<br>
outgoing audio. Seems like I am not able to pick up the callee's RTP.<br>
<br>
INFO:remove_session: RTP stats: 0 in from callee, 872 in from caller,<br>
872 relayed, 0 dropped<br>
INFO:remove_session: RTCP stats: 8 in from callee, 2 in from caller,<br>
10 relayed, 0 dropped<br>
<br>
Basic layout of the network<br>
<br>
router 192.168.2.1<br>
opensips 192.168.2.102 (bridged virutal box, ports forwarded)<br>
asterisk 192.168.2.110 (bridged virutal box)<br>
Polycom 192.168.2.11<br>
<br>
[router]-----[opensips]-------[asterisk]--------[SIP Trunk]<br>
<br>
Please bare with the virtual box setup, I am just trying to get all<br>
the configs together before deploying onto the servers. I know i'm<br>
really<br>
close, and would love to be able to move on to the other parts (i.e.,<br>
dialplan, routing etc...)<br>
<br>
I pasted an ngrep trace at <a href="http://pastebin.com/A39vBG3t" target="_blank">http://pastebin.com/A39vBG3t</a>.<br>
<br>
Thank you Kindly,<br>
<br>
Nick.<br>
<br>
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</blockquote></div><br>