Hi,<div><br><div>What is of more interest to me is this:</div><div><span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13.333333969116211px;background-color:rgb(255,255,255)"><br></span></div><div><span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13.333333969116211px;background-color:rgb(255,255,255)">INFO:remove_session: RTP stats: </span><span style="font-family:arial,sans-serif;font-size:13.333333969116211px;background-color:rgb(255,255,255)"><font color="#ff0000">0 in from callee</font></span><span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13.333333969116211px;background-color:rgb(255,255,255)">, 872 in from caller,</span><span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13.333333969116211px;background-color:rgb(255,255,255)">872 relayed, 0 dropped</span><br style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13.333333969116211px;background-color:rgb(255,255,255)">
<span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13.333333969116211px;background-color:rgb(255,255,255)">INFO:remove_session: RTCP stats: </span><span style="font-family:arial,sans-serif;font-size:13.333333969116211px;background-color:rgb(255,255,255)"><font color="#ff0000">8 in from callee</font></span><span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13.333333969116211px;background-color:rgb(255,255,255)">, 2 in from caller,</span><span style="color:rgb(34,34,34);font-family:arial,sans-serif;font-size:13.333333969116211px;background-color:rgb(255,255,255)">10 relayed, 0 dropped</span> <br>
<br>While there are no RTPs from the Asterisk/or SIP trunk, RTCPs are still detected. </div><div><br></div><div>Just before you look deeply in the RTPproxy module make sure that the same setup, with phone directly connected to asterisk makes an outbound call via the SIP trunk and there is two-way audio. </div>
<div><br></div><div>Are there NAT issues with SIP trunk and Asterisk !! If they are resolved (in my opinion easily) only then move to the difficult part, the RTPproxy.</div><div><br></div><div>BR</div><div>Sammy</div><div>
<br></div><div><br></div><div><br><div class="gmail_quote">On Wed, Aug 29, 2012 at 11:01 AM, <a href="mailto:qasimakhan@gmail.com">qasimakhan@gmail.com</a> <span dir="ltr"><<a href="mailto:qasimakhan@gmail.com" target="_blank">qasimakhan@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">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" target="_blank">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<div class="HOEnZb"><div class="h5">
<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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