Very handy tool indeed <a href="http://fs_logger.pl">fs_logger.pl</a>.<br><br><div class="gmail_quote">2012/5/5 Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Did you ever resolve this? If not, get a debug console log and put it up on pastebin and then put the pb url here in this thread.<span class="HOEnZb"><font color="#888888"><br>
-MC</font></span><div class="HOEnZb"><div class="h5"><br><br><div class="gmail_quote">On Fri, Apr 27, 2012 at 4:44 AM, rajprithiv89 <span dir="ltr"><<a href="mailto:rajkumar.kanniappan@sasken.com" target="_blank">rajkumar.kanniappan@sasken.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br>
Hi,<br>
<br>
we are using csipsimple android voip application which uses Freeswitch as<br>
SIP/Media server.<br>
<br>
We established voip calls between two android clients(A, B) by enabling SRTP<br>
in both the sides.<br>
<br>
Now the problem observed here is, when we captured the packet flow in<br>
wireshark, the client which initiates the call(A) is sending SRTP packets<br>
and the client which receives the call sends only RTP packets and the<br>
FreeSwitch sends SRTP packets to both the clients.<br>
<br>
We have tested FreeSwitch with other application, there also we ended up in<br>
the same problem(one sided SRTP packets flow).<br>
Also we have tested the same CSipSimple application with Asterisk server. It<br>
works fine by sending SRTP packets from both the clients.<br>
<br>
<br>
When i captured the Wireshark packets SDP contains:<br>
<br>
<br>
>From A to Server:<br>
<br>
m= audio 4006 RTP/SAVP 98 97 99<br>
<br>
<br>
<br>
>From Server to B<br>
<br>
m = audio 31562 RTP/SAVP 3 98 99 9<br>
m = audio 31562 RTP/AVP 3 98 99 9<br>
(sends two media line, one for SRTP and one for RTP)<br>
<br>
<br>
<br>
>From B to Server(Negotiation done)<br>
<br>
m = audio 4014 RTP/SAVP 3 96<br>
m = audio 0 RTP/AVP 3 98 99 9 (This is to inform, B is not going to use<br>
RTP by putting the port number as 0)<br>
<br>
<br>
>From Server to A(Negotiation done)<br>
<br>
m = audio 32338 RTP/SAVP 3 96<br>
<br>
<br>
<br>
But in the end,<br>
A is sending SRTP packets<br>
Server is sending SRTP packets<br>
B is sending only RTP packets (But in the negotiation RTP/AVP port was 0)<br>
<br></blockquote></div><br>
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<br></blockquote></div><br>