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. -MC On Fri, Apr 27, 2012 at 4:44 AM, rajprithiv89 < rajkumar.kanniappan@sasken.com> wrote:
Hi,
we are using csipsimple android voip application which uses Freeswitch as SIP/Media server.
We established voip calls between two android clients(A, B) by enabling SRTP in both the sides.
Now the problem observed here is, when we captured the packet flow in wireshark, the client which initiates the call(A) is sending SRTP packets and the client which receives the call sends only RTP packets and the FreeSwitch sends SRTP packets to both the clients.
We have tested FreeSwitch with other application, there also we ended up in the same problem(one sided SRTP packets flow). Also we have tested the same CSipSimple application with Asterisk server. It works fine by sending SRTP packets from both the clients.
When i captured the Wireshark packets SDP contains:
From A to Server:
m= audio 4006 RTP/SAVP 98 97 99
From Server to B
m = audio 31562 RTP/SAVP 3 98 99 9 m = audio 31562 RTP/AVP 3 98 99 9 (sends two media line, one for SRTP and one for RTP)
From B to Server(Negotiation done)
m = audio 4014 RTP/SAVP 3 96 m = audio 0 RTP/AVP 3 98 99 9 (This is to inform, B is not going to use RTP by putting the port number as 0)
From Server to A(Negotiation done)
m = audio 32338 RTP/SAVP 3 96
But in the end, A is sending SRTP packets Server is sending SRTP packets B is sending only RTP packets (But in the negotiation RTP/AVP port was 0)