[OpenSIPS-Users] Simple User Scenario to get two way audio working (Please Help)

[Nick Khamis]

Hello Everyone,

I am testing a simple case where we have:

UC--->OpenSIPS---->ITSP Gateway

The UC and OpenSIPS are on the same network behind a router. The
related parts of my config look like:

route{

        .....
	
	if (is_method("INVITE") && has_totag()) engage_rtp_proxy("ie","127.0.0.1");
	

        .....
}

route[1] {	
	if (is_method("INVITE")) {
		xlog("Start Call for route [ fu=$fu/ tu=$tu /ru=$ru/ ci=$ci]");
		prefix("00111");
		rewritehostport("itsp.serviceprovider.com:5060");
		if (has_body("application/sdp")) {
			if (rtpproxy_offer()) t_on_reply("1");
			else t_on_reply("2");
		}

		t_on_branch("2");
		t_on_failure("1");
	}

	if (is_method("ACK") && has_body("application/sdp")) rtpproxy_answer();


	if (!t_relay()) {
		sl_reply_error();
	}		

	exit;
}

onreply_route[1] {
	xlog("incoming reply\n");
	
	if (has_body("application/sdp")) rtpproxy_answer();
	exit;
}

onreply_route[2] {
	xlog("incoming reply\n");
	
	if (has_body("application/sdp")) rtpproxy_offer();
	exit;
}

Everything seems to be starting fine except there is no audio both
ways. RTPProxy is started using the following:

./rtpproxy -s udp:127.0.0.1:7789 -l 192.168.2.102 -m 10000 -M 20000 -u
root root -F -f -d INFO LOG_LOCAL0
INFO:main: rtpproxy started, pid 2185
INFO:handle_command: new session
7e6a2939-599870c3-1f489940 at 192.168.2.11, tag D6D9E7BD-BC7B6732;1
requested, type strong
INFO:handle_command: new session on a port 11420 created, tag
D6D9E7BD-BC7B6732;1
INFO:handle_command: pre-filling caller's address with 192.168.2.11:2222
INFO:handle_command: lookup on ports 11420/15872, session timer restarted
INFO:handle_command: pre-filling callee's address with 95.211.119.251:36076
INFO:process_rtp: session timeout
INFO:remove_session: RTP stats: 0 in from callee, 326 in from caller,
326 relayed, 0 dropped
INFO:remove_session: RTCP stats: 0 in from callee, 0 in from caller, 0
relayed, 0 dropped
INFO:remove_session: session on ports 11420/15872 is cleaned up

A short sip trace:

 4.714356 192.168.2.11 -> 192.168.2.102 SIP/SDP Request: INVITE
sip:15078392302 at opensips1.test.com;user=phone, with session
description
  4.790394 192.168.2.102 -> 192.168.2.11 SIP Status: 100 Giving a try
  4.802400 192.168.2.102 -> 202.163.224.113 DNS Standard query A
itsp.serviceprovider.com
  4.982490 202.163.224.113 -> 192.168.2.102 DNS Standard query
response A 93.221.112.231
  4.982490 192.168.2.102 -> 93.221.112.231 SIP/SDP Request: INVITE
sip:001110315148392007 at sbc.voxbeam.com:5060;user=phone, with session
description
  5.070534 93.221.112.231 -> 192.168.2.102 SIP Status: 100 Trying
incoming reply
  6.339168 93.221.112.231 -> 192.168.2.102 SIP/SDP Status: 183 Session
Progress, with session description
  6.339168 192.168.2.102 -> 192.168.2.11 SIP/SDP Status: 183 Session
Progress, with session description
  6.467232 192.168.2.11 -> 192.168.2.102 RTP PT=ITU-T G.711 PCMU,
SSRC=0x7E7F31B9, Seq=38666, Time=2196142936, Mark

As mentioned earlier the UC and OpenSIPS+RTPProxy are behind a router
with ports 5060, and 10000-20000 being forwarded to the OpenSIPS
server.
I even tried putting the server in the DMZ, but still to no avail.


Thanks in Advance, and Happy Holidays,

Nick