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

[Razvan Crainea]

Hi, Nick!

In the traces you sent, it seems that the callee's RTP traffic doesn't 
reach the media relay server. This is most likely because the RTPProxy 
is behind NAT. In this case you have to double check two things:
* if the SDP is properly changed for the INVITEs and 200OKs
* if the media ports are properly forwarded to RTPProxy.

The second thing I noticed is that you use both engage_rtp_proxy and 
rtpproxy_* functions. I don't think this is such a good idea to mix 
them. Anyway, in your script, engage_rtp_proxy doesn't have any effect 
because you are not supposed to call it on sequential requests, only on 
initial request. So you can simply delete it.

Regards,

--
Răzvan Crainea
OpenSIPS Developer


On 12/23/2011 05:25 AM, Nick Khamis wrote:
> 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
>
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