Re: [OpenSIPS-Users] RTP Proxy Problem - No Way Audio (RTP Traces Within)

2013-03-19 Thread Răzvan Crainea

Hi, Nick!

From your traces, I can see that the RTPProxy session is properly 
established (you have both an offer and an answer). But on the media 
level, all I can see is that Asterisk (the callee) is sending RTP to 
caller, but the caller doesn't send anything. Also, this is what 
RTPProxy indicates (RTP stats: 86 in from callee, 0 in from caller). 
Most likely you should checkwhere is the NAT Box trying to sendRTP.


Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 03/15/2013 02:00 AM, Nick Khamis wrote:

Hello Everyone,

I am having problem getting RTP packets flowing smoothly. The setup is

NAT Box (192.168.2.1) <-> OpenSIPS/RTPProxy (192.168.2.5) <-> Asterisk
(192.168.2.10)

I know that media is reaching the boxes since I see:

OpenSIPS (192.168.2.5)

0.00 192.168.2.10 -> 192.168.2.5  UDP 214 Source port: 24454
Destination port: 20198
   0.99  192.168.2.5 -> 81.201.85.45 UDP 214 Source port: 39810
Destination port: 13272
   0.017956 192.168.2.10 -> 192.168.2.5  UDP 214 Source port: 24454
Destination port: 20198
   0.018028  192.168.2.5 -> 81.201.85.45 UDP 214 Source port: 39810
Destination port: 13272
   0.037760 192.168.2.10 -> 192.168.2.5  UDP 214 Source port: 24454
Destination port: 20198
   0.037814  192.168.2.5 -> 81.201.85.45 UDP 214 Source port: 39810
Destination port: 13272


Asterisk CLI (192.168.2.10)

Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)

RTPProxy Messages:

INFO:handle_command: new session
KN74JOEJTRFDVOR3PEH7I5XGBA@81.201.85.45, tag 86219;1 requested, type
strong
INFO:handle_command: new session on a port 20198 created, tag 86219;1
INFO:handle_command: pre-filling caller's address with 81.201.85.45:13272
INFO:handle_command: lookup on ports 20198/39810, session timer restarted
INFO:handle_command: pre-filling callee's address with 192.168.2.10:24454
INFO:handle_delete: forcefully deleting session 1 on ports 20198/39810
INFO:remove_session: RTP stats: 86 in from callee, 0 in from caller,
86 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 20198/39810 is cleaned up


It says 86 in from callee but we do not even have incoming audio. I'm
pretty sure it's "rtpproxy_offer/answer" issue so bellow is my
configuration:

route[1] {
 xlog("Start Call Route For: [ fu=$fu/ tu=$tu /ru=$ru/
ci=$ci]\n");

 if (has_body("application/sdp")) {
 xlog("Has SDP: $fu\n");
 rtpproxy_offer();
 }
}

onreply_route[1] {
 xlog("Reply Route 1: [ fu=$fu/ tu=$tu /ru=$ru/ ci=$ci]\n");
 if (has_body("application/sdp")) {
 xlog("Answering  RTP Proxy: $fu\n");
 rtpproxy_answer();
   }
}

Your help is greatly appreciated,

Nick.

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[OpenSIPS-Users] RTP Proxy Problem - No Way Audio (RTP Traces Within)

2013-03-14 Thread Nick Khamis
Hello Everyone,

I am having problem getting RTP packets flowing smoothly. The setup is

NAT Box (192.168.2.1) <-> OpenSIPS/RTPProxy (192.168.2.5) <-> Asterisk
(192.168.2.10)

I know that media is reaching the boxes since I see:

OpenSIPS (192.168.2.5)

0.00 192.168.2.10 -> 192.168.2.5  UDP 214 Source port: 24454
Destination port: 20198
  0.99  192.168.2.5 -> 81.201.85.45 UDP 214 Source port: 39810
Destination port: 13272
  0.017956 192.168.2.10 -> 192.168.2.5  UDP 214 Source port: 24454
Destination port: 20198
  0.018028  192.168.2.5 -> 81.201.85.45 UDP 214 Source port: 39810
Destination port: 13272
  0.037760 192.168.2.10 -> 192.168.2.5  UDP 214 Source port: 24454
Destination port: 20198
  0.037814  192.168.2.5 -> 81.201.85.45 UDP 214 Source port: 39810
Destination port: 13272


Asterisk CLI (192.168.2.10)

Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)
Sent RTP P2P packet to 192.168.2.5:20198 (type 00, len 000160)

RTPProxy Messages:

INFO:handle_command: new session
KN74JOEJTRFDVOR3PEH7I5XGBA@81.201.85.45, tag 86219;1 requested, type
strong
INFO:handle_command: new session on a port 20198 created, tag 86219;1
INFO:handle_command: pre-filling caller's address with 81.201.85.45:13272
INFO:handle_command: lookup on ports 20198/39810, session timer restarted
INFO:handle_command: pre-filling callee's address with 192.168.2.10:24454
INFO:handle_delete: forcefully deleting session 1 on ports 20198/39810
INFO:remove_session: RTP stats: 86 in from callee, 0 in from caller,
86 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 20198/39810 is cleaned up


It says 86 in from callee but we do not even have incoming audio. I'm
pretty sure it's "rtpproxy_offer/answer" issue so bellow is my
configuration:

route[1] {
xlog("Start Call Route For: [ fu=$fu/ tu=$tu /ru=$ru/
ci=$ci]\n");

if (has_body("application/sdp")) {
xlog("Has SDP: $fu\n");
rtpproxy_offer();
}
}

onreply_route[1] {
xlog("Reply Route 1: [ fu=$fu/ tu=$tu /ru=$ru/ ci=$ci]\n");
if (has_body("application/sdp")) {
xlog("Answering  RTP Proxy: $fu\n");
rtpproxy_answer();
  }
}

Your help is greatly appreciated,

Nick.

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