Hey,
Unfortunately your packet dumps are truncated and don't show the complete SDP bodies. It would also be interesting to see which options and parameters are passed to mediaproxy-ng when processing the SDP. You would find this info in the log produced by mediaproxy-ng, which should also include the full SDP bodies going in and out (unless your syslog daemon also truncates those messages). So, the most useful way to debug this is to post the complete log lines.
cheers
On 04/01/14 13:19, Olli Heiskanen wrote:
Hello,
I've been experimenting with Kamailio with ws and sip clients and could need a hand in getting a call between those two to work.
I have Kamailio 4.1.2 (using rtpproxy-ng instead of rtpproxy) on a CentOS 6.5 and a mediaproxy-ng running. I have clients wsclient@testers.com mailto:wsclient@testers.com and gsclient@testers.com mailto:gsclient@testers.com and I try to make call from wsclient to gsclient. The wsclient is a jssip client running on chrome and gsclient is a grandstream desk phone. My config file is the default one enhanced by online examples.
I use a html5 <audio> element for the media streams, and configured my jssip phone to accept audio options like this: var options = { 'eventHandlers': eventHandlers, 'mediaConstraints': {'audio': true, 'video': false } }; sipUA.call(callto, options);
I used the instructions from here: http://www.slideshare.net/crocodilertc/webrtc-websockets
What I get is gsclient ringing, and as I answer there is no audio and call hangs up in a few seconds. I guess this is a SDP problem, something between Kamailio and Mediaproxy-ng but SDP is not my strong point so I'd appreciate advice.
Question is where's my misconfiguration/problem? I would like to learn why this problem occurs and how to fix it rather than getting a solution right away, but please bear in mind I don't know much about SDP.
In Kamailio log I see: kamailio[27059]: ERROR: rtpproxy-ng [rtpproxy.c:1346]: rtpp_function_call(): proxy replied with error: Error rewriting SDP kamailio[27058]: ERROR: rtpproxy-ng [rtpproxy.c:1346]: rtpp_function_call(): proxy replied with error: Unknown call-id kamailio[27057]: ERROR: rtpproxy-ng [rtpproxy.c:1346]: rtpp_function_call(): proxy replied with error: Unknown call-id
Following are the INVITEs and 200 OKs from my SIP trace (1.1.1.1 is the ip of my Kamailio & mediaproxy-ng box and 2.2.2.2 is the public ip behind which both my clients are). The gsclient has port 5066.
U 2014/04/01 20:03:41.060009 1.1.1.1:5060 http://1.1.1.1:5060 -> 2.2.2.2:5066 http://2.2.2.2:5066 INVITE sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0. Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes. Record-Route: sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Max-Forwards: 16. To: <sip:gsclient@testers.com mailto:sip%3Agsclient@testers.com>. From: <sip:wsclient@testers.com mailto:sip%3Awsclient@testers.com>;tag=hhcd99tmvm. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. Contact: <sip:wsclient@testers.com mailto:sip%3Awsclient@testers.com;gr=urn:uuid:f6014564-88cb-4f57-9ae5-3b4336ef9db8;ob;alias=2.2.2.2~38986~5;alias=2.2.2.2~38986~5>. Allow: ACK,CANCEL,BYE,OPTIONS,INVITE. Content-Type: application/sdp. Supported: path, outbound, gruu. User-Agent: JsSIP 0.3.0. Content-Length: 2211. . v=0. o=- 4897716268503406223 2 IN IP4 1.1.1.1. s=-. t=0 0. a=group:BUNDLE audio. a=msid-semantic: WMS vMh5vhUEQzvVKJYdqRkAuCcXVa2blgbEXARZ. m=audio 30028 RTP/SAVPF 111 103 104 0 8 106 105 13 126. c=IN IP4 1.1.1.1. a=candidate:2999745851 1 udp 2113937151 192.168.56.1 63341 typ host generation 0. a=candidate:2999745851 2 udp 2113937151 192.168.56.1 63341 typ host generation 0. a=candidate:3350409123 1 udp 2113937151 192.168.0.101 63342 typ host generation 0. a=candidate:3350409123 2 udp 2113937151 192.168.0.101 63342 typ host generation 0. a=candidate:4233069003 1 tcp 1509957375 192.168.56.1 0 typ host generation 0. a=candidate:4233069003 2 tcp 150995
T 2014/04/01 20:03:41.119806 2.2.2.2:38986 http://2.2.2.2:38986 -> 1.1.1.1:5060 http://1.1.1.1:5060 [A] ......
U 2014/04/01 20:03:41.159086 2.2.2.2:5066 http://2.2.2.2:5066 -> 1.1.1.1:5060 http://1.1.1.1:5060 SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes. Record-Route: sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes. From: <sip:wsclient@testers.com mailto:sip%3Awsclient@testers.com>;tag=hhcd99tmvm. To: <sip:gsclient@testers.com mailto:sip%3Agsclient@testers.com>;tag=7875f08763872c34. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. User-Agent: Grandstream GXP2000 1.2.2.26. Warning: 304 GS "Media type not available". Content-Length: 0. .
U 2014/04/01 20:03:41.159392 1.1.1.1:5060 http://1.1.1.1:5060 -> 2.2.2.2:5066 http://2.2.2.2:5066 ACK sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.0. Max-Forwards: 16. To: <sip:gsclient@testers.com mailto:sip%3Agsclient@testers.com>;tag=7875f08763872c34. From: <sip:wsclient@testers.com mailto:sip%3Awsclient@testers.com>;tag=hhcd99tmvm. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 ACK. Content-Length: 0. .
U 2014/04/01 20:03:41.161085 1.1.1.1:5060 http://1.1.1.1:5060 -> 2.2.2.2:5066 http://2.2.2.2:5066 INVITE sip:gsclient@192.168.0.106:5066;transport=udp SIP/2.0. Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes. Record-Route: sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.1. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Max-Forwards: 16. To: <sip:gsclient@testers.com mailto:sip%3Agsclient@testers.com>. From: <sip:wsclient@testers.com mailto:sip%3Awsclient@testers.com>;tag=hhcd99tmvm. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. Contact: <sip:wsclient@testers.com mailto:sip%3Awsclient@testers.com;gr=urn:uuid:f6014564-88cb-4f57-9ae5-3b4336ef9db8;ob;alias=2.2.2.2~38986~5;alias=2.2.2.2~38986~5>. Allow: ACK,CANCEL,BYE,OPTIONS,INVITE. Content-Type: application/sdp. Supported: path, outbound, gruu. User-Agent: JsSIP 0.3.0. Content-Length: 3136. . v=0. o=- 4897716268503406223 2 IN IP4 1.1.1.1. s=-. t=0 0. a=group:BUNDLE audio. a=msid-semantic: WMS vMh5vhUEQzvVKJYdqRkAuCcXVa2blgbEXARZ. m=audio 30028 RTP/AVP 111 103 104 0 8 106 105 13 126. c=IN IP4 1.1.1.1. a=fingerprint:sha-256 72:54:87:EC:D2:4C:D1:70:C2:FE:69:08:20:5C:92:1D:E0:EA:BD:45:09:E0:90:62:27:B6:34:60:54:E2:99:28. a=setup:actpass. a=mid:audio. a=sendrecv. a=rtpmap:111 opus/48000/2. a=fmtp:111 minptime=10. a=rtpmap:103 ISAC/16000. a=rtpmap:104 ISAC/32000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:106 CN/32000. a=rtpmap:105 CN/16000. a=rtpmap:13 CN/8000. a=rtpmap:126 telephone-event/8000. a=maxptime:60. a=ssrc:3298511848 cnam
And here are the 200 OK messages when answering the call:
U 2014/04/01 20:03:46.049711 2.2.2.2:5066 http://2.2.2.2:5066 -> 1.1.1.1:5060 http://1.1.1.1:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKb703.fbb259c1d8c17e163876ec760e086145.1. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes. Record-Route: sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes. From: <sip:wsclient@testers.com mailto:sip%3Awsclient@testers.com>;tag=hhcd99tmvm. To: <sip:gsclient@testers.com mailto:sip%3Agsclient@testers.com>;tag=fb215901a251c9a0. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. User-Agent: Grandstream GXP2000 1.2.2.26. Contact: sip:gsclient@192.168.0.106:5066;transport=udp. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE. Content-Type: application/sdp. Supported: replaces, timer. Content-Length: 216. . v=0. o=gsclient 8000 8000 IN IP4 192.168.0.106. s=SIP Call. c=IN IP4 192.168.0.106. t=0 0. m=audio 5026 RTP/AVP 0 13. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. m=audio 0 RTP/SAVPF 111 103 104 0 8 106 105 13 126.
T 2014/04/01 20:03:46.051127 1.1.1.1:5060 http://1.1.1.1:5060 -> 2.2.2.2:38986 http://2.2.2.2:38986 [AP] .~.dSIP/2.0 200 OK. Via: SIP/2.0/WS kj59uak271em.invalid;rport=38986;received=2.2.2.2;branch=z9hG4bK9891267. Record-Route: sip:1.1.1.1;r2=on;lr=on;nat=yes. Record-Route: sip:1.1.1.1;transport=ws;r2=on;lr=on;nat=yes. From: <sip:wsclient@testers.com mailto:sip%3Awsclient@testers.com>;tag=hhcd99tmvm. To: <sip:gsclient@testers.com mailto:sip%3Agsclient@testers.com>;tag=fb215901a251c9a0. Call-ID: 1dluvk38g1j22fn96t4b. CSeq: 7237 INVITE. User-Agent: Grandstream GXP2000 1.2.2.26. Contact: sip:gsclient@192.168.0.106:5066;transport=udp. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE. Content-Type: application/sdp. Supported: replaces, timer. Content-Length: 216. . v=0. o=gsclient 8000 8000 IN IP4 192.168.0.106. s=SIP Call. c=IN IP4 192.168.0.106. t=0 0. m=audio 5026 RTP/AVP 0 13. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. m=audio 0 RTP/SAVPF 111 103 104 0 8 106 105 13 126.
cheers, Olli
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