[SR-Users] Kamailio with rtpproxy-ng and mediaproxy-ng: Error rewriting SDP

Tue Apr 1 20:41:09 CEST 2014

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 at testers.com <mailto:wsclient at testers.com> and
> gsclient at testers.com <mailto:gsclient at 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 at 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 at testers.com <mailto:sip%3Agsclient at testers.com>>.
> From: <sip:wsclient at testers.com
> <mailto:sip%3Awsclient at testers.com>>;tag=hhcd99tmvm.
> Call-ID: 1dluvk38g1j22fn96t4b.
> CSeq: 7237 INVITE.
> Contact: <sip:wsclient at testers.com
> <mailto:sip%3Awsclient at 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 at testers.com
> <mailto:sip%3Awsclient at testers.com>>;tag=hhcd99tmvm.
> To: <sip:gsclient at testers.com
> <mailto:sip%3Agsclient at 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 at 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 at testers.com
> <mailto:sip%3Agsclient at testers.com>>;tag=7875f08763872c34.
> From: <sip:wsclient at testers.com
> <mailto:sip%3Awsclient at 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 at 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 at testers.com <mailto:sip%3Agsclient at testers.com>>.
> From: <sip:wsclient at testers.com
> <mailto:sip%3Awsclient at testers.com>>;tag=hhcd99tmvm.
> Call-ID: 1dluvk38g1j22fn96t4b.
> CSeq: 7237 INVITE.
> Contact: <sip:wsclient at testers.com
> <mailto:sip%3Awsclient at 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 at testers.com
> <mailto:sip%3Awsclient at testers.com>>;tag=hhcd99tmvm.
> To: <sip:gsclient at testers.com
> <mailto:sip%3Agsclient at testers.com>>;tag=fb215901a251c9a0.
> Call-ID: 1dluvk38g1j22fn96t4b.
> CSeq: 7237 INVITE.
> User-Agent: Grandstream GXP2000 1.2.2.26.
> Contact: <sip:gsclient at 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 at testers.com
> <mailto:sip%3Awsclient at testers.com>>;tag=hhcd99tmvm.
> To: <sip:gsclient at testers.com
> <mailto:sip%3Agsclient at testers.com>>;tag=fb215901a251c9a0.
> Call-ID: 1dluvk38g1j22fn96t4b.
> CSeq: 7237 INVITE.
> User-Agent: Grandstream GXP2000 1.2.2.26.
> Contact: <sip:gsclient at 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
> 
> 
> 
> 
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
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> 

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