1 Apr
2014
1 Apr
'14
9:41 p.m.
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(a)testers.com <mailto:wsclient@testers.com> and
gsclient(a)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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