Hello.
I’m having some problems using websocket to communicate a webRTC client with the SIP world.
I have a Kamailio with a websocket port running on 5062, from that socket I’m receiving a SIP INVITE from a sipML5 client with 2531 bytes of length. When I made the capture on the other leg (the pure SIP side) I only see a SIP INVITE with 1500 bytes. Seems that something is fragmenting the packet but not putting all the parts together. Could this be a problem with Kamailio?. Does someone has the same problem?
Hope that someone could help me.
Best Regards,
Ricardo Martinez.-
Hello.
I’m having some problems using websocket to communicate a webRTC client with the SIP world.
I have a Kamailio with a websocket port running on 5062, from that socket I’m receiving a SIP INVITE from a sipML5 client with 2531 bytes of length. When I made the capture on the other leg (the pure SIP side) I only see a SIP INVITE with 1500 bytes. Seems that something is fragmenting the packet but not putting all the parts together. Could this be a problem with Kamailio?. Does someone has the same problem?
Hope that someone could help me.
Best Regards,
Ricardo Martinez.-
Hello Daniel.
I have printed the $mb in the kamailio debug and the $ml :
The SIP message in the client side has 2759 bytes.
This is what I get from the kamailio at the entrance leg :
Oct 29 17:27:24 webrtc /usr/local/sbin/kamailio[846]: DEBUG: <script>: Mensaje SIP INVITE de 2759 bytes
Oct 29 17:27:24 webrtc /usr/local/sbin/kamailio[846]: DEBUG: <script>: Mensaje SIP entrada : INVITE sip:005622408596@200.100.14.88 SIP/2.0#015#012Via: SIP/2.0/WS df7jal
23ls0d.invalid;branch=z9hG4bK7k8psiscD4Ni3RO86o9WXxpbUiCeQAMw;rport#015#012From: "Ricardo Martinez"sip:12234@200.100.155.88;tag=VKVFOTMCZGZkvOcoMpvl#015#012To: <sip:0
05622408596@200.100.14.88>#015#012Contact: "Ricardo Martinez"sip:12234@df7jal23ls0d.invalid ;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;+sip.ice;langua
ge="en,fr"#015#012Call-ID: 7e329fce-f9db-e096-a647-f0bf755a46cd#015#012CSeq: 4803 INVITE#015#012Content-Type: application/sdp#015#012Content-Length: 2077#015#012Route:
sip:200.100.14.88:5060;lr;sipml5-outbound;transport=udp#015#012Max-Forwards: 70#015#012User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18#015#012Organization: Doubango
Telecom#015#012#015#012v=0#015#012o=- 5131738957380134000 2 IN IP4 127.0.0.1#015#012s=Doubango Telecom - chrome#015#012t=0 0#015#012a=group:BUNDLE audio#015#012a=msid-
semantic: WMS Ii2wNuOAldC4ODWBex4rBga19yyGclSsmJNx#015#012m=audio 57516 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126#015#012c=IN IP4 200.100.15.218#015#012a=rtcp:57
516 IN IP4 200.100.15.218#015#012a=candidate:2975380780 1 udp 2122194687 100.150.0.30 55624 typ host generation 0#015#012a=candidate:2975380780 2 udp 2122194687 100.150
.0.30 55624 typ host generation 0#015#012a=candidate:1374240324 1 udp 2122129151 200.100.14.102 55625 typ host generation 0#015#012a=candidate:1374240324 2 udp 21221291
51 200.100.14.102 55625 typ host generation 0#015#012a=candidate:4292561372 1 tcp 1518214911 100.150.0.30 0 typ host generation 0#015#012a=candidate:4292561372 2 tcp 15
18214911 100.150.0.30 0 typ host generation 0#015#012a=candidate:527090356 1 tcp 1518149375 200.100.14.102 0 typ host generation 0#015#012a=candidate:527090356 2 tcp 15
18149375 200.100.14.102 0 typ host generation 0#015#012a=candidate:643094781 1 udp 41754367 200.100.15.218 57516 typ relay raddr 200.100.14.102 rport 55631 generation 0
#015#012a=candidate:643094781 2 udp 41754367
And in the output leg :
Oct 29 17:27:24 webrtc /usr/local/sbin/kamailio[846]: DEBUG: <script>: Mensaje SIP INVITE de 2759 bytes
Oct 29 17:27:24 webrtc /usr/local/sbin/kamailio[846]: DEBUG: <script>: Mensaje SIP salida : INVITE sip:005622408596@200.100.14.88 SIP/2.0#015#012Via: SIP/2.0/WS df7jal2
3ls0d.invalid;branch=z9hG4bK7k8psiscD4Ni3RO86o9WXxpbUiCeQAMw;rport#015#012From: "Ricardo Martinez"sip:12234@200.100.14.88;tag=VKVFOTMCZGZkvOcoMpvl#015#012To: <sip:005
622408596@200.100.14.88>#015#012Contact: "Ricardo Martinez"sip:12234@df7jal23ls0d.invalid ;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;+sip.ice;language
="en,fr"#015#012Call-ID: 7e329fce-f9db-e096-a647-f0bf755a46cd#015#012CSeq: 4803 INVITE#015#012Content-Type: application/sdp#015#012Content-Length: 2077#015#012Route: <s
ip:200.100.14.88:5060;lr;sipml5-outbound;transport=udp>#015#012Max-Forwards: 69#015#012User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18#015#012Organization: Doubango T
elecom#015#012#015#012v=0#015#012o=- 5131738957380134000 2 IN IP4 127.0.0.1#015#012s=Doubango Telecom - chrome#015#012t=0 0#015#012a=group:BUNDLE audio#015#012a=msid-se
mantic: WMS Ii2wNuOAldC4ODWBex4rBga19yyGclSsmJNx#015#012m=audio 57516 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126#015#012c=IN IP4 200.100.15.218#015#012a=rtcp:5751
6 IN IP4 200.100.15.218#015#012a=candidate:2975380780 1 udp 2122194687 100.150.0.30 55624 typ host generation 0#015#012a=candidate:2975380780 2 udp 2122194687 100.150.0
.30 55624 typ host generation 0#015#012a=candidate:1374240324 1 udp 2122129151 200.100.14.102 55625 typ host generation 0#015#012a=candidate:1374240324 2 udp 2122129151
200.100.14.102 55625 typ host generation 0#015#012a=candidate:4292561372 1 tcp 1518214911 100.150.0.30 0 typ host generation 0#015#012a=candidate:4292561372 2 tcp 1518
214911 100.150.0.30 0 typ host generation 0#015#012a=candidate:527090356 1 tcp 1518149375 200.100.14.102 0 typ host generation 0#015#012a=candidate:527090356 2 tcp 1518
149375 200.100.14.102 0 typ host generation 0#015#012a=candidate:643094781 1 udp 41754367 200.100.15.218 57516 typ relay raddr 200.100.14.102 rport 55631 generation 0#0
15#012a=candidate:643094781 2 udp 41754367
This call goes to a Gateway and I receive the message in two parts :
U 200.100.14.86:50602 -> 200.100.15.112:5060
INVITE sip:005622408596@200.100.14.86 SIP/2.0.
Via: SIP/2.0/UDP 200.100.14.86:50602;branch=z9hG4bKac2107949381.
Max-Forwards: 10.
From: "Ricardo Martinez" sip:12234@200.100.14.88;tag=1c2107343225.
To: sip:005622408596@200.100.14.88.
Call-ID: 21072976071782014233052@200.100.14.86.
CSeq: 1 INVITE.
Contact: "Ricardo Martinez" sip:12234@200.100.14.86:50602 ;rtcweb-breaker=no;click2call=no;+g.oma.sip-im;+sip.ice;language="en,fr".
User-Agent: Mediant 800 - MSBG/v.6.60A.217.001.
Content-Type: application/sdp.
Content-Length: 1512.
Organization: Doubango Telecom.
.
v=0.
o=- 2107099153 2107099119 IN IP4 200.100.14.86.
s=Doubango Telecom - chrome.
t=0 0.
a=msid-semantic: WMS A77eee9RXVQxRqfZbjG52wdDjNozkaKFTIVk.
m=audio 7000 UDP/TLS/R 18 13.
c=IN IP4 200.100.14.86.
a=candidate:2975380780 1 udp 2122194687 100.150.0.30 56398 typ host generation 0.
a=candidate:2975380780 2 udp 2122194687 100.150.0.30 56398 typ host generation 0.
a=candidate:1374240324 1 udp 2122129151 200.100.14.102 56399 typ host generation 0.
a=candidate:1374240324 2 udp 2122129151 200.100.14.102 56399 typ host generation 0.
a=candidate:4292561372 1 tcp 1518214911 100.150.0.30 0 typ host generation 0.
a=candidate:4292561372 2 tcp 1518214911 100.150.0.30 0 typ host generation 0.
a=candidate:527090356 1 tcp 1518149375 200.100.14.102 0 typ host generation 0.
a=candidate:527090356 2 tcp 1518149375 200.100.14.102 0 typ host generation 0.
a=candidate:643094781 1 udp 41754367 200.100.15.218 60088 typ relay
U 200.100.14.86 -> 200.100.15.112 +43945@1480:603
raddr 200.100.14.102 rport 55129 generation 0.
a=candidate:643094781 2 udp 41754367 200.100.15.218 60088 typ relay raddr 200.100.14.102 rport 55129 generation 0.
a=ice-ufrag:W7TkIfz4YwhbjShF.
a=ice-pwd:6uOQuwCfJNAovtb+RRMVRhCL.
a=ice-options:google-ice.
a=fingerprint:sha-256 F8:F4:4E:ED:17:7A:9C:0A:E9:C5:F0:97:C1:CF:C3:88:05:64:6C:9A:9B:F1:56:0F:30:08:86:86:FD:7D:E8:C9.
a=setup:actpass.
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level.
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time.
a=sendrecv.
a=rtcp-mux.
a=rtpmap:13 CN/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
What could be the problem?
Thanks in advance.
Ricardo Martinez.-
*Subject:* websocket and SIP
Hello.
I’m having some problems using websocket to communicate a webRTC client with the SIP world.
I have a Kamailio with a websocket port running on 5062, from that socket I’m receiving a SIP INVITE from a sipML5 client with 2531 bytes of length. When I made the capture on the other leg (the pure SIP side) I only see a SIP INVITE with 1500 bytes. Seems that something is fragmenting the packet but not putting all the parts together. Could this be a problem with Kamailio?. Does someone has the same problem?
Hope that someone could help me.
Best Regards,
Ricardo Martinez.-
Hello,
the problem can be UDP fragmentation -- the gateway stack is not able to handle UDP fragments. If the gateway supports tcp, then use this transport layer.
Cheers, Daniel
On 29/10/14 21:42, Ricardo Martinez wrote:
Hello Daniel.
I have printed the $mb in the kamailio debug and the $ml :
The SIP message in the client side has 2759 bytes.
This is what I get from the kamailio at the entrance leg :
Oct 29 17:27:24 webrtc /usr/local/sbin/kamailio[846]: DEBUG: <script>: Mensaje SIP INVITE de 2759 bytes
Oct 29 17:27:24 webrtc /usr/local/sbin/kamailio[846]: DEBUG: <script>: Mensaje SIP entrada : INVITE sip:005622408596@200.100.14.88 mailto:sip%3A005622408596@200.100.14.88 SIP/2.0#015#012Via: SIP/2.0/WS df7jal
23ls0d.invalid;branch=z9hG4bK7k8psiscD4Ni3RO86o9WXxpbUiCeQAMw;rport#015#012From: "Ricardo Martinez"<sip:12234@200.100.155.88 mailto:sip%3A12234@200.100.155.88>;tag=VKVFOTMCZGZkvOcoMpvl#015#012To: <sip:0
05622408596@200.100.14.88 mailto:05622408596@200.100.14.88>#015#012Contact: "Ricardo Martinez"sip:12234@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;+sip.ice;langua
ge="en,fr"#015#012Call-ID: 7e329fce-f9db-e096-a647-f0bf755a46cd#015#012CSeq: 4803 INVITE#015#012Content-Type: application/sdp#015#012Content-Length: 2077#015#012Route:
sip:200.100.14.88:5060;lr;sipml5-outbound;transport=udp#015#012Max-Forwards: 70#015#012User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18#015#012Organization: Doubango
Telecom#015#012#015#012v=0#015#012o=- 5131738957380134000 2 IN IP4 127.0.0.1#015#012s=Doubango Telecom - chrome#015#012t=0 0#015#012a=group:BUNDLE audio#015#012a=msid-
semantic: WMS Ii2wNuOAldC4ODWBex4rBga19yyGclSsmJNx#015#012m=audio 57516 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126#015#012c=IN IP4 200.100.15.218#015#012a=rtcp:57
516 IN IP4 200.100.15.218#015#012a=candidate:2975380780 1 udp 2122194687 100.150.0.30 55624 typ host generation 0#015#012a=candidate:2975380780 2 udp 2122194687 100.150
.0.30 55624 typ host generation 0#015#012a=candidate:1374240324 1 udp 2122129151 200.100.14.102 55625 typ host generation 0#015#012a=candidate:1374240324 2 udp 21221291
51 200.100.14.102 55625 typ host generation 0#015#012a=candidate:4292561372 1 tcp 1518214911 100.150.0.30 0 typ host generation 0#015#012a=candidate:4292561372 2 tcp 15
18214911 100.150.0.30 0 typ host generation 0#015#012a=candidate:527090356 1 tcp 1518149375 200.100.14.102 0 typ host generation 0#015#012a=candidate:527090356 2 tcp 15
18149375 200.100.14.102 0 typ host generation 0#015#012a=candidate:643094781 1 udp 41754367 200.100.15.218 57516 typ relay raddr 200.100.14.102 rport 55631 generation 0
#015#012a=candidate:643094781 2 udp 41754367
And in the output leg :
Oct 29 17:27:24 webrtc /usr/local/sbin/kamailio[846]: DEBUG: <script>: Mensaje SIP INVITE de 2759 bytes
Oct 29 17:27:24 webrtc /usr/local/sbin/kamailio[846]: DEBUG: <script>: Mensaje SIP salida : INVITE sip:005622408596@200.100.14.88 mailto:sip%3A005622408596@200.100.14.88 SIP/2.0#015#012Via: SIP/2.0/WS df7jal2
3ls0d.invalid;branch=z9hG4bK7k8psiscD4Ni3RO86o9WXxpbUiCeQAMw;rport#015#012From: "Ricardo Martinez"<sip:12234@200.100.14.88 mailto:sip%3A12234@200.100.14.88>;tag=VKVFOTMCZGZkvOcoMpvl#015#012To: <sip:005
622408596@200.100.14.88 mailto:622408596@200.100.14.88>#015#012Contact: "Ricardo Martinez"sip:12234@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;+sip.ice;language
="en,fr"#015#012Call-ID: 7e329fce-f9db-e096-a647-f0bf755a46cd#015#012CSeq: 4803 INVITE#015#012Content-Type: application/sdp#015#012Content-Length: 2077#015#012Route: <s
ip:200.100.14.88:5060;lr;sipml5-outbound;transport=udp>#015#012Max-Forwards: 69#015#012User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18#015#012Organization: Doubango T
elecom#015#012#015#012v=0#015#012o=- 5131738957380134000 2 IN IP4 127.0.0.1#015#012s=Doubango Telecom - chrome#015#012t=0 0#015#012a=group:BUNDLE audio#015#012a=msid-se
mantic: WMS Ii2wNuOAldC4ODWBex4rBga19yyGclSsmJNx#015#012m=audio 57516 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126#015#012c=IN IP4 200.100.15.218#015#012a=rtcp:5751
6 IN IP4 200.100.15.218#015#012a=candidate:2975380780 1 udp 2122194687 100.150.0.30 55624 typ host generation 0#015#012a=candidate:2975380780 2 udp 2122194687 100.150.0
.30 55624 typ host generation 0#015#012a=candidate:1374240324 1 udp 2122129151 200.100.14.102 55625 typ host generation 0#015#012a=candidate:1374240324 2 udp 2122129151
200.100.14.102 55625 typ host generation 0#015#012a=candidate:4292561372 1 tcp 1518214911 100.150.0.30 0 typ host generation 0#015#012a=candidate:4292561372 2 tcp 1518
214911 100.150.0.30 0 typ host generation 0#015#012a=candidate:527090356 1 tcp 1518149375 200.100.14.102 0 typ host generation 0#015#012a=candidate:527090356 2 tcp 1518
149375 200.100.14.102 0 typ host generation 0#015#012a=candidate:643094781 1 udp 41754367 200.100.15.218 57516 typ relay raddr 200.100.14.102 rport 55631 generation 0#0
15#012a=candidate:643094781 2 udp 41754367
This call goes to a Gateway and I receive the message in two parts :
U 200.100.14.86:50602 http://200.100.14.86:50602 -> 200.100.15.112:5060 http://200.100.15.112:5060
INVITE sip:005622408596@200.100.14.86 mailto:sip%3A005622408596@200.100.14.86 SIP/2.0.
Via: SIP/2.0/UDP 200.100.14.86:50602;branch=z9hG4bKac2107949381.
Max-Forwards: 10.
From: "Ricardo Martinez" <sip:12234@200.100.14.88 mailto:sip%3A12234@200.100.14.88>;tag=1c2107343225.
To: <sip:005622408596@200.100.14.88 mailto:sip%3A005622408596@200.100.14.88>.
Call-ID: 21072976071782014233052@200.100.14.86 mailto:21072976071782014233052@200.100.14.86.
CSeq: 1 INVITE.
Contact: "Ricardo Martinez" sip:12234@200.100.14.86:50602;rtcweb-breaker=no;click2call=no;+g.oma.sip-im;+sip.ice;language="en,fr".
User-Agent: Mediant 800 - MSBG/v.6.60A.217.001.
Content-Type: application/sdp.
Content-Length: 1512.
Organization: Doubango Telecom.
.
v=0.
o=- 2107099153 2107099119 IN IP4 200.100.14.86.
s=Doubango Telecom - chrome.
t=0 0.
a=msid-semantic: WMS A77eee9RXVQxRqfZbjG52wdDjNozkaKFTIVk.
m=audio 7000 UDP/TLS/R 18 13.
c=IN IP4 200.100.14.86.
a=candidate:2975380780 1 udp 2122194687 100.150.0.30 56398 typ host generation 0.
a=candidate:2975380780 2 udp 2122194687 100.150.0.30 56398 typ host generation 0.
a=candidate:1374240324 1 udp 2122129151 200.100.14.102 56399 typ host generation 0.
a=candidate:1374240324 2 udp 2122129151 200.100.14.102 56399 typ host generation 0.
a=candidate:4292561372 1 tcp 1518214911 100.150.0.30 0 typ host generation 0.
a=candidate:4292561372 2 tcp 1518214911 100.150.0.30 0 typ host generation 0.
a=candidate:527090356 1 tcp 1518149375 200.100.14.102 0 typ host generation 0.
a=candidate:527090356 2 tcp 1518149375 200.100.14.102 0 typ host generation 0.
a=candidate:643094781 1 udp 41754367 200.100.15.218 60088 typ relay
U 200.100.14.86 -> 200.100.15.112 +43945@1480:603
raddr 200.100.14.102 rport 55129 generation 0.
a=candidate:643094781 2 udp 41754367 200.100.15.218 60088 typ relay raddr 200.100.14.102 rport 55129 generation 0.
a=ice-ufrag:W7TkIfz4YwhbjShF.
a=ice-pwd:6uOQuwCfJNAovtb+RRMVRhCL.
a=ice-options:google-ice.
a=fingerprint:sha-256 F8:F4:4E:ED:17:7A:9C:0A:E9:C5:F0:97:C1:CF:C3:88:05:64:6C:9A:9B:F1:56:0F:30:08:86:86:FD:7D:E8:C9.
a=setup:actpass.
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level.
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time.
a=sendrecv.
a=rtcp-mux.
a=rtpmap:13 CN/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
What could be the problem?
Thanks in advance.
Ricardo Martinez.-
*Subject:*websocket and SIP
Hello.
I’m having some problems using websocket to communicate a webRTC client with the SIP world.
I have a Kamailio with a websocket port running on 5062, from that socket I’m receiving a SIP INVITE from a sipML5 client with 2531 bytes of length. When I made the capture on the other leg (the pure SIP side) I only see a SIP INVITE with 1500 bytes. Seems that something is fragmenting the packet but not putting all the parts together. Could this be a problem with Kamailio?. Does someone has the same problem?
Hope that someone could help me.
Best Regards,
Ricardo Martinez.-
Hi Ricardo, I have a similar setup working:
sipml5 -wss-> Kamailio -udp-> GW (FS)
I use Freeswitch with UDP and works fine, as you can see initial Invite with SDP for Webrtc clients using sipMl5 is normally pretty big (audio+video) and normally if you are proxying that message the remote end should reassamble it, the best way to test it is to a tcpdump on the other side and see what the OS is receiving.
On Thu, Oct 30, 2014 at 3:52 AM, Daniel-Constantin Mierla <miconda@gmail.com
wrote:
Hello,
the problem can be UDP fragmentation -- the gateway stack is not able to handle UDP fragments. If the gateway supports tcp, then use this transport layer.
Cheers, Daniel
On 29/10/14 21:42, Ricardo Martinez wrote:
Hello Daniel.
I have printed the $mb in the kamailio debug and the $ml :
The SIP message in the client side has 2759 bytes.
This is what I get from the kamailio at the entrance leg :
Oct 29 17:27:24 webrtc /usr/local/sbin/kamailio[846]: DEBUG: <script>: Mensaje SIP INVITE de 2759 bytes
Oct 29 17:27:24 webrtc /usr/local/sbin/kamailio[846]: DEBUG: <script>: Mensaje SIP entrada : INVITE sip:005622408596@200.100.14.88 SIP/2.0#015#012Via: SIP/2.0/WS df7jal
23ls0d.invalid;branch=z9hG4bK7k8psiscD4Ni3RO86o9WXxpbUiCeQAMw;rport#015#012From: "Ricardo Martinez"sip:12234@200.100.155.88;tag=VKVFOTMCZGZkvOcoMpvl#015#012To: <sip:0
05622408596@200.100.14.88>#015#012Contact: "Ricardo Martinez" sip:12234@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws ;+g.oma.sip-im;+sip.ice;langua
ge="en,fr"#015#012Call-ID: 7e329fce-f9db-e096-a647-f0bf755a46cd#015#012CSeq: 4803 INVITE#015#012Content-Type: application/sdp#015#012Content-Length: 2077#015#012Route:
sip:200.100.14.88:5060;lr;sipml5-outbound;transport=udp#015#012Max-Forwards: 70#015#012User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18#015#012Organization: Doubango
Telecom#015#012#015#012v=0#015#012o=- 5131738957380134000 2 IN IP4 127.0.0.1#015#012s=Doubango Telecom - chrome#015#012t=0 0#015#012a=group:BUNDLE audio#015#012a=msid-
semantic: WMS Ii2wNuOAldC4ODWBex4rBga19yyGclSsmJNx#015#012m=audio 57516 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126#015#012c=IN IP4 200.100.15.218#015#012a=rtcp:57
516 IN IP4 200.100.15.218#015#012a=candidate:2975380780 1 udp 2122194687 100.150.0.30 55624 typ host generation 0#015#012a=candidate:2975380780 2 udp 2122194687 100.150
.0.30 55624 typ host generation 0#015#012a=candidate:1374240324 1 udp 2122129151 200.100.14.102 55625 typ host generation 0#015#012a=candidate:1374240324 2 udp 21221291
51 200.100.14.102 55625 typ host generation 0#015#012a=candidate:4292561372 1 tcp 1518214911 100.150.0.30 0 typ host generation 0#015#012a=candidate:4292561372 2 tcp 15
18214911 100.150.0.30 0 typ host generation 0#015#012a=candidate:527090356 1 tcp 1518149375 200.100.14.102 0 typ host generation 0#015#012a=candidate:527090356 2 tcp 15
18149375 200.100.14.102 0 typ host generation 0#015#012a=candidate:643094781 1 udp 41754367 200.100.15.218 57516 typ relay raddr 200.100.14.102 rport 55631 generation 0
#015#012a=candidate:643094781 2 udp 41754367
And in the output leg :
Oct 29 17:27:24 webrtc /usr/local/sbin/kamailio[846]: DEBUG: <script>: Mensaje SIP INVITE de 2759 bytes
Oct 29 17:27:24 webrtc /usr/local/sbin/kamailio[846]: DEBUG: <script>: Mensaje SIP salida : INVITE sip:005622408596@200.100.14.88 SIP/2.0#015#012Via: SIP/2.0/WS df7jal2
3ls0d.invalid;branch=z9hG4bK7k8psiscD4Ni3RO86o9WXxpbUiCeQAMw;rport#015#012From: "Ricardo Martinez"sip:12234@200.100.14.88;tag=VKVFOTMCZGZkvOcoMpvl#015#012To: <sip:005
622408596@200.100.14.88>#015#012Contact: "Ricardo Martinez" sip:12234@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws ;+g.oma.sip-im;+sip.ice;language
="en,fr"#015#012Call-ID: 7e329fce-f9db-e096-a647-f0bf755a46cd#015#012CSeq: 4803 INVITE#015#012Content-Type: application/sdp#015#012Content-Length: 2077#015#012Route: <s
ip:200.100.14.88:5060;lr;sipml5-outbound;transport=udp>#015#012Max-Forwards: 69#015#012User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18#015#012Organization: Doubango T
elecom#015#012#015#012v=0#015#012o=- 5131738957380134000 2 IN IP4 127.0.0.1#015#012s=Doubango Telecom - chrome#015#012t=0 0#015#012a=group:BUNDLE audio#015#012a=msid-se
mantic: WMS Ii2wNuOAldC4ODWBex4rBga19yyGclSsmJNx#015#012m=audio 57516 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126#015#012c=IN IP4 200.100.15.218#015#012a=rtcp:5751
6 IN IP4 200.100.15.218#015#012a=candidate:2975380780 1 udp 2122194687 100.150.0.30 55624 typ host generation 0#015#012a=candidate:2975380780 2 udp 2122194687 100.150.0
.30 55624 typ host generation 0#015#012a=candidate:1374240324 1 udp 2122129151 200.100.14.102 55625 typ host generation 0#015#012a=candidate:1374240324 2 udp 2122129151
200.100.14.102 55625 typ host generation 0#015#012a=candidate:4292561372 1 tcp 1518214911 100.150.0.30 0 typ host generation 0#015#012a=candidate:4292561372 2 tcp 1518
214911 100.150.0.30 0 typ host generation 0#015#012a=candidate:527090356 1 tcp 1518149375 200.100.14.102 0 typ host generation 0#015#012a=candidate:527090356 2 tcp 1518
149375 200.100.14.102 0 typ host generation 0#015#012a=candidate:643094781 1 udp 41754367 200.100.15.218 57516 typ relay raddr 200.100.14.102 rport 55631 generation 0#0
15#012a=candidate:643094781 2 udp 41754367
This call goes to a Gateway and I receive the message in two parts :
U 200.100.14.86:50602 -> 200.100.15.112:5060
INVITE sip:005622408596@200.100.14.86 SIP/2.0.
Via: SIP/2.0/UDP 200.100.14.86:50602;branch=z9hG4bKac2107949381.
Max-Forwards: 10.
From: "Ricardo Martinez" sip:12234@200.100.14.88;tag=1c2107343225.
To: sip:005622408596@200.100.14.88.
Call-ID: 21072976071782014233052@200.100.14.86.
CSeq: 1 INVITE.
Contact: "Ricardo Martinez" sip:12234@200.100.14.86:50602;rtcweb-breaker=no;click2call=no ;+g.oma.sip-im;+sip.ice;language="en,fr".
User-Agent: Mediant 800 - MSBG/v.6.60A.217.001.
Content-Type: application/sdp.
Content-Length: 1512.
Organization: Doubango Telecom.
.
v=0.
o=- 2107099153 2107099119 IN IP4 200.100.14.86.
s=Doubango Telecom - chrome.
t=0 0.
a=msid-semantic: WMS A77eee9RXVQxRqfZbjG52wdDjNozkaKFTIVk.
m=audio 7000 UDP/TLS/R 18 13.
c=IN IP4 200.100.14.86.
a=candidate:2975380780 1 udp 2122194687 100.150.0.30 56398 typ host generation 0.
a=candidate:2975380780 2 udp 2122194687 100.150.0.30 56398 typ host generation 0.
a=candidate:1374240324 1 udp 2122129151 200.100.14.102 56399 typ host generation 0.
a=candidate:1374240324 2 udp 2122129151 200.100.14.102 56399 typ host generation 0.
a=candidate:4292561372 1 tcp 1518214911 100.150.0.30 0 typ host generation 0.
a=candidate:4292561372 2 tcp 1518214911 100.150.0.30 0 typ host generation 0.
a=candidate:527090356 1 tcp 1518149375 200.100.14.102 0 typ host generation 0.
a=candidate:527090356 2 tcp 1518149375 200.100.14.102 0 typ host generation 0.
a=candidate:643094781 1 udp 41754367 200.100.15.218 60088 typ relay
U 200.100.14.86 -> 200.100.15.112 +43945@1480:603
raddr 200.100.14.102 rport 55129 generation 0.
a=candidate:643094781 2 udp 41754367 200.100.15.218 60088 typ relay raddr 200.100.14.102 rport 55129 generation 0.
a=ice-ufrag:W7TkIfz4YwhbjShF.
a=ice-pwd:6uOQuwCfJNAovtb+RRMVRhCL.
a=ice-options:google-ice.
a=fingerprint:sha-256 F8:F4:4E:ED:17:7A:9C:0A:E9:C5:F0:97:C1:CF:C3:88:05:64:6C:9A:9B:F1:56:0F:30:08:86:86:FD:7D:E8:C9.
a=setup:actpass.
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level.
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time.
a=sendrecv.
a=rtcp-mux.
a=rtpmap:13 CN/8000.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
What could be the problem?
Thanks in advance.
Ricardo Martinez.-
*Subject:* websocket and SIP
Hello.
I’m having some problems using websocket to communicate a webRTC client with the SIP world.
I have a Kamailio with a websocket port running on 5062, from that socket I’m receiving a SIP INVITE from a sipML5 client with 2531 bytes of length. When I made the capture on the other leg (the pure SIP side) I only see a SIP INVITE with 1500 bytes. Seems that something is fragmenting the packet but not putting all the parts together. Could this be a problem with Kamailio?. Does someone has the same problem?
Hope that someone could help me.
Best Regards,
Ricardo Martinez.-
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
have you printed the message in syslog with $mb to see its content?
What is the content-lenght value sent by client and the one sent out by kamailio.
Cheers, Daniel
On 28/10/14 16:21, Ricardo Martinez wrote:
Hello.
I’m having some problems using websocket to communicate a webRTC client with the SIP world.
I have a Kamailio with a websocket port running on 5062, from that socket I’m receiving a SIP INVITE from a sipML5 client with 2531 bytes of length. When I made the capture on the other leg (the pure SIP side) I only see a SIP INVITE with 1500 bytes. Seems that something is fragmenting the packet but not putting all the parts together. Could this be a problem with Kamailio?. Does someone has the same problem?
Hope that someone could help me.
Best Regards,
Ricardo Martinez.-
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-
Daniel-Constantin Mierla -
Gonzalo Gasca -
Ricardo Martinez