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.-

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

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