Hi, I am trying to use Kamailio with WebTRC to make and receive calls from the browser.
Using the rpms from the Kamailio repository I have installed and tried the websocket config example from the source code. Using the first debug example from here https://www.kamailio.org/wiki/tutorials/tls/testing-and-debugginga I can see that my TLS wss setup is working. But I try JSCommunicator and sipML5 I get no success, they fail to register in Kamailio. From the logs I can see that something goes wrong with TLS. Doing some research, it is more like that several products are currently broken and not working...
So can anyone help me (share a config and tricky) or point to a document that would help me to get it running?
Thanks, Moacir
You can start with websockets module documentation. It gives details explaination of how it works and what needs to be done in config.
http://kamailio.org/docs/modules/4.4.x/modules/websocket.html#idp1214176
I have worked with JSSIP, SIPML5 and a few others using Kamailio and FreeSWITCH, all worked fine for me. So i guess the only problem is your understanding and configuration of the setup NOT these products.
Thank you.
On Mon, May 16, 2016 at 6:11 PM, Moacir Ferreira <moacirferreira@hotmail.com
wrote:
Hi, I am trying to use Kamailio with WebTRC to make and receive calls from the browser.
Using the rpms from the Kamailio repository I have installed and tried the websocket config example from the source code. Using the first debug example from here https://www.kamailio.org/wiki/tutorials/tls/testing-and-debugginga I can see that my TLS wss setup is working. But I try JSCommunicator and sipML5 I get no success, they fail to register in Kamailio. From the logs I can see that something goes wrong with TLS. Doing some research, it is more like that several products are currently broken and not working...
So can anyone help me (share a config and tricky) or point to a document that would help me to get it running?
Thanks, Moacir
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,
if you don't have a trusted certificate, then browse first to https://kamailioip:5061 (or your wss port) and accept the certificate.
If not working, maybe we can figure out what is the issue if you post the logs with debug=3 here.
Cheers, Daniel
On 16/05/16 18:11, Moacir Ferreira wrote:
Hi, I am trying to use Kamailio with WebTRC to make and receive calls from the browser.
Using the rpms from the Kamailio repository I have installed and tried the websocket config example from the source code. Using the first debug example from here https://www.kamailio.org/wiki/tutorials/tls/testing-and-debugginga I can see that my TLS wss setup is working. But I try JSCommunicator and sipML5 I get no success, they fail to register in Kamailio. From the logs I can see that something goes wrong with TLS. Doing some research, it is more like that several products are currently broken and not working...
So can anyone help me (share a config and tricky) or point to a document that would help me to get it running?
Thanks, Moacir
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
Great Daniel! Problem solved. Thanks, Moacir
To: sr-users@lists.sip-router.org From: miconda@gmail.com Date: Wed, 18 May 2016 07:08:02 +0200 Subject: Re: [SR-Users] WebRTC
Hello,
if you don't have a trusted certificate, then browse first to https://kamailioip:5061 (or your wss port) and accept the certificate.
If not working, maybe we can figure out what is the issue if you post the logs with debug=3 here.
Cheers,
Daniel
On 16/05/16 18:11, Moacir Ferreira wrote:
Hi,
I am trying to use Kamailio with WebTRC to make and receive calls from the browser.
Using the rpms from the Kamailio repository I have installed and tried the websocket config example from the source code. Using the first debug example from here https://www.kamailio.org/wiki/tutorials/tls/testing-and-debugginga I can see that my TLS wss setup is working. But I try JSCommunicator and sipML5 I get no success, they fail to register in Kamailio. From the logs I can see that something goes wrong with TLS. Doing some research, it is more like that several products are currently broken and not working...
So can anyone help me (share a config and tricky) or point to a document that would help me to get it running?
Thanks,
Moacir
_______________________________________________ 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 http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
_______________________________________________ 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
A question for the community:
What would be your best advice for a RTP proxy/transcoder to allow browser WebRTC calls to legacy VoIP?
Moacir
Hello,
kamailio+rtpengine should do this job quite well.
Cheers, Daniel
On 18/05/16 19:16, Moacir Ferreira wrote:
A question for the community:
What would be your best advice for a RTP proxy/transcoder to allow browser WebRTC calls to legacy VoIP?
Moacir
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
Hey Daniel,
If you say so, you probably right... I did not try it because on the sipwise GitHub (https://github.com/sipwise/rtpengine) they mention:
"Rtpengine does not (yet) support:
Repacketization or transcodingPlayback of pre-recorded streams/announcementsRecording of media streamsZRTP" So I did not test it... I will give it a try. By the way, from where should I download the source code? Also, any "tricky" (common mistake like my last one on WebRTC TLS) I should care about before trying it?
Thanks, Moacir
To: sr-users@lists.sip-router.org From: miconda@gmail.com Date: Wed, 18 May 2016 21:18:01 +0200 Subject: Re: [SR-Users] Browser WebRTC transcoder
Hello,
kamailio+rtpengine should do this job quite well.
Cheers,
Daniel
On 18/05/16 19:16, Moacir Ferreira wrote:
A question for the community:
What would be your best advice for a RTP proxy/transcoder to allow browser WebRTC calls to legacy VoIP?
Moacir
_______________________________________________ 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 http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
_______________________________________________ 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
On 18/05/16 04:57 PM, Moacir Ferreira wrote:
Hey Daniel,
If you say so, you probably right... I did not try it because on the sipwise GitHub (https://github.com/sipwise/rtpengine) they mention:
/"Rtpengine does not (yet) support:/ //
- /Repacketization or transcoding/
This refers to translating one audio codec into another (e.g. opus to PCM). Translating between RTP and SRTP (i.e. encrypting and decrypting) is supported.
Cheers
I did not dig into the problem but on my tests I saw that my (old) Grandstream phone was refusing the call for not having a compatible codec to talk with the offered ones by the browser (Firefox). Being this the case, I guess I must include a translator, and all routing logic, in between the callers. It points to Asterisk that I would like to avoid for now. But I guess this is not a problem that only affects me. Someone else must have faced this before. So the question still open: What solution would be recommended for such case?
Cheers, Moacir
To: sr-users@lists.sip-router.org From: rfuchs@sipwise.com Date: Wed, 18 May 2016 19:03:10 -0400 Subject: Re: [SR-Users] Browser WebRTC transcoder
On 18/05/16 04:57 PM, Moacir Ferreira wrote:
Hey Daniel,
If you say so, you probably right... I did not try it because on the sipwise GitHub (https://github.com/sipwise/rtpengine) they mention:
/"Rtpengine does not (yet) support:/ //
- /Repacketization or transcoding/
This refers to translating one audio codec into another (e.g. opus to PCM). Translating between RTP and SRTP (i.e. encrypting and decrypting) is supported.
Cheers
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
What codecs are supported by your grandstream? Isn't the g711 in the group?
Cheers, Daniel
On 19/05/16 01:51, Moacir Ferreira wrote:
I did not dig into the problem but on my tests I saw that my (old) Grandstream phone was refusing the call for not having a compatible codec to talk with the offered ones by the browser (Firefox). Being this the case, I guess I must include a translator, and all routing logic, in between the callers. It points to Asterisk that I would like to avoid for now. But I guess this is not a problem that only affects me. Someone else must have faced this before. So the question still open: What solution would be recommended for such case?
Cheers, Moacir
To: sr-users@lists.sip-router.org From: rfuchs@sipwise.com Date: Wed, 18 May 2016 19:03:10 -0400 Subject: Re: [SR-Users] Browser WebRTC transcoder
On 18/05/16 04:57 PM, Moacir Ferreira wrote:
Hey Daniel,
If you say so, you probably right... I did not try it because on the sipwise GitHub (https://github.com/sipwise/rtpengine) they mention:
/"Rtpengine does not (yet) support:/ //
- /Repacketization or transcoding/
This refers to translating one audio codec into another (e.g. opus to PCM). Translating between RTP and SRTP (i.e. encrypting and decrypting) is supported.
Cheers
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
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
Yes G711 is offered... I guess the Grandstream phone is "confused" about the way the SIP browser stack talks. I got the following logs taken on Kamailio, using ngrep:
######################### START OF LOG ########################## # Call from Grandstream to browser ################################## [root@sip ~]# ngrep -d any -qt -W byline port 5060 interface: any filter: ( port 5060 ) and (ip or ip6) ... U 2016/05/19 09:15:24.701195 192.168.100.85:5060 -> 192.168.100.159:5060 INVITE sip:1001@192.168.100.159 SIP/2.0. Via: SIP/2.0/UDP 192.168.100.85:5060;branch=z9hG4bK402e7247e47576ca. From: "Fernando" sip:1002@192.168.100.159;tag=b0d53bed080e1b0f. To: sip:1001@192.168.100.159. Contact: sip:1002@192.168.100.85:5060;transport=udp. Supported: replaces, timer, path. P-Early-Media: Supported. Proxy-Authorization: Digest username="1002", realm="192.168.100.159", algorithm=MD5, uri="sip:1001@192.168.100.159", nonce="Vz13SFc9dhwixIQaHqfrXSDvtNLkf+guJwNHi4A=", response="9c4f2fc2b7f179172fe9a6adf0d2f60f". Call-ID: dda078a035a57ecb@192.168.100.85. CSeq: 8700 INVITE. User-Agent: Grandstream BT200 1.2.5.3. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Type: application/sdp. Content-Length: 385. . v=0. o=1002 8000 8001 IN IP4 192.168.100.85. s=SIP Call. c=IN IP4 192.168.100.85. t=0 0. m=audio 11022 RTP/AVP 18 8 0 3 9 2 97 101. a=sendrecv. a=rtpmap:18 G729/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:9 G722/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=20. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11.
U 2016/05/19 09:15:24.710083 192.168.100.159:5060 -> 192.168.100.85:5060 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 192.168.100.85:5060;branch=z9hG4bK402e7247e47576ca. From: "Fernando" sip:1002@192.168.100.159;tag=b0d53bed080e1b0f. To: sip:1001@192.168.100.159;tag=56f8047e80cfcc9e90a3acc9609da1ba-9e91. Call-ID: dda078a035a57ecb@192.168.100.85. CSeq: 8700 INVITE. Server: kamailio (4.2.3 (x86_64/linux)). Content-Length: 0.
######################### END OF LOG ##########################
So the Grandstream offers a lot of codecs but will get a "Not Found" from Kamailio. Look in the other way:
######################### START OF LOG ########################## # Call from browser to Grandstream ################################## [root@sip ~]# ngrep -d any -qt -W byline port 5060 interface: any filter: ( port 5060 ) and (ip or ip6) U 2016/05/19 09:25:11.285826 192.168.100.159:5060 -> 192.168.100.85:5060 INVITE sip:1002@192.168.100.85:5060;transport=udp SIP/2.0. Record-Route: sip:192.168.100.159;r2=on;lr=on. Record-Route: sip:192.168.100.159:4443;transport=ws;r2=on;lr=on. Via: SIP/2.0/UDP 192.168.100.159;branch=z9hG4bK5ffb.16431fc4d55b38465ed5bfedf4063ead.0. Via: SIP/2.0/WSS df7jal23ls0d.invalid;received=192.168.100.249;branch=z9hG4bKTJM96ETVTs9QgSxvCWCBgWN1APMWKQVz;rport=59318. From: "Moacir"sip:1001@my.lab;tag=MhRTswgaXENAxcDi25HJ. To: sip:1002@my.lab. Contact: "Moacir"sips:1001@df7jal23ls0d.invalid;alias=192.168.100.249~59318~6;rtcweb-breaker=no;click2call=no;transport=wss;+g.oma.sip-im;language="en,fr". Call-ID: 2ecd7f42-ae98-563b-f0a7-00a4fe94c62d. CSeq: 15365 INVITE. Content-Type: application/sdp. Content-Length: 1182. Max-Forwards: 69. User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04. Organization: Doubango Telecom. . v=0. o=mozilla...THIS_IS_SDPARTA-46.0.1 947803314240298800 0 IN IP4 127.0.0.1. s=Doubango Telecom - firefox. t=0 0. a=sendrecv. a=fingerprint:sha-256 44:F1:6D:31:F4:D6:D9:43:1D:38:0B:8E:67:1E:5F:DD:10:F4:5F:1C:4B:7E:7A:47:F8:85:C4:93:40:A7:2D:5E. a=ice-options:trickle. a=msid-semantic:WMS *. m=audio 61455 UDP/TLS/RTP/SAVPF 109 9 0 8. c=IN IP4 192.168.100.249. a=candidate:0 1 UDP 2122252543 192.168.100.249 61455 typ host. a=candidate:1 1 UDP 2122187007 2001:0:5ef5:79fd:24be:2fcf:fa06:dbc8 61456 typ host. a=candidate:0 2 UDP 2122252542 192.168.100.249 61457 typ host. a=candidate:1 2 UDP 2122187006 2001:0:5ef5:7 U 2016/05/19 09:25:11.348673 192.168.100.85:5060 -> 192.168.100.159:5060 SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/UDP 192.168.100.159;branch=z9hG4bK5ffb.16431fc4d55b38465ed5bfedf4063ead.0. Via: SIP/2.0/WSS df7jal23ls0d.invalid;received=192.168.100.249;branch=z9hG4bKTJM96ETVTs9QgSxvCWCBgWN1APMWKQVz;rport=59318. Record-Route: sip:192.168.100.159;r2=on;lr=on. Record-Route: sip:192.168.100.159:4443;transport=ws;r2=on;lr=on. From: "Moacir"sip:1001@my.lab;tag=MhRTswgaXENAxcDi25HJ. To: sip:1002@my.lab;tag=9ea71e1d1f839cef. Call-ID: 2ecd7f42-ae98-563b-f0a7-00a4fe94c62d. CSeq: 15365 INVITE. User-Agent: Grandstream BT200 1.2.5.3. Warning: 304 GS "Media type not available". Content-Length: 0.
######################### END OF LOG ##########################
Here the Grandstream says "Media type not available". As I am not a real SIP guy, I got no clue why does not work!
Anyway, I am using the latest RPMs from Kamailo, running it using the websocket.cfg suggested configuration, no rtpengine installed on it. At the WebRTC, I am using sipml5 configuring it not to use STUN/TURN.
Cheers! Moacir
To: sr-users@lists.sip-router.org From: miconda@gmail.com Date: Thu, 19 May 2016 06:22:57 +0200 Subject: Re: [SR-Users] Browser WebRTC transcoder
What codecs are supported by your grandstream? Isn't the g711 in the group?
Cheers,
Daniel
On 19/05/16 01:51, Moacir Ferreira wrote:
I did not dig into the problem but on my tests I saw that my (old) Grandstream phone was refusing the call for not having a compatible codec to talk with the offered ones by the browser (Firefox). Being this the case, I guess I must include a translator, and all routing logic, in between the callers. It points to Asterisk that I would like to avoid for now. But I guess this is not a problem that only affects me. Someone else must have faced this before. So the question still open: What solution would be recommended for such case?
Cheers,
Moacir
> To: sr-users@lists.sip-router.org
> From: rfuchs@sipwise.com
> Date: Wed, 18 May 2016 19:03:10 -0400
> Subject: Re: [SR-Users] Browser WebRTC transcoder
>
> On 18/05/16 04:57 PM, Moacir Ferreira wrote:
> > Hey Daniel,
> >
> > If you say so, you probably right... I did not try it because on the
> > sipwise GitHub (https://github.com/sipwise/rtpengine) they mention:
> >
> > /"Rtpengine does not (yet) support:/
> > //
> >
> > * /Repacketization or transcoding/
>
> This refers to translating one audio codec into another (e.g. opus to
> PCM). Translating between RTP and SRTP (i.e. encrypting and decrypting)
> is supported.
>
> Cheers
>
> _______________________________________________
> 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
_______________________________________________ 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 http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com
_______________________________________________ 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
On 05/19/2016 04:52 AM, Moacir Ferreira wrote: ...
So the Grandstream offers a lot of codecs but will get a "Not Found" from Kamailio. Look in the other way:
That would be a SIP signalling (e.g. Kamailio config) problem. Perhaps a missing registration.
Here the Grandstream says "Media type not available". As I am not a real SIP guy, I got no clue why does not work!
This you can solve with rtpengine. The required codecs (PCM) are there, you just need to break the encryption (RTP <> SRTP) and some other features of WebRTC (ICE, BUNDLE, rtcp-mux, ...), all of which rtpengine can do.
Cheers
Yes, now it is working both ways. I was missing the rtpengine and right configuration. I will do some further testing to see if everything is functional.
Thanks,
To: sr-users@lists.sip-router.org From: rfuchs@sipwise.com Date: Thu, 19 May 2016 08:22:26 -0400 Subject: Re: [SR-Users] Browser WebRTC transcoder
On 05/19/2016 04:52 AM, Moacir Ferreira wrote: ...
So the Grandstream offers a lot of codecs but will get a "Not Found" from Kamailio. Look in the other way:
That would be a SIP signalling (e.g. Kamailio config) problem. Perhaps a missing registration.
Here the Grandstream says "Media type not available". As I am not a real SIP guy, I got no clue why does not work!
This you can solve with rtpengine. The required codecs (PCM) are there, you just need to break the encryption (RTP <> SRTP) and some other features of WebRTC (ICE, BUNDLE, rtcp-mux, ...), all of which rtpengine can do.
Cheers
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 -
M S -
Moacir Ferreira -
Richard Fuchs