Kamailio is just acting as a proxy and protocol modifier so to say. It is workin with rtpengine from sipwise to handle media as evident from he logs. This architectue uses a TURN server and the browser is chrome with latest updates.
The only thing whih I haven't done is enable TLS in kamailio and create certs. (which I'm not completely sure how to do).. Also, does it necessitates to have Apache ruuning https on 443 ?
Thanks in advance
Sent from Samsung Mobile
-------- Original message -------- From: Gonzalo Gasca Meza gascagonzalo@gmail.com Date:27/01/2015 4:07 AM (GMT+05:30) To: "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org Subject: Re: [SR-Users] Need help on WebRTC with Kamailio as proxy
Are you terminating media in Kamailio or just handling WS communication? If yes which version of Kamailio and rtp-proxy ? Have you tried passing media directly between Browser and Kamailio with any TURN server?
Are you using latest Chrome version or FF ?
A working sample config using the following architecture:
https://github.com/spicyramen/llamato/tree/LlamatoReg
signalling: sipml5 -- ws/wss --> Ec2 Kamailio --sip udp--> FS --sip udp--> * media: sipml5 ------------------------------------------------------------------------> *
On Mon, Jan 26, 2015 at 12:44 PM, Rahul MathuR rahul.ultimate@gmail.com wrote: Hi Richard,
Thanks for spending some cycles on it.
It is OpenSSL 1.0.1e-fips 11 Feb 2013
On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs rfuchs@sipwise.com wrote: On 26/01/15 02:21 PM, Rahul MathuR wrote: Hello,
I am totally struck at a point while implementing Kamailio as proxy for WebRTC enabled UAC (Jssip). I am using Google's TURN server (rfc5766-turn-server for ICE/STUN). I am able to get to the point where the SIP server sends 183 session in progress to kamailio but after that I can only see - "STUN: using this candidate" "Successful STUN binding request from .." "SRTP output wanted, but no crypto suite was negotiated"
This is fairly strange:
Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated
Are you running a very old OpenSSL version by any chance?
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
Any thoughts on this gents ?
On Tue, Jan 27, 2015 at 8:09 AM, rahul.ultimate rahul.ultimate@gmail.com wrote:
Kamailio is just acting as a proxy and protocol modifier so to say. It is workin with rtpengine from sipwise to handle media as evident from he logs. This architectue uses a TURN server and the browser is chrome with latest updates.
The only thing whih I haven't done is enable TLS in kamailio and create certs. (which I'm not completely sure how to do).. Also, does it necessitates to have Apache ruuning https on 443 ?
Thanks in advance
Sent from Samsung Mobile
-------- Original message -------- From: Gonzalo Gasca Meza Date:27/01/2015 4:07 AM (GMT+05:30) To: "Kamailio (SER) - Users Mailing List" Subject: Re: [SR-Users] Need help on WebRTC with Kamailio as proxy
Are you terminating media in Kamailio or just handling WS communication? If yes which version of Kamailio and rtp-proxy ? Have you tried passing media directly between Browser and Kamailio with any TURN server?
Are you using latest Chrome version or FF ?
A working sample config using the following architecture:
https://github.com/spicyramen/llamato/tree/LlamatoReg
signalling: sipml5 -- ws/wss --> Ec2 Kamailio --sip udp--> FS --sip udp--> * media: sipml5 ------------------------------------------------------------------------> *
On Mon, Jan 26, 2015 at 12:44 PM, Rahul MathuR rahul.ultimate@gmail.com wrote:
Hi Richard,
Thanks for spending some cycles on it.
It is OpenSSL 1.0.1e-fips 11 Feb 2013
On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs rfuchs@sipwise.com wrote:
On 26/01/15 02:21 PM, Rahul MathuR wrote:
Hello,
I am totally struck at a point while implementing Kamailio as proxy for WebRTC enabled UAC (Jssip). I am using Google's TURN server (rfc5766-turn-server for ICE/STUN). I am able to get to the point where the SIP server sends 183 session in progress to kamailio but after that I can only see - "STUN: using this candidate" "Successful STUN binding request from .." "SRTP output wanted, but no crypto suite was negotiated"
This is fairly strange:
Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port
30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated
Are you running a very old OpenSSL version by any chance?
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
-- Warm Regds. MathuRahul
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
Hi Rahul,
Don't take me wrong, but you still have some homework to do. Apache is not a requirement for webrtc ( apart hosting the website ). The only difference between using http and https is that by default on http, most browsers will always ask for the user to confirm usage of the mic/cam.
WS and WSS works with Kamailio, it's only a question of configuration ( for which there are many examples, most are broken but easily fixed, for example for https://github.com/caruizdiaz/kamailio-ws , it's only fixing the record routes to get sip2ws signaling working ).
Regarding rtp, you have to use rtpengine ( master from repo, not a release, dtls broken in latest 3.7.1, fixed in 3.8 ) or something else to be able to terminate ICE/DTLS when remote endpoints don't support them ( most of SIP ua's today unfortunately ), again, read, experiment, you'll eventually get it and the most important, know how your platform works !
Start with basic browser to browser calls, without a rtp proxy, it should work almost out of the box, then you can add some functionnality to the basic scenario, and I'll be glad to point you to the right direction !
Good luck !
Le 27/01/2015 03:21, Rahul MathuR a écrit :
Any thoughts on this gents ?
On Tue, Jan 27, 2015 at 8:09 AM, rahul.ultimate <rahul.ultimate@gmail.com mailto:rahul.ultimate@gmail.com> wrote:
Kamailio is just acting as a proxy and protocol modifier so to say. It is workin with rtpengine from sipwise to handle media as evident from he logs. This architectue uses a TURN server and the browser is chrome with latest updates. The only thing whih I haven't done is enable TLS in kamailio and create certs. (which I'm not completely sure how to do).. Also, does it necessitates to have Apache ruuning https on 443 ? Thanks in advance Sent from Samsung Mobile -------- Original message -------- From: Gonzalo Gasca Meza Date:27/01/2015 4:07 AM (GMT+05:30) To: "Kamailio (SER) - Users Mailing List" Subject: Re: [SR-Users] Need help on WebRTC with Kamailio as proxy Are you terminating media in Kamailio or just handling WS communication? If yes which version of Kamailio and rtp-proxy ? Have you tried passing media directly between Browser and Kamailio with any TURN server? Are you using latest Chrome version or FF ? A working sample config using the following architecture: https://github.com/spicyramen/llamato/tree/LlamatoReg signalling: sipml5 -- ws/wss --> Ec2 Kamailio --sip udp--> FS --sip udp--> * media: sipml5 ------------------------------------------------------------------------> * On Mon, Jan 26, 2015 at 12:44 PM, Rahul MathuR <rahul.ultimate@gmail.com <mailto:rahul.ultimate@gmail.com>> wrote: Hi Richard, Thanks for spending some cycles on it. It is OpenSSL 1.0.1e-fips 11 Feb 2013 On Tue, Jan 27, 2015 at 2:04 AM, Richard Fuchs <rfuchs@sipwise.com <mailto:rfuchs@sipwise.com>> wrote: On 26/01/15 02:21 PM, Rahul MathuR wrote: Hello, I am totally struck at a point while implementing Kamailio as proxy for WebRTC enabled UAC (Jssip). I am using Google's TURN server (rfc5766-turn-server for ICE/STUN). I am able to get to the point where the SIP server sends 183 session in progress to kamailio but after that I can only see - "STUN: using this candidate" "Successful STUN binding request from .." "SRTP output wanted, but no crypto suite was negotiated" This is fairly strange: Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated Jan 27 00:35:46 localhost rtpengine[5262]: [tsb1jrsqsadn33jjsi4f port 30794] Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated Are you running a very old OpenSSL version by any chance? cheers _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Warm Regds. MathuRahul _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users-- Warm Regds. MathuRahul
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
-
Rahul MathuR -
rahul.ultimate
-
Tristan Mahé