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> 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:


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


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--
Warm Regds.
MathuRahul

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--
Warm Regds.
MathuRahul


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