[SR-Users] Need help on WebRTC with Kamailio as proxy

Tristan Mahé t.mahe at b-and-c.net
Tue Jan 27 19:31:25 CET 2015


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 at gmail.com <mailto:rahul.ultimate at 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 at gmail.com <mailto:rahul.ultimate at 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 at sipwise.com <mailto:rfuchs at 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 at lists.sip-router.org
>             <mailto:sr-users at 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 at lists.sip-router.org
>         <mailto:sr-users at lists.sip-router.org>
>         http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
>
>
> -- 
> Warm Regds.
> MathuRahul
>
>
> _______________________________________________
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