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

[rahul.ultimate]

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 at gmail.com> 
Date:27/01/2015  4:07 AM  (GMT+05:30) 
To: "Kamailio (SER) - Users Mailing List" <sr-users at 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 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> 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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