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