Hello Richard,

Now, after upgrading to - OpenSSL 1.0.1j 15 Oct 2014
Errors like - 
"Failed to set up SRTP after DTLS negotiation: no SRTP protection profile negotiated" are not seen but still the call is not getting through.

Please let me know how to proceed..

Thanks in advance


On Tue, Jan 27, 2015 at 2:14 AM, 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



--
Warm Regds.
MathuRahul