[SR-Users] Need help on WebRTC with Kamailio as proxy
Richard Fuchs
rfuchs at sipwise.com
Mon Jan 26 21:34:04 CET 2015
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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