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

[Rahul MathuR]

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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