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

[Gonzalo Gasca Meza]

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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