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

[Rahul MathuR]

Any thoughts on this gents ?



On Tue, Jan 27, 2015 at 8:09 AM, rahul.ultimate <rahul.ultimate at gmail.com>
wrote:

> Kamailio is just acting as a proxy and protocol modifier so to say. It is
> workin with rtpengine from sipwise to handle media as evident from he logs.
> This architectue uses a TURN server  and the browser  is chrome with
> latest updates.
>
> The only thing whih I haven't done is enable TLS in kamailio and create
> certs. (which I'm not completely sure how to do)..
> Also, does it necessitates to have Apache ruuning https on 443 ?
>
> Thanks in advance
>
>
> Sent from Samsung Mobile
>
>
> -------- Original message --------
> From: Gonzalo Gasca Meza
> Date:27/01/2015 4:07 AM (GMT+05:30)
> To: "Kamailio (SER) - Users Mailing List"
> Subject: Re: [SR-Users] Need help on WebRTC with Kamailio as proxy
>
> 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
>>>
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at 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 at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>


-- 
Warm Regds.
MathuRahul
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