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

[Rahul MathuR]

Hi Tristan,

Thanks for looking into it.

I toggled advertised_address to static one, but no luck :(



On Tue, Jan 27, 2015 at 1:13 AM, Tristan Mahé <t.mahe at b-and-c.net> wrote:

>  Hi Rahul,
>
> Check your record-route, it should not be static. ( maybe use
> advertised_address and allow double record-route for rr module ).
>
> My 2 cents.
>
> Le 26/01/2015 11:21, Rahul MathuR a écrit :
>
> 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"
>
> a) What should I do to resolve this issue ?
>  b) Why is that I never get 200 OK for the INVITE ?
>
>  I am attaching the logfile & configs herewith.
>
>
>  Can somebody please help me out here...
>
>
>  --
> Warm Regds.
> MathuRahul
>
>
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-- 
Warm Regds.
MathuRahul
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