[SR-Users] Kamailio + FreeSwitch + WebRTC

[David Villasmil]

Yeah, you need to set the correct offer, i did that a while ago, but i
can't remember how i did it.

Check out https://github.com/havfo/WEBRTC-to-SIP

Hope it help.

David

On Thu, Jun 14, 2018, 22:26 Daniel Tryba <d.tryba at pocos.nl> wrote:

> On Thu, Jun 14, 2018 at 04:48:40PM -0300, Emanuel Gianico wrote:
> > From the logs I see the jssip throw this error:
> >
> > "Failed to set remote offer sdp: Called with SDP without DTLS
> fingerprint."
> >
> > I would like to avoid RTPEngine, because from what I understand,
> FreeSwitch
> > can handle the media part.
>
> IIRC I got the same error in my tries to transcode/bridge SIP over TLS
> with SRTP to just plain old SIP with RTP. I haven't put any effort in it
> to get it working. You'll need to play around with rtpengine
> offer/answer, I based my test on
>
> https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg
> but I blamed my failure on an old rtpengine :)
>
>
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