[SR-Users] Kamailio + FreeSwitch + WebRTC

[Emanuel Gianico]

I don't think so... The idea is to use FS as a media server. Why I would
use RTPEngine? FS offers the same and more:

- WebRTC support
- VoiceMail Server
- Queues
- IVR
- Announcements

And so on...

The idea is to use Kamailio in front and throw all media related stuff to
FS.

There is a way to accomplish this? I searched a lot but I couldn't find
nothing about it, only about Kamailio and RTPProxy (or RTPEngine)

Best regards,
Emanuel.

El vie., 15 de jun. de 2018 09:32, Pan Christensen <
pan.christensen at phonect.no> escribió:

> Or maybe FreeSwitch is redundant if you use rtpengine…
>
>
>
> With kind regards
> *Pan B. Christensen*
> Developer
> Phonect AS
>
>
>
> *From:* sr-users <sr-users-bounces at lists.kamailio.org> * On Behalf Of *Emanuel
> Gianico
> *Sent:* fredag 15. juni 2018 13:29
> *To:* Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
> *Subject:* Re: [SR-Users] Kamailio + FreeSwitch + WebRTC
>
>
>
> I'm going to investigate Kazoo samples as Gorlichenko suggested because I
> think using RTPEngine or rtp proxy seems redundant/unnecesary to me since
> FreeSwitch fully supports WebRTC
>
>
>
> El jue., 14 de jun. de 2018 17:42, Yuriy Gorlichenko <ovoshlook at gmail.com>
> escribió:
>
> You can watch at the kazoo project examples if you want to avoid rtp proxy
>
>
>
> On Thu, Jun 14, 2018, 23:26 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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