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@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@lists.kamailio.org> On Behalf Of Emanuel Gianico
Sent: fredag 15. juni 2018 13:29
To: Kamailio (SER) - Users Mailing List <sr-users@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@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@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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