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