https://github.com/2600hz/kazoo-configs-kamailio/blob/master/kamailio/websoc...
2018-06-21 18:35 GMT+03:00 Emanuel Gianico emanuelgianico@gmail.com:
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/kamai lio/kamailio.cfg but I blamed my failure on an old rtpengine :)
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