[Freeswitch-users] Create webrtc call with custom signaling

[אריה קלטר]

Hello

It there a way to create a webrtc call with custom signaling?

I want that when freeswitch receive a sip call, send it to the browser
without registration or authentication

Freeswitch receive incoming call:
Create second leg for webrtc
Get sdp of the second leg
I will send it to the browser using custom signaling,
Get answer sdp from the browser,
Send the answer sdp to freeswitch
Bridge both legs

I saw media_webrtc=true, but i did not understand how to receive the offer
sdp and how to send the answer back to freeswitch using the api.
Can i do it using esl?

Any idea how to implement it? Is it possible?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20201105/7be1ff85/attachment.html>