WebRTC (Web Real-Time Communications) is a content delivery protocol.
By default we get a stream in RTMP format, transcode and then send it to the end-users via HTTP (HLS). In comparison to the default approach delivery via WebRTC produce a noticeably smaller delay (a couple of seconds) which can't be achieved by RTMP/HTTP approach.
To start streaming via WebRTC, contact us via chat or email to firstname.lastname@example.org.
For WebRTC delivery the G-Core Labs player have to be used.