Design

Goal

The idea is to use OBS great capturer and input mixers instead of WebRTC capturers, but then use webrtc for the media stack (encoder, RTP packetizer, RTCP controller), security (DTLS-SRTP encryption), NAT traversal (ICE Transport), and Network Transport.

Implementation

In principle, WebRTC is closer to RTMP than to HLS or MPEG-DASH, in the sense that it is not file-based, and not HTTP-based. It made sense to re-use the rtmp-service and obs-outputs core plugins to implement WebRTC support.

However, unlike RTMP or any other protocol before, WebRTC 1.0 does not mandate any signaling protocol, or signaling transport protocol, even if the handshake itself is standardized (JSEP). We use WHIP for that.