The code base of OBS Studio, a package for streaming, compositing and recording video, has been changed to support WebRTC technology, which can be used instead of the RTMP protocol for streaming video without an intermediate server, in which P2P content is directly transmitted to the user's browser.
The implementation of WebRTC is based on the use of the libdatachannel library written in C++. In its current form, only broadcasting (video output) in WebRTC is supported, and a service is provided with support for the WHIP process used to establish sessions between the WebRTC server and client. The code to support WebRTC as a source is currently under review.
WebRTC allows you to achieve a reduction in video delivery delays to fractions of a second, which makes it possible to create interactive content and interact with viewers in real time, for example, arrange a talk show. Using WebRTC, you can switch between networks without interrupting the broadcast (for example, switch from Wi-Fi to a mobile network) and organize the transmission of several video streams within a single session, for example, to shoot from different angles or organize interactive videos.
WebRTC also allows you to download several versions of already transcoded streams with different quality levels for users with different bandwidth of communication channels, so as not to do the transcoding work on the server side. It is possible to use different video codecs such as H.265 and AV1 to reduce bandwidth requirements. As a reference server implementation for WebRTC-based broadcasts, it is proposed to use the Broadcast Box, but for broadcasting to a small audience, you can do without a server by setting it up in P2P mode.
Source: opennet.ru