Changes have been made to the OBS Studio codebase, a package for streaming, compositing, and video recording, implementing support for WebRTC technology, which can be used instead of the RTMP protocol for streaming video without an intermediary server, where content is transmitted directly to the user's browser in P2P mode.
The implementation of WebRTC is based on the use of the libdatachannel library, written in C++. Currently, only broadcasting (video output) in WebRTC is supported, and a service is provided that supports the WHIP process used for establishing sessions between the server and the WebRTC client. The code to support WebRTC as a source is still under review.
WebRTC allows for a reduction in video delivery latency to a fraction of a second, enabling the creation of interactive content and real-time engagement with viewers, such as hosting talk shows. With WebRTC, it is possible to switch between networks without interrupting the broadcast (for example, transitioning from Wi-Fi to a mobile network) and to organize the transmission of multiple video streams within a single session, such as filming from different angles or organizing interactive videos.
WebRTC also allows uploading multiple pre-encoded stream variants with different quality levels for users with varying bandwidth, so that encoding work does not need to be done on the user's side. serverVarious video codecs can be used, such as H.265 and AV1, to reduce bandwidth requirements. As a reference implementation of a server for WebRTC broadcasts, it is suggested to use Broadcast Box, but for broadcasting to a small audience, it is possible to manage without a server, setting up P2P mode.
Source: opennet.ru
