WebRTC
Broadcast audio, and video material, as well as transmit arbitrary data between browsers without the need for a middleman
Broadcast audio, and video material, as well as transmit arbitrary data between browsers without the need for a middleman
(Web Real-Time Communication) is a technology that allows Web apps and sites to record and potentially broadcast audio and/or video material, as well as transmit arbitrary data between browsers without the need for a middleman.
To access this setting, go to Administration > Workspace > Settings > WebRTC.
Enable for Public Channels: WebRTC will be enabled for all public channels if set to true.
Enable for Private Channels: When enabled, private channels will have WebRTC.
Enable for Direct Messages: If set to true, direct messages will have WebRTC.
STUN/TURN Servers: A list of STUN and TURN servers separated by a comma.
Username, password, and port are allowed in the format username:password@stun:host:port or username:password@turn:host:port.