TRTC Web SDK
Introduction
TRTC Web SDK is the Web SDK of Tencent Cloud's real-time audio and video communication solution. It is a JavaScript library loaded through HTML web pages. Developers can use the APIs provided by TRTC Web SDK to establish connections, control real-time audio and video calls or live streaming services.
Currently, TRTC Web SDK mainly supports Chrome M58+ and Safari browsers.
Please be sure to use HTTPS protocol or localhost to deploy your Web App, otherwise the error of navigator.mediaDevices not found will occur!
Basic Working Principle
TRTC Web SDK is a real-time audio and video call solution based on WebRTC. It establishes a signaling channel with Tencent Cloud's real-time audio and video backend server through WebSocket. The signaling channel is mainly used for signaling communication such as user entering and exiting the room, media capability exchange, and room user status change notification; audio and video data transmission is performed through WebRTC's RTP/RTCP.
API Overview
- TRTC is the main entry point of the entire SDK, providing functions such as creating TRTC objects TRTC.create, obtaining camera lists, microphone lists, checking browser compatibility, controlling log level and log upload.
- trtc object created by TRTC.create, providing the core capabilities of real-time audio and video calls:
- Enter room enterRoom()
- Exit room exitRoom()
- Publish/preview local video startLocalVideo()
- Publish/preview local audio startLocalAudio()
- Unpublish/unpreview local video stopLocalVideo()
- Unpublish/unpreview local audio stopLocalAudio()
- Watch remote video startRemoteVideo()
- Stop watching video stopRemoteVideo()
- Mute/unmute remote audio muteRemoteAudio()
