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!
Quick Start
Integrate the SDK and make your first call
Run Demo
Download the sample project and run locally
API Reference
Methods, parameters, and type definitions
Reference
ChangelogRelease notes and version history
Upgrade GuideNotes for upgrading between versions
Browser Support
Website
Latest Version v5.16.0
v5.16.0 - 2026.03.13
- trtc.updateLocalAudio now supports `mute: 'microphone'` to mute only the microphone while continuing to send mixed audio, and the `muteKeepVolumeDetection` parameter to continue detecting volume while muted.
- VideoMixer plugin now supports using the SDK's internal camera and screen share as input sources.
- trtc.startScreenShare added the `selfBrowserSurface` parameter to control whether the current tab is included in the screen share picker.
- Added the experimental `preconnect` API to support pre-connection for optimizing room entry time. To use this feature, please contact Technical Support. Refer to callExperimentalAPI.
- Optimized default buffer configuration for audience role, better balancing smoothness and latency. For ultra-low latency, please submit a ticket to contact us.
- Avoided occasional encoding failure on iOS 26.2.
- Fixed an issue where `stopLocalAudio` could remain pending during phone call scenarios on iOS.
- Fixed autoplay recovery failure on WeChat H5 for iOS 26.
- Fixed an issue where audio continued playing after stopping the AudioMixer plugin, causing audio duplication.