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SRTC is a converged audio and video communication component. Built on low-latency audio and video and real-time multi-user interaction, it combines standard audio/video protocol interoperability, MCU (server-side mixing and recording), self-hosted live streaming, and more. It can be deployed on public cloud, private cloud, or hybrid cloud, giving developers a low-cost solution for interactive audio and video. If you are still deciding between SRTC and SMeeting (the SDK behind STMLink Meeting), see Choosing SRTC or SMeeting first.

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Architecture

The real-time audio and video component focuses on cross-platform multi-user audio and video calls and low-latency interaction. It provides SDKs for WeChat Mini Program, Web, Android, iOS, Windows, Linux, and other platforms, so you can integrate quickly and connect to third-party private cloud backends. Combined with our other products, it also supports capabilities such as instant messaging and whiteboards, extending to more business scenarios. SRTC architecture SRTC only handles the audio and video transport itself. It has no user system and no business rules—your backend owns both and works with SRTC through the server API.

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Platform support

All platforms interoperate, and behavior is aligned across them.
Each platform is distributed differently: Web via npm, Android via Maven, and Windows as a zip downloaded from the artifact repository. See the corresponding integration page for each. Swift uses Swift Package Manager (prebuilt XCFramework), and Python uses PyPI (pip install srtc). Contact us for the iOS and C SDK packages.