WebRTC is a free, open project providing browsers and mobile applications with Real-Time Communications (RTC) capabilities (audio, video, and binary data streaming) via simple APIs and common set of protocols. Tags for operating environment may be helpful, e.g. [node.js] or [reactjs] or [ios] as well as specific browser, e.g. [firefox]. Questions concerning Session Description Protocol use [sdp]. Use [ortc] for Object RTC.
WebRTC offers web application developers the ability to write rich, real-time multimedia applications (think video chat) on the web, without requiring plugins, downloads, or installs. Its purpose is to help build a strong RTC platform that works across multiple web browsers, across multiple platforms.
WebRTC API support and implementations vary, widely, across browsers and operating systems. There is at least some support (as of late 2019) in up-to-date versions of most browsers. canIUse.com for WebRTC tracks the latest levels of support. Please, when asking questions about WebRTC here on Stack Overflow, mention your OS and browser by version.
- Official samples from the WebRTC project. Try using these samples on your browser and OS as you figure out how to ask questions here on Stack Overflow.
- getUserMedia is WebRTC's web browser API for capturing video and audio (seegetusermedia).
- MediaRecorder is the browser API for recording audio and video data (see mediarecorder).
- Project site: webrtc.org
- Native API: code.google.com/p/webrtc
- Introduction from HTML5 Rocks
- extensive listing of presentations/books/tutorials google doc
- Real-Time Communications on the Universal Windows Platform with WebRTC and ORTC
Resources to learn simple examples
