WebRTC (Web Real-Time Communications) is a technology which enables Web applications and sites to capture and optionally stream audio and/or video media, as well as to exchange arbitrary data between browsers without requiring an intermediary. The set of standards that comprises WebRTC makes it possible to share data and perform teleconferencing peer-to-peer, without requiring that the user install plug-ins or any other third-party software.

API Landing WebRTC WebRTC API

Advanced Guide NeedsContent WebRTC

API Experimental Interface Reference RTCIdentityAssertion WebRTC WebRTC API

API Experimental Interface Reference RTCIdentityEVent WebRTC WebRTC API

API Deprecated Method Reference WebRTC

Constructor Experimental Reference RTCDataChannelEvent WebRTC

API Experimental Interface RTCConfiguration WebRTC

API Interface Reference Référence WebRTC

While the WebRTC specification is relatively stable, not all browsers have fully implemented all of its features. In addition, some browsers still have prefixes on some or all WebRTC APIs. While you can manually code around these issues, there is an easier way. The WebRTC organization provides on GitHub the WebRTC adapter to work around compatibility issues in different browsers' WebRTC implementations. The adapter is a JavaScript shim which lets your code to be written to the specification so that it will "just work" in all browsers with WebRTC support. There's no need to conditionally use prefixed APIs or implement other workarounds.

API Audio Guide Video WebRTC

API Guide Media Tutorial WebRTC

The RTCDataChannel interface is a feature of the WebRTC API which lets you open a channel between two peers over which you may send and receive arbitrary data. The API is intentionally similar to the WebSocket API, so that the same programming model can be used for each.

API Communication Example Tutorial WebRTC

Advanced Draft Guide NeedsUpdate WebRTC

API Experimental Interface Media MediaDevicesInfo WebRTC

API Experimental MediaDevices Method Reference WebRTC

API Experimental Expérimental Interface Reference Référence WebRTC

Constructor Experimental Expérimental MediaStreamEvent Reference Référence WebRTC

The read-only property MediaStreamTrack.id returns a DOMString containing a unique identifier (GUID) for the track; it is generated by the browser.

MediaStreamTrack Property Read-only Reference Référence WebRTC

The read-only property MediaStreamTrack.kind returns a DOMString set to "audio" if the track is an audio track and to "video", if it is a video track. It doesn't change if the track is deassociated from its source.

MediaStreamTrack Property Read-only Reference WebRTC

The read-only property MediaStreamTrack.label returns a DOMString containing a user agent-assigned label that identifies the track source, as in "internal microphone". The string may be left empty and is empty as long as no source has been connected. When the track is deassociated from its source, the label is not changed.

MediaStreamTrack Property Read-only Reference Référence WebRTC

The MediaStreamTrack.onmute event handler is a property called when the mute event is received. Such an event is sent when the track is temporarily not able to send data.

Event Handler MediaStreamTrack Property Reference Référence WebRTC