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.
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.
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.
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.