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WebRTC (Web Real-Time Communication) is an API that can be used by video-chat, voice-calling, and P2P-file-sharing Web apps.

WebRTC consists mainly of these parts:

getUserMedia
Grants access to a device's camera and/or microphone, and can plug in their signals to a RTC connection.
RTCPeerConnection
An interface to configure video chat or voice calls.
RTCDataChannel
Provides a method to set up a peer-to-peer data pathway between browsers.

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 Contributors to this page: Tigt, marie-ototoi, Andrew_Pfeiffer, Jeremie, teoli, Sebastianz, klez, ajinkya_p
 Last updated by: Tigt,