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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.
Learn more
- WebRTC on Wikipedia
- Guide to WebRTC on MDN
- Browser support of WebRTC