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wikipedia:WebRTC (2011) audio and video communication to work inside web pages by allowing direct peer-to-peer communication, eliminating the need to install plugins or download native apps

Major components of WebRTC include several JavaScript APIs:

  • getUserMedia acquires the audio and video media (e.g., by accessing a device's camera and microphone).
  • RTCPeerConnection enables audio and video communication between peers. It performs signal processing, codec handling, peer-to-peer communication, security, and bandwidth management.
  • RTCDataChannel allows bidirectional communication of arbitrary data between peers. The data is transported using SCTP over DTLS. It uses the same API as WebSockets and has very low latency.

W3C is developing ORTC (Object Real-Time Communications) for WebRTC.

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