WebRTC

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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.[21]
  • RTCDataChannel allows bidirectional communication of arbitrary data between peers. The data is transported using SCTP over DTLS.[22] It uses the same API as WebSockets and has very low latency.[23]

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

Related terms

See also

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