Editing WebRTC
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* <code>[[RTCPeerConnection]]</code | * <code>[[RTCPeerConnection]]</code | ||
> enables audio and video communication between peers. It performs signal processing, codec handling, peer-to-peer communication, security, and bandwidth management. | > enables audio and video communication between peers. It performs signal processing, codec handling, peer-to-peer communication, security, and bandwidth management. | ||
| − | * <code>[[RTCDataChannel]]</code> 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. | + | * <code>[[RTCDataChannel]]</code> 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.[23] |
[[W3C]] is developing [[ORTC]] (Object Real-Time Communications) for WebRTC. | [[W3C]] is developing [[ORTC]] (Object Real-Time Communications) for WebRTC. | ||
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