Difference between revisions of "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. |
[[W3C]] is developing [[ORTC]] (Object Real-Time Communications) for WebRTC. | [[W3C]] is developing [[ORTC]] (Object Real-Time Communications) for WebRTC. | ||
Revision as of 13:54, 16 February 2023
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:
getUserMediaacquires the audio and video media (e.g., by accessing a device's camera and microphone).RTCPeerConnectionenables audio and video communication between peers. It performs signal processing, codec handling, peer-to-peer communication, security, and bandwidth management.RTCDataChannelallows 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.
Related terms
- Vonage Video API (formerly TokBox OpenTok)
- Session Traversal of UDP Through NAT (STUN)
- Ant Media Server
- PCMA/PCMU
See also
- WebRTC, Ant Media Server, DTLS, SCTP,
webrtc-cli, coturn - HTTP, HTTP client, HTTP/1.1, HTTP/2, HTTP/3, HTTPS, HSTS CSR, TLS, SSL,
openSSL, WebSockets, WebRTC,ssl_certificateQUIC, HPKP, CT, List of HTTP status codes, URL redirection, Content-type:, Webhook, HTTP headers,--insecure, Axios HTTP client, HTTP cookies, HTTP ETag, Hypertext Transfer Protocol -- HTTP/1.1
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