Difference between revisions of "WebRTC"
Jump to navigation
Jump to search
| Line 8: | Line 8: | ||
* <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.[21] | > 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] | + | * <code>[[RTCDataChannel]]</code> 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. | [[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.[21]RTCDataChannelallows 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
- 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
Advertising:
