WebRTC

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).[20]
• 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]

Related terms

• Vonage Video API (formerly TokBox OpenTok)
• Session Traversal of UDP Through NAT (STUN)
• Ant Media Server

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_certificate QUIC, 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