WebRTC

E3760

open web standard real-time communication protocol suite web communication technology

WebRTC is an open web technology that enables real-time audio, video, and data communication directly between browsers and devices without requiring plugins.

Aliases (12)

MediaStream API ×1
MediaStreamTrack API ×1
RTCDataChannel API ×1
RTCPeerConnection API ×1
RTCRtpSender ×1
W3C WebRTC specification ×1
Web Real-Time Communication ×1
WebRTC 1.0 ×1
WebRTC 1.0 specification ×1
WebRTC 1.0: Real-Time Communication Between Browsers ×1
WebRTC API ×1
getUserMedia API ×1

Statements (74)

Predicate	Object
instanceOf	open web standard → real-time communication protocol suite → web communication technology →
category	real-time communication → web technology →
commonlyUsedIn	IoT and embedded communication devices → customer support and contact center applications → gaming and interactive applications → online education platforms → video conferencing services →
designedFor	interoperability between browsers → low-latency communication → real-time multimedia applications →
developedBy	IETF RTCWEB Working Group → W3C WebRTC Working Group →
doesNotSpecify	signaling protocol →
enables	file transfer via data channels → low-latency streaming → plugin-free real-time communication in browsers → video conferencing in web applications → voice calling in web applications →
firstAnnounced	2011 →
fullName	Web Real-Time Communication →
hasComponent	MediaStream API → MediaStreamTrack API → RTCDataChannel API → RTCIceCandidate → RTCPeerConnection API → RTCRtpReceiver → RTCRtpSender → RTCSessionDescription → getUserMedia API →
implementedIn	Android browsers → Apple Safari → Google Chrome → Microsoft Edge → Mozilla Firefox → iOS Safari →
license	various open-source licenses for reference implementations →
openSource	true →
originatedFrom	Google →
relatedStandard	ORTC → RTP → SIP →
requires	secure context (HTTPS) in modern browsers → signaling mechanism external to the core specification →
securityFeature	mandatory encryption of media and data → user consent for camera and microphone access →
standardizedBy	Internet Engineering Task Force → World Wide Web Consortium →
supports	DTLS → ICE → NAT traversal → RTP → SCTP → SRTP → STUN → TURN → browser-to-browser communication → browser-to-native application communication → data channels → end-to-end encryption at the transport level → media streams → peer-to-peer communication → real-time audio communication → real-time data communication → real-time video communication → screen sharing →
uses	DTLS for key negotiation and encryption → ICE for connectivity checks → JavaScript APIs in web browsers → SCTP over DTLS for data channels → SDP for session description → SRTP for secure media transport →

Referenced by (28)

Subject (surface form when different)	Predicate
WebRTC ("getUserMedia API") → WebRTC ("RTCPeerConnection API") → WebRTC ("RTCDataChannel API") → WebRTC ("MediaStream API") → WebRTC ("MediaStreamTrack API") → WebRTC ("RTCRtpSender") →	hasComponent
Blink → Microsoft Edge → Mozilla Firefox → Safari →	supports
Apple Safari → Chromium → Opera →	supportsStandard
RTCRtpReceiver ("WebRTC 1.0 specification") → RTCRtpReceiver ("W3C WebRTC specification") →	definedIn
RTCIceCandidate → RTCIceCandidate ("WebRTC 1.0: Real-Time Communication Between Browsers") →	definedInSpecification
RTCIceCandidate ("WebRTC API") → RTCRtpReceiver →	partOf
Nearby Share →	communicationProtocol
World Wide Web Consortium →	developsStandard
IETF RTCWEB Working Group →	field
WebRTC ("Web Real-Time Communication") →	fullName
ORTC ("WebRTC 1.0") →	relatedConcept
ORTC →	relatedStandard
Google Duo →	supportsProtocol
TURN →	usedFor
Secure Real-time Transport Protocol →	usedIn