WebRTCenabled

WebRTC-enabled refers to systems, software, or devices that support Web Real-Time Communication (WebRTC), an open standard that enables real-time audio, video, and data exchange directly between web browsers and devices without the need for plugins. Implementations rely on browser APIs such as getUserMedia for media capture, RTCPeerConnection for media negotiation, and RTCDataChannel for peer-to-peer data. WebRTC operates in secure contexts and typically uses signaling to establish connections, while the media path traverses network infrastructures via NATs using ICE, STUN, and TURN servers.

To establish a call, a WebRTC-enabled application collects user media, creates a peer connection, and exchanges session descriptions (SDP) and ICE candidates through a signaling server. Once negotiated, audio, video, and optionally data channels flow directly between peers or via media servers when relay is needed. Codecs like VP8, VP9, H.264 for video, and Opus for audio are negotiated during the offer/answer process. Security is provided by DTLS-SRTP encryption for both media and data channels.

Applications include browser-based video conferencing, real-time collaboration tools, live broadcasting, and peer-to-peer file transfer. WebRTC-enabled systems are supported by major browsers such as Chrome, Firefox, Edge, and Safari, though capabilities vary by platform and version. Real-world deployments often rely on third-party signaling servers and media relays to handle connectivity in restrictive networks or to scale group conversations.

Security and privacy considerations include mandatory encryption, user consent for device access, and controls over data channels. WebRTC’s security model emphasizes end-to-end encryption for media, with signaling and data channel access managed by the application.