WebRTCfeatures

WebRTC features refer to the capabilities provided by Web Real-Time Communication technologies that enable real-time audio, video, and data communication in web browsers and apps without plugins. The core components are designed to work together to establish and maintain media paths between peers.

Key APIs and capabilities include getUserMedia for capturing local media such as microphone and camera, getDisplayMedia for screen sharing, RTCPeerConnection for negotiating and streaming media across networks, and RTCDataChannel for peer-to-peer data transfer. Media streams are managed with the MediaStream API, allowing tracks to be added or removed during a session.

Connectivity and security are central to WebRTC. The framework uses ICE (with STUN and TURN servers) to traverse NATs and firewalls, establishing the best path between peers. Media and data channels are secured by default using DTLS-SRTP, providing encryption for both audio/video and data transmissions. WebRTC relies on signaling to exchange session descriptions and network candidates, but signaling mechanisms are outside the standard and implemented by applications.

Common features include real-time low-latency audio and video, adaptive bitrate and codec negotiation (support for VP8/VP9, H.264, and audio codecs like Opus), congestion control, and support for simulcast or scalable video in some deployments. Data channels enable text chat, file transfer, and game state synchronization, with configurable reliability and ordering.

Platform support varies by browser and device, but modern browsers generally provide broad WebRTC support. Applications typically implement their own signaling layer and user interface while leveraging WebRTC’s standard media and data pathways.