WebRTC is a set of open-source browser APIs enabling real-time audio, video, and data communication between peers without plugins, built on standardized protocols like DTLS and SRTP for encryption and security. The technology uses three main APIs—MediaStream (for accessing camera/microphone), RTCPeerConnection (for peer-to-peer streaming), and RTCDataChannel (for generic data exchange)—along with signaling protocols and NAT traversal mechanisms like STUN and TURN to establish connections across firewalls. WebRTC democratizes real-time communication by eliminating the need for downloads, plugins, or proprietary software while keeping encryption mandatory and requiring explicit user permission for camera and microphone access.
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