RTCPeerConnection

RTCPeerConnection is a core WebRTC API that represents a direct, peer-to-peer connection between the local user agent and a remote peer. It is responsible for negotiating, establishing, and maintaining the media and data exchange between two endpoints. The object lives in the browser and is used to set up real-time communication for audio, video, and arbitrary data.

Key responsibilities include handling the signaling and connection negotiations required to establish a media path. The signaling protocol is not defined by the API; developers must implement their own signaling channel to exchange session descriptions and network information. The connection uses ICE (Interactive Connectivity Establishment) to traverse NATs and firewalls, gathering local candidates and exchanging them with the remote party. STUN and TURN servers may be used to discover public addresses and relay media when direct paths are not available.

Security is integral to RTCPeerConnection. All media and data channels are encrypted by DTLS and SRTP, providing confidentiality and integrity. The data channel, created with RTCDataChannel, enables bidirectional arbitrary data transfer alongside media streams.

Typical lifecycle entails creating the connection with a configuration object that may specify iceServers, adding local media tracks or transceivers, and performing a signaling exchange: createOffer, setLocalDescription, send the offer to the remote peer, receive and set the remote description, and then create and exchange an answer. ICE candidates are gathered and exchanged via onicecandidate and addIceCandidate. Events such as ontrack, ondatachannel, and oniceconnectionstatechange inform about media arrival, data channel setup, and connection status. The RTCPeerConnection can be closed to terminate the session.