PeerConnection

PeerConnection (RTCPeerConnection) is the core JavaScript API in WebRTC. It represents a connection between the local device and a remote peer, handling the full lifecycle of a WebRTC session including SDP negotiation, ICE NAT traversal, and DTLS/SRTP encrypted media exchange.

PeerConnection lifecycle

1. Creation: new RTCPeerConnection(config) – configuration includes STUN/TURN server URLs
2. Adding media: addTrack() or addTransceiver() to attach audio/video tracks
3. SDP negotiation: The JSEP offer/answer exchange via createOffer(), setLocalDescription(), setRemoteDescription(), createAnswer()
4. ICE connectivity: Candidates are gathered and exchanged, connectivity checks run
5. Connected: Media flows between peers via encrypted SRTP
6. Monitoring: getStats() provides real-time quality metrics
7. Closure: close() terminates the session

Key features

• Unified Plan SDP: Each media track gets its own m= line in the SDP
• Trickle ICE: ICE candidates are sent incrementally for faster connection setup
• Data Channel: Created via createDataChannel() on the PeerConnection
• BUNDLE: All media multiplexed over a single transport connection

PeerConnection is defined by the W3C WebRTC specification and implemented in all modern browsers via libWebRTC.

Additional reading

• What is WebRTC?
• WebRTC RTCPeerConnection. One to rule them all, or one per stream?