WebRTC

WebRTC enables peer-to-peer audio, video, and data channels in browsers and native apps without proprietary plugins.

What WebRTC provides

• Capture from camera/mic (getUserMedia)
• Encrypted transports (DTLS-SRTP)
• NAT traversal via ICE with STUN/TURN servers
• Data channels for arbitrary binary/text messages

In this section

Page	Focus
Open-source WebRTC Projects	Kurento, Janus, mediasoup, etc.
Testing WebRTC	Debug connectivity and media paths
Setting up STUN and TURN servers	Relay for restrictive NATs

Architecture tip

Browser A ←→ STUN (discover reflexive addr)
         ↘ TURN (relay when P2P fails) ↙
Browser B

Most production apps need a signaling server (WebSocket/HTTP) to exchange SDP offers/answers and ICE candidates — WebRTC does not define signaling.

Related

• Web Sockets
• Software Engineering