🐋 Intro

WebRTC enables real time communication between browsers directly.

• Capturing media: grabbing your camera and microphone.
• Making a connection: setting up a secure “pipe” between them
• Sending encrypted data: ****All streams are encrypted (DTLS-SRTP).

🛠️ How WebRTC Works

1. Media Capture usinggetUserMedia().
2. Signaling (the “handshake”)
   • WebRTC needs SDP & ICE to connect, so apps use a signaling server (WebSockets or HTTP) to exchange them.
   • SDP (Session Description Protocol): describes what codecs, resolutions, and formats each side supports.
   • ICE Candidates: possible best network paths (IP addresses, ports) to connect.
3. Peer-to-Peer Connection
   • Once signaling is done, browsers try to connect directly using RTCPeerConnection.
   • They use STUN/TURN servers:
     • STUN servers help reveal your public IP so peers can attempt a direct connection.
     • TURN servers step in when firewalls/NATs are too strict, relaying traffic through a middleman.
4. Data Flow — After the connection is established, streams flow directly

🧊 ICE (Interactive Connectivity Establishment)

• A framework WebRTC uses to figure out how two devices can connect over the internet.
• Why it’s needed: Devices can be behind firewalls, routers, or NATs (Network Address Translators), so their “real” address isn’t always obvious.
• How it works:
  1. Each browser gathers possible connection points (called ICE candidates) — these might be local IPs, public IPs, or relay addresses.
  2. The browsers exchange these candidates via signaling.
  3. They test which path actually works best (fastest, most reliable).