WebRTC Peer Connection

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WebRTC Peer Connection is the lower-level API for creating direct browser-to-browser connections for audio, video, and data. Specialists understand ICE, STUN/TURN, SDP offer/answer signaling, and media constraints. Used by video conferencing, screen sharing, IoT, and real-time collaboration platforms. Salary band: $130–190k mid-level. 4–6 weeks to proficiency with networking fundamentals; 2+ months from basics.

WebRTC Peer Connection이란

WebRTC Peer Connection (RTCPeerConnection API) is the core browser API for establishing direct peer-to-peer connections between browsers. It handles the complex network negotiation (ICE candidate exchange, STUN/TURN, SDP offer/answer) so developers can focus on application logic. Once connected, you can transmit audio, video, and arbitrary data with minimal latency. The connection requires a signaling mechanism (separate from WebRTC), usually a WebSocket server, to exchange SDP offers/answers and ICE candidates. After connection is negotiated, data flows peer-to-peer.

🔧 도구 & 생태계

WebRTC API (RTCPeerConnection)Signaling Server (WebSocket, Socket.io)STUN / TURN Servers (Coturn)Chrome DevTools WebRTC InternalsBrowser APIs (getUserMedia, getDisplayMedia)SDP ProtocolICE Candidate HandlingTypeScript / JavaScript

💰 지역별 연봉

지역	주니어	미들	시니어
USA	$105k	$170k	$245k
UK	$65k	$110k	$160k
EU	$70k	$120k	$175k
CANADA	$100k	$160k	$225k

🎓 자격증

WebRTC Specification (W3C)WebRTC Fundamentals Course

⚖ 비교하기

Webrtc Multiplayer

❓ 자주 묻는 질문

What's the difference between Peer Connection and Data Channel?

RTCPeerConnection is the connection; RTCDataChannel is a data transport over that connection. You can have multiple data channels per peer connection.

What are ICE candidates and why do I need them?

ICE candidates are potential connection paths (IP/port pairs). Browser collects candidates via STUN/TURN. Most direct path wins; TURN is fallback if direct P2P impossible.

Do I need a STUN/TURN server?

Yes. STUN helps discover your external IP (works 85% of time). TURN relays traffic when P2P impossible (NAT traversal). Both are essential for real-world deployments.

What's SDP and how do I handle it?

SDP (Session Description Protocol) describes media capabilities and connection info. You don't generate SDP manually; WebRTC does. You exchange SDP blobs with signaling server.

How do I test WebRTC connections?

Chrome DevTools WebRTC Internals (chrome://webrtc-internals) shows connection stats, candidates, and codec info. Use this for debugging connection issues.

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