Fundamentals
What is WebRTC and How Does It Work?
The architecture behind browser-to-browser audio, video, and data: media capture, RTCPeerConnection, the role of signaling, and the built-in encryption every connection gets for free.
Understanding STUN Servers: The Gateway to NAT Traversal
How a STUN binding request reveals your public IP and port, what srflx candidates are, and when public STUN servers are enough — and when they aren't.
Complete Guide to TURN Servers: When Direct Connections Fail
Relay candidates, UDP vs TCP vs TLS transports, time-limited credentials, and the bandwidth math you need before putting a TURN server into production.
ICE Framework Explained: The Brain Behind WebRTC Connectivity
Candidate gathering, connectivity checks, and pair selection: how ICE decides whether your call goes direct, through STUN, or over a relay.
NAT Traversal Challenges in WebRTC
Full-cone to symmetric: how NAT types differ, why symmetric NAT defeats STUN, and what CGNAT on mobile networks means for your relay budget.
WebRTC Signaling Explained: How Peers Find Each Other
SDP offers and answers, trickle ICE candidate exchange, and what a minimal WebSocket signaling server actually has to do.
Operations & Deployment
Setting Up a Coturn TURN Server: Complete Installation Guide
From apt install to production hardening: turnserver.conf walkthrough, TLS certificates, firewall rules, and the authentication options that matter.
WebRTC Security Best Practices
DTLS and SRTP are table stakes — the real work is TURN credential hygiene, signaling authentication, and understanding what mDNS candidates do and don't protect.
TURN Server Cost Optimization: Reducing Your Relay Bandwidth Bill
Relay traffic is where WebRTC infrastructure money goes. Measure your relay ratio, pick the right hosting, tune coturn quotas, and run the build-vs-buy math.
SFU vs MCU vs Mesh: Choosing the Right WebRTC Architecture
The bandwidth math for each topology, where simulcast and SVC fit, and a decision framework for rooms from two participants to two hundred.
Debugging & Compatibility
Debugging WebRTC Connections: A Practical Toolkit
A triage workflow for dead connections: webrtc-internals, reading ICE candidates, getStats(), and command-line checks for your STUN/TURN ports.
Common WebRTC Errors and How to Fix Them
Symptom, cause, fix — for ICE failures, getUserMedia rejections, missing relay candidates, SDP glare, and DTLS handshake problems.
WebRTC Browser Compatibility: What Works Where in 2026
Chrome, Firefox, Safari, and Edge compared: codec availability, API gaps, mobile quirks, and feature detection that won't break next release.
Media & Advanced Topics
WebRTC Codec Selection: Opus, VP8, VP9, H.264, and AV1 Compared
Compression efficiency vs CPU vs hardware support: how to choose codecs per use case and control the choice with setCodecPreferences().
WebRTC Data Channels: Peer-to-Peer Data Transfer Beyond Audio and Video
SCTP under the hood, reliable vs unreliable delivery modes, backpressure with bufferedAmount, and a working chunked file-transfer pattern.
WebRTC on Mobile Devices: iOS and Android in the Real World
Native SDKs vs mobile web, hardware codecs and battery budgets, CallKit and ConnectionService, and why mobile networks need TURN more often.