Build Real-Time Streaming Systems with WebRTC
WebRTC is the browser-native protocol stack that enables peer-to-peer audio, video, and data transfer without plugins — and it underpins everything from video conferencing to live collaboration tools. This track covers the full engineering surface: signaling, NAT traversal, media pipeline, data channels, and the server-side infrastructure needed to take a real-time system from prototype to production scale.
What You Will Learn
You will learn how peer connections are established and negotiated, how to capture and manipulate media streams, and how WebRTC data channels carry arbitrary real-time data. The track covers STUN and TURN server configuration for network traversal, integrating live data sources into WebRTC workflows, and building a scalable signaling server. Later courses address advanced live data architectures that extend beyond WebRTC itself, streaming security hardening, performance profiling, and operational concerns including deployment, monitoring, and troubleshooting.
The Learning Path
Twelve courses span A1 through C1. The first course introduces real-time streaming fundamentals at the foundational level. B1 and B2 courses move through peer connection setup, media streams, data channels, and live data integration, finishing with deployment and monitoring practices. The final five C1 courses — Mastering Network Traversal with STUN/TURN, Building a Scalable WebRTC Signaling Server, Advanced Live Data Architectures beyond WebRTC, Scaling Real-Time Streaming Architectures, and Security, Performance, and Optimization for Real-Time — target production-grade concerns that separate working prototypes from reliable systems.
How It Works
Each course is broken into short, hands-on lessons you complete in the built-in code editor with real-time feedback. An AI tutor is available whenever you get stuck, so you can move through signaling logic or TURN configuration at your own pace without losing momentum.
