WebRTC Core Technology Development: Building Real-Time Audio, Video & Data Applications for the Web
“WebRTC Core Technology Development” is an advanced, project-based course designed to teach you how to build real-time communication systems using WebRTC — the powerful open-source technology behind modern peer-to-peer voice, video, and data sharing in the browser.
Whether you're building video conferencing platforms, multiplayer games, screen-sharing tools, IoT dashboards, or collaborative web apps, WebRTC (Web Real-Time Communication) is the key to making it all happen without plugins or third-party installations.
This course demystifies WebRTC from the ground up. You’ll start with the fundamentals of how peer-to-peer communication works, including ICE (Interactive Connectivity Establishment), STUN, and TURN servers. Then, you'll explore core APIs such as getUserMedia, RTCPeerConnection, and RTCDataChannel, with real-world coding exercises at every step.
You’ll build practical, full-stack WebRTC applications — including:
- One-on-one video chat
- Group conferencing (with SFU/MCU architecture)
- Real-time screen sharing and whiteboarding
- Peer-to-peer file transfer and messaging systems
- Low-latency data sync across distributed clients
We’ll also dive deep into signaling servers, media negotiation, NAT traversal, and bandwidth optimization. You'll learn how to set up scalable back-end infrastructure using Node.js, WebSocket, or WebSocket alternatives, and how to implement secure communications with TLS, DTLS, and SRTP.
In the later modules, you'll explore:
- Media recording and streaming
- Adaptive bitrate streaming and fallback mechanisms
- Integration with third-party platforms like Janus, Jitsi, and mediasoup
- Mobile support and browser compatibility issues
- Performance debugging and real-time analytics
The course is structured to give you production-ready patterns and tools so you can confidently build and deploy scalable, secure, and responsive real-time systems.
By the end of this course, you will be able to:
- Understand and configure all the moving parts of a WebRTC system.
- Design your own signaling logic and peer discovery flow.
- Use advanced WebRTC features such as simulcast, SFU routing, and network adaptation.
- Integrate WebRTC into full-stack applications for web, desktop, and mobile use.
- Apply optimization techniques for low-latency, high-quality communication.
This course is perfect for:
- Full-stack developers building communication apps
- Engineers transitioning from native video stacks to browser-based RTC
- Startups creating Zoom, Discord, or Google Meet-style platforms
- Anyone who wants to harness real-time capabilities for web-based innovation
All examples are written in modern JavaScript (ES6+), and supporting frameworks like Node.js and Express are used where needed. You don’t need prior experience with WebRTC, but familiarity with web development and networking concepts will help you move faster.
Get ready to dive into the real-time web — one stream, one peer, and one packet at a time.
Course Outline:
01-Introduction to The Basic Concepts of WebRTC
02-WebRTC Working Principle and Basic APIs
03-WebRTC Quick Start Establishing Real-Time Communication Flow
04-WebRTC JavaScript API Usage and Introduction
05-WebRTC Advanced Media Processing
06-WebRTC Signaling Service and Integration
07-WebRTC Error Handling and Debugging
08-WebRTC Security and Privacy
09-WebRTC One-to-One Real-Time Communication Implementation
10-WebRTC Implementation for One-to-Many and Many-to-Many Real-Time Communication
11-WebRTC Advanced Network Transmission
12-WebRTC Large-Scale Deployment and Management
13-WebRTC Cross-Platform Development
14-WebRTC Advanced Features and Optimization
15-WebRTC Advanced API and Extensions
16-WebRTC Source Code Architecture
17-Classic Module Source Code Analysis
18-Frontier Technology and Source Code Analysis