Ultimate Webrtc Bootcamp For Beginner: Webrtc, Socketio & Js
Published 12/2024
MP4 | Video: h264, 1920x1080 | Audio: AAC, 44.1 KHz
Language: English | Size: 5.39 GB | Duration: 7h 38m
Published 12/2024
MP4 | Video: h264, 1920x1080 | Audio: AAC, 44.1 KHz
Language: English | Size: 5.39 GB | Duration: 7h 38m
Learn WebRTC with Socket io in JavaScript environment to build real-time communication web app from scratch.
What you'll learn
Learn what WebRTC is, how it works, and why it’s essential for real-time communication on the web.
Get step-by-step guidance to configure your tools and start building WebRTC projects with JavaScript
Use the getUserMedia API to access video and audio streams from cameras and microphones, and manage them efficiently.
Learn how to create direct connections between users using RTCPeerConnection and understand the connection process.
Discover what signaling is, why it’s necessary, and how to implement a simple signaling server using Socket IO.
Step-by-step implementation of a functional video chat application that allows users to connect, call, and interact in real time.
Learn to resolve common WebRTC problems such as connection failures, audio/video not showing, and browser compatibility issues.
Explore how STUN/TURN servers help WebRTC handle NAT traversal and enable connections across different networks.
Learn about challenges in scaling WebRTC applications and get introduced to media servers like SFU (Selective Forwarding Unit) and MCU (Multipoint Control Unit)
Build confidence by developing projects and exploring WebRTC’s use cases, such as video conferencing, screen sharing, and real-time collaboration tools.
Requirements
Basic understanding of HTML , CSS and JavaScript is recommended but not mandatory.
No prior experience with WebRTC is needed—you will learn everything from scratch.
Description
Are you ready to take your first step into the world of real-time communication on the web? Whether you’re a beginner developer, a curious programmer, or someone interested in building video and audio chat applications, this course is the perfect starting point for you.Welcome to the Ultimate WebRTC Bootcamp for Beginners—a comprehensive, step-by-step course designed specifically for absolute beginners. In this course, you’ll unlock the potential of WebRTC (Web Real-Time Communication), an exciting technology that powers real-time video, audio, and data-sharing applications in the browser—without plugins or extra software. You will also learn how to combine WebRTC with Socket IO for signaling and JavaScript to tie it all together, giving you everything you need to create real-time applications from scratch.This course is beginner-friendly and structured to help you build a solid foundation in WebRTC concepts and hands-on implementation. You don’t need any prior knowledge of WebRTC—we will walk you through each concept, break it down into easy-to-understand steps, and help you apply what you learn through practical projects.By the end of this course, you will have built your very own basic video chat application—a fully functional project that demonstrates real-time peer-to-peer communication. You will also gain the confidence and skills to explore advanced WebRTC concepts, enabling you to build more sophisticated applications like video conferencing platforms, collaborative tools, or even low-latency communication apps.
Overview
Section 1: Introduction to WebRTC
Lecture 1 What is WebRTC?
Lecture 2 Evolution of communication Technolog
Lecture 3 The WebRTC Ecosystem Overview
Lecture 4 Understanding Webrtc Core Component
Lecture 5 Introduction to Webrtc Protocol
Lecture 6 Supported Platforms and Browsers for WebRTC
Section 2: Understanding the WebRTC API
Lecture 7 Setting Up Your WebRTC Development Environment
Lecture 8 Hands on WebRTC getUserMedia
Lecture 9 WebRTC Stop Stream
Lecture 10 Initiating and Understanding RTCPeerConnection
Lecture 11 Establishing Connection with RTCPeerConnection
Lecture 12 Introduction to Data Channels RTCDataChanne
Section 3: Working with Media Streams
Lecture 13 Hands on Audio and Video Stream
Lecture 14 Manipulating Media Streams in WebRTC
Lecture 15 Media Stream recording
Lecture 16 Screen Sharing
Section 4: Signaling and Peer Connections
Lecture 17 What is Signaling in WebRTC
Lecture 18 Implementing a Simple Signaling Server
Lecture 19 Establishing a WebRTC Peer Connection
Section 5: Building Your First WebRTC Application
Lecture 20 Initate the Project
Lecture 21 Creating Server
Lecture 22 Register User at Server
Lecture 23 Call User Functionality
Lecture 24 Completing Signaling Process
Lecture 25 Reject Call Funcationality
Section 6: Ways to Debug a WebRTC Application
Lecture 26 Issue 1: Connection Fails to Establish
Lecture 27 Issue 2: Video or Audio Not Showing
Lecture 28 Issue 3: High Latency and Poor Video Quality
Lecture 29 Issue 4 ICE Candidate Gathering Timeout
Lecture 30 Issue 5: Browser Compatibility Issues
Lecture 31 Issue 6: Handling Connection Drops and Reconnects
Lecture 32 Tools for Debugging WebRTC Applications
Section 7: Advanced WebRTC Concepts
Lecture 33 Introduction to STUN and TURN Servers
Lecture 34 Setting Up a STUN and TURN Server
Lecture 35 Understanding NAT Traversal and Dealing with Firewall Issues
Lecture 36 Working with ICE Candidates in WebRTC
Section 8: Scaling and Performance Optimization
Lecture 37 Challenges in Scaling WebRTC Applications
Lecture 38 Introduction to Media Servers SFU MCU
Lecture 39 Best Practices for Performance Optimization
Section 9: WebRTC Security and Privacy
Lecture 40 WebRTC Security Layers From IP to Application
Lecture 41 Data Channels Security with DTLS and SCTP
Lecture 42 Media Streams Security with SRTP and RTP
Lecture 43 Privacy and Anonymity After Managing IP Exposure and Device Access
Section 10: Build Complete Webrtc App with Signaling
Lecture 44 Interface overview
Lecture 45 Register user
Lecture 46 Call User Functionality
Lecture 47 Call Action Button
Lecture 48 Create Chat System
Lecture 49 Adding Screen Sharing system
Lecture 50 Recording System
Lecture 51 Camera Selection
Lecture 52 Logout
Section 11: Real-World Use cases of WebRTC
Lecture 53 WebRTC in Video Conferencing Applications
Lecture 54 WebRTC for Real-Time Collaboration Tools
Lecture 55 WebRTC for Gaming and Low-Latency Applications
Lecture 56 WebRTC in IoT and Smart Devices
Section 12: WebRTC Ecosystem and Future Developments
Lecture 57 WebRTC API Updates and New Features
Lecture 58 How WebRTC is Evolving for New Use Cases
Section 13: Conclusion
Lecture 59 Congratulation
Beginners interested in real-time communication technologies.,JavaScript developers looking to expand their skill set with WebRTC.,Anyone aspiring to build real-time applications like video chat app.,Students or developers curious about integrating Socket io with WebRTC.