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    Ultimate Webrtc Bootcamp For Beginner: Webrtc, Socketio & Js

    Posted By: ELK1nG
    Ultimate Webrtc Bootcamp For Beginner: Webrtc, Socketio & Js

    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

    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.