Master WebRTC: Deep dive, build Video Call with SocketIO | Udemy


Master WebRTC: Deep dive, build Video Call with SocketIO | Udemy [Update 06/2025]
English | Size: 2.23 GB
Genre: eLearning

Fundamentals covered. Learn Core WebRTC APIs, ICE Candidates, TURN, media server. Make VideoCalling App with GenAI/Peer

What you’ll learn
How does WebRTC works internally
How to use Core components – getUserMedia, RTCPeerConnection and RTCDataChannel
How a peer-to-peer call is established
How ICE is used to establish connection between two users
Role of Media Server in peer-to-peer communication
Know about the security mechanism inbuilt.
Make VideoChat App with GenAI Assistant

Unlock the power of WebRTC and dive deep into the world of real-time communication! Whether you’re a developer, engineer, or technology enthusiast, this course will guide you through the intricate workings of WebRTC, showing you how to prepare demos and telling you exactly what happens under the hood. From understanding the fundamentals to breaking down the key protocols that make real-time communication possible, you’ll gain the skills to confidently implement and troubleshoot WebRTC technology in your own projects.

Prepare Demo for :

  • Getting Access of Camera and Mic from browser.
  • Video call between two tabs in browser using RTCPeerConnection.
  • Send and Receive messages between two tabs in browser using RTCDataChannel.
  • Prepare Peer-to-Peer Video Call app with SocketIO and Node.JS.
  • Video Call with GenAI Assistant.

What You Will Learn:

  • Understand the core concepts behind WebRTC technology and how real-time communication works.
  • Explore the WebRTC architecture, including peer-to-peer connectionssignaling, and ICE (Interactive Connectivity Establishment).
  • Deep dive into the protocols that power WebRTC, such as SDP (Session Description Protocol)RTP, RTCP and SRTP.
  • Discover how NAT traversal works in WebRTC using STUN and TURN servers.
  • Learn about the media path, including media encoding, packetization, and security mechanisms.
  • Learn about various Media Server architecture such as SFU, MCU which are used in WebRTC.
  • Understand which Media Server architecture is suitable for your WebRTC Application.

Who This Course Is For:

  • Web Developers looking to add real-time communication capabilities to their applications.
  • Software Engineers who want to understand the inner workings of WebRTC technology.
  • Technology Enthusiasts interested in how modern web browsers handle real-time media communication.
  • Engineers seeking career in WebRTC.

Requirements:

  • Basic understanding of networking and web development concepts HTML, CSS and JavaScript is helpful.

MOREInterested in understanding and deploying your own TURNSERVERHead on to my course on TURNSERVER. Their you will learn why NATs exist, what is STUN and TURN Protocol. Will deploy your own TURNSERVER for your application and then verify it too for its functionality.

Who this course is for:

  • New/Beginner to WebRTC.
  • Intermediate level developer in WebRTC client/server side development.
  • Preparing to switch to WebRTC domain.
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