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WebRTC Explained: Deep dive, build Projects using WebRTC

Name: WebRTC Explained: Deep dive, build Projects using WebRTC
Rating: 4.1 (13 reviews)

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

Role Play

Created byVineet Srivastava

Last updated 10/2025

English

English [Auto],

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

Course content

9 sections • 48 lectures • 4h 42m total length

Introduction2:17
Discover how WebRTC enables encrypted, peer-to-peer video, audio, and data communication directly in browsers, with interoperable APIs for video conferencing and live streaming.
Why Study WebRTC2:29
Course Contents3:52
Explore how WebRTC enables real-time video chat by breaking down four sections: core components and IETF, W3C standard APIs, ICE connectivity across firewalls, and media transfer protocols.

Peer to peer call flow10:17
Visualize the peer-to-peer audio-video WebRTC call flow, where A and B register with a signaling server, negotiate media via SDP, and use ICE with TURN to traverse firewalls.
Protocol Stack3:49
Explore how WebRTC relies on a multi-layer protocol stack, including signaling options, ice with stun and turn, dtls and srtp encryption, and tcp/tls for secure, reliable peer-to-peer video and audio.
ICE9:34
Signalling Plane18:09
Signalling and ICE
Media Plane8:02
Security and Encryption2:55
Media Plane and Security

Introduction3:00
Explore how webRTC enables real-time audio, video conferencing, and peer-to-peer data exchange through standard web APIs like getUserMedia, RTCPeerConnection, and data channels.
getUserMedia12:31
Request audio and video streams with the getUserMedia API to capture mic and camera input, producing a media stream with synchronized tracks for local display and peer transmission.
RTCPeerConnection13:48
Code used in this lecture can be downloaded from GitHub.
https://github.com/vineetdev/WebRTC-Samples
RTCPeerConnection with Signalling6:51
RTCDataChannel5:42
Explore RTCDataChannel API to build peer-to-peer data channels for text and non-audio data with reliable or partially reliable delivery. Learn signaling and data channel open state from the GitHub sample.

Introduction7:37
ICE Candidates2:54
Ice candidates in WebRTC include host, server reflexive, relay, and Pir reflexive addresses, showing how turn servers reveal transport addresses for peer-to-peer connections.
STUN - What and How5:24
Discover how the Stun protocol enables nat traversal in WebRTC by revealing the public IP and port through binding requests, keeping the binding alive with periodic messages.
STUN and ICE Canddidates
TURN - What and How7:27
ICE Connectivity Checks4:32
Trickle Ice4:29
TURN, ICE Connectivity Checks and Trickle Ice
Understand the role of TURN in MEDIA transfer of Call using WebRTC.

Project Demo1:31
Watch a WebRTC peer app demo, start the signaling server, join with two tabs in the same room, view local and remote video, and mute or end the call.
Project Overview2:16
Prepare Landing Webpage8:28
Prepare Call webpage6:37
Apply Styles to Webpages3:25
Implement Signalling Server10:19
Create Peer Connection8:01
Signalling and Handling OfferSDP9:18
Signalling and Handling ICE Candidates4:25
Implement End Call Functionality4:05
Fixing OfferSDP Creation8:03
Signalling And Handling AnswerSDP3:43
Log and handle answer SDP in signaling for a WebRTC real-time video call, including logging offer, logging ICE, emitting and sending answer SDP, and setting remote description.
Fixing Remote Video1:25
Add Accessories And Final Demo1:33

Media Transport in SDP5:28
Learn how SDP enables media transport for WebRTC by using RTP over DTLS with UDP, defining audio and video payload types, and explaining Opus, DTMF, VP8, RTX, and RTP multiplexing.
Real-Time Transport Protocol RTP12:24
Explore the RTP protocol, its packet structure, and implementation for real-time audio and video delivery over UDP, including payload types, sequence numbers, timestamps, and jitter buffering.
Media Transport
Real-Time Transport Control Protocol RTCP10:13
SRTP3:03
SCTP in DataChannel3:05
RTCP, SRTP and DataChannel

Requirements

Basic level of HTML CSS and JavaScript.

Description

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 connections, signaling, 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.

MORE: Interested in understanding and deploying your own TURNSERVER. Head 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.

WebRTC Explained: Deep dive, build Projects using WebRTC

What you'll learn

Explore related topics

Course content

Introduction3 lectures • 9min

WebRTC Explained6 lectures • 53min

Core WebRTC APIs5 lectures • 42min

Understanding ICE7 lectures • 32min

Project: Peer to Peer Video Call using Node JS and Socket IO14 lectures • 1hr 13min

Media Transport Explained5 lectures • 34min

Project: Video Chat with Gen AI Assistant4 lectures • 15min

Media Servers In WebRTC3 lectures • 22min

Whats Next2 lectures • 3min

Requirements

Description

Who this course is for: