TL;DR - you'll learn WebRTC in this course - how to get a video feed and setup a video chat between browsers with just JavaScript. The first section and a few later projects are front-end only, the connection section and later projects require Node.js and Socketio.
It's 2024. Whether because of the pandemic, chat bots, or cost savings, working remote is a thing. Telemedicine is a thing. Talking to people 6 time zones away is a thing. You can abstract your users from your app to Zoom because it always works, but you lose all control and tracking of the interaction.
Enter webRTC.
TL;DR - you'll learn WebRTC in this course - how to get a video feed and setup a video chat between browsers with just JavaScript. The first section and a few later projects are front-end only, the connection section and later projects require Node.js and Socketio.
It's 2024. Whether because of the pandemic, chat bots, or cost savings, working remote is a thing. Telemedicine is a thing. Talking to people 6 time zones away is a thing. You can abstract your users from your app to Zoom because it always works, but you lose all control and tracking of the interaction.
Enter webRTC.
WebRTC is one of the browsers must mind blowing apis. It allows access to the mic, camera, even the screen AND to share them across a network socket No server (mostly) or other middleman to add bandwidth, bugs, and chaos.
Along with websockets, webRTC presents the video-side of browser real-time communication, bridging one of the last gaps in both human and web-based communication. There's a good chance if you're reading this, you've heard about webRTC. Maybe even done a tutorial on it. But how far did you get? In my experience, the vast majority of the material on the web goes no farther than a quick-start, zoom clone. You don't learn how anything works, never look at the docs, and are stuck at the end wondering what to do now. Is that all webRTC can do? The remaining shred of material is waaaaay over everyone's head. The fact that the webRTC was released about the same time as the websocket API and most developers still don't know how to use it is evidence of the gap.
This course is the first step to alleviate that. It is not a quick start guide to webRTC. There are loads of those all over the Internet. You should definitely look elsewhere if you are wanting a 10 minute intro to the 3-4 things you need to know to make a basic zoom clone. On the other hand, if you are looking to really learn one of the most awesome JavaScript APIs that no one seems to know, you should stick around. Like Express and other JavaScript/Node pieces, it's getting passed over in the wave to learn just enough to get to the term "full-stack developer." My main goal is to help you figure out how to go from being a good developer to a great developer. Understanding... not just knowing a few methods... WebRTC is part of that.
I first used webRTC in 2015 for a startup similar to telemedicine. I've been following since and have been frustrated that it hasn't gotten more mainstream notice due to Apple's reluctant support, but mostly because devs don't know it. It opens the way for so many improvements to existing applications and obvious groundwork for new ones. Let's change that :) Prepare to for a detailed look at webRTC and start going in-app real-time video/screen chat.
What we cover:
Front-end only portion
getUserMedia() - getting access to the mic and camera in the browser
playing the feed in a <video />
MediaStream and MediaStreamTrack - what makes up a video feed
Constraints - getSupportedConstraints() and getCapabilities() - seeing what this browser can do
applyConstraints - changing the feed on the fly
Recording video/audio and playing it back
Capturing the Screen for screen share and recording it
Changing input/output devices in your feed
Back-End required (node, socketio)
RTCPeerConnection - the heart of webRTC
Creating an offer and an answer
Setting RTCSessionDescription
The Signaling Process
Build a signaling server with socketio
Gathering ICE Candidates
Building a React app w/redux that uses webRTC
git checkout Set-Local-Description
git checkout Add-Local-Description
git checkout Add-Socketio
git checkout Emit-Offer
git checkout Load-Existing-Offers
git checkout Create-Answer
git checkout Signal-Error-Handling
git checkout Emitting-Answer
git checkout On-Answer-Response
git checkout Add-Ice
git checkout Apply-Ice
git checkout Add-Tracks
git checkout Create-JWT
git checkout Add-React-Router
git checkout Join-Video
git checkout Starting-Components
git checkout Wire-Up-Redux
git checkout Add-Action-Buttons
git checkout GUM-Store-Stream
git checkout Create-Peer-Connection
git checkout Thinking-Functions
git checkout Abstract-Video-Audio-Buttons
git checkout Add-Video-Feed
git checkout Enable-Disable-Video
git checkout Display-Local-Video-Inputs
git checkout Set-New-Video-Device
git checkout Replace-Tracks
git checkout Abstract-DropDown
git checkout Setup-Audio-Button
git checkout Switch-Audio-Devices
git checkout Start-Mute-Audio
git checkout Organize-Offers
git checkout Create-Offer-Tele
git checkout Add-Dashboard
git checkout Auth-Pro
git checkout Socket-Refactor
git checkout Reorg-Appt-Data
git checkout Pull-Appt-Data
git checkout Listen-Offers-Tele
git checkout Join-Video-Route
git checkout Call-Info-Pro
git checkout Create-Answer-Tele
git checkout Emit-Answer-Server
git checkout Listen-For-Answer
git checkout Emit-Ice-Candidates-Server
git checkout Ice-To-Clients
git checkout Add-Ice-PC
git checkout Add-Tracks-Victory
git checkout Tele-Bug-Fix-1
git checkout HangUp-Button
git checkout Replace-Tracks-Tele
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