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    Mastering Webrtc Part 2 - Real-Time Video And Screen-Share

    Posted By: ELK1nG
    Mastering Webrtc Part 2 - Real-Time Video And Screen-Share

    Mastering Webrtc Part 2 - Real-Time Video And Screen-Share
    Published 7/2023
    MP4 | Video: h264, 1280x720 | Audio: AAC, 44.1 KHz
    Language: English | Size: 4.59 GB | Duration: 8h 39m

    webRTC, Part 2 - connecting the browsers - PeerConnection & Signaling, w/Reach

    What you'll learn

    Connect two or more computers audio/video streams

    Set up a signaling server to act as a middle man to negotiate the connection

    How to integrate webRTC into a React App w/redux

    Code organization to minimize confusion as your app grows

    Manage 4 (yes 4) different async sources to create an incredible web app

    Requirements

    Need to be familiar with getUserMedia() or have taken part 1

    Experience with JavaScript and Node (not a ninja, but decent exposure)

    Socketio is used to a lesser degree in second 1, and very heavily in second 2

    For later sections: Access to a Linux machine (like AWS, Azure, DigitalOcean, etc)

    Description

    This course is part 2  covering the second part of the webRTC api: connecting to another browser and everything that goes into it (namely signaling!). I won't assume you know a single thing about RTCPeerConnection of signaling.It's 2023. 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 DIRECTLY to another browser. 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:RTCPeerConnection - the heart of webRTCCreating an offer and an answerSetting RTCSessionDescriptionThe Signaling ProcessBuild a signaling server with socketioGathering ICE CandidatesBuilding a React app w/redux that uses webRTC

    Overview

    Section 1: Introduction

    Lecture 1 Introduction

    Lecture 2 All my code (github link)

    Lecture 3 How I code

    Section 2: rtcPeerConnection - Stream video, peer to peer

    Lecture 4 Section Demo & Overview

    Lecture 5 rtcPeerConnection and signaling

    Lecture 6 Signaling Part 1 - SDP

    Lecture 7 Signaling Part 2 - ICE (and STUN)

    Lecture 8 File Structure

    Lecture 9 local RTC

    Lecture 10 setLocalDescription()

    Lecture 11 Socket.io and HTTPS setup

    Lecture 12 Connection TaskList

    Lecture 13 Connection TaskList - part 2

    Lecture 14 emit newOffer

    Lecture 15 Emit iceCandidates

    Lecture 16 Load existing and new offers on all clients

    Lecture 17 Create answer

    Lecture 18 Error handling the signaling process

    Lecture 19 Emitting answer

    Lecture 20 Listening for answer and setRemoteDescription(answer)

    Lecture 21 Apply ICE candidates - Part 1

    Lecture 22 Apply ICE candidates - Part 2

    Lecture 23 Add tracks from remote peer - MAGIC!!

    Lecture 24 Loading on another device on the same network

    Section 3: webrtc and React - a TeleLegal site

    Lecture 25 Project Demo

    Lecture 26 Project Structure and Front-end Setup

    Lecture 27 Back-end Setup

    Lecture 28 Creating a JWT & link to simulate scheduling

    Lecture 29 Add React-Router for our front-end

    Lecture 30 Setup Join-Video route and get the decoded data in React

    Lecture 31 Add starting components

    Lecture 32 Wire up redux and make callStatus reducer

    Lecture 33 Add action buttons, bootstrap, and fontawesome

    Lecture 34 getUserMedia() and store the stream in redux

    Lecture 35 Create peerConnection and store it in redux

    Lecture 36 Thinking through where our functions belong (& a few bug fixes)

    Lecture 37 Abstracting the Video and Audio buttons

    Lecture 38 Adding the local video feed

    Lecture 39 Add our tracks to the peerConnection

    Lecture 40 Enable and disable (mute) the local video feed

    Lecture 41 Display local video inputs (camera options)

    Lecture 42 Set new video device on select

    Lecture 43 replaceTracks()

    Lecture 44 Abstract DropDown component

    Lecture 45 Set up AudioButton component

    Lecture 46 Switch Audio Input and Output Devices

    Lecture 47 Start, mute, unmute audio

    Lecture 48 Organize offers on backEnd and add uuid

    Lecture 49 createOffer() once the tracks are shared

    Lecture 50 Add Dashboard markup for professional

    Lecture 51 Optional - Overview of markup

    Lecture 52 Authenticate professional

    Lecture 53 socket refactoring

    Lecture 54 Reorganize Appointment Data

    Lecture 55 Pull Appointment Data

    Lecture 56 Listen for offers

    Lecture 57 Create join video route for professional

    Lecture 58 Add Call Info to Professional Video Page

    Lecture 59 Create Answer and set Descriptions

    Lecture 60 Review Task List and Cheer for Our Progress

    Lecture 61 Emit Answer Up To Server

    Lecture 62 Listen for New Answer

    Lecture 63 Emit Ice Candidates

    Lecture 64 Send Ice Candidates to clients

    Lecture 65 Add Ice Candidates to Peer Connection

    Lecture 66 AddTracks and… VICTORY!!! (test app)

    Lecture 67 Fix 2 small bugs

    Lecture 68 Make the HangupButton do something!

    Lecture 69 ReplaceTracks on change device

    Anyone familiar with JavaScript and node who is new to webRTC!,Anyone familiar with the browser webRTC api, but can't seem to get the browsers to connect,Anyone who wants to put video conferencing in their web app for free (no 3rd party required)!