Tutorials > Mastering Webrtc Part 2 – Real–Time Video And Screen–Share
Mastering Webrtc Part 2 – Real–Time Video And Screen–ShareDate: 21 July 2023 | Posted by: voska89 |
![]() Free Download 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 jаvascript 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 jаvascript APIs that no one seems to know, you should stick around. Like Express and other jаvascript/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 jаvascript 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)! 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