1. html5rocs: https://www.html5rocks.com/ko/tutorials/webrtc/basics/
Getting Started with WebRTC - HTML5 Rocks
Plugin-free, realtime communication of video, audio and data using WebRTC.
www.html5rocks.com
1-1. stun, turn server: https://www.html5rocks.com/en/tutorials/webrtc/infrastructure/
WebRTC in the real world: STUN, TURN and signaling - HTML5 Rocks
Build the back-end services you need to run a WebRTC application.
www.html5rocks.com
1-2 socket.io
Socket.IO
SOCKET.IO 2.0 IS HERE FEATURING THE FASTEST AND MOST RELIABLE REAL-TIME ENGINE ~/Projects/tweets/index.js var io = require('
socket.io
1-2-1 socket.io emit cheat seat
https://socket.io/docs/emit-cheatsheet/
Emit cheatsheet
io.on('connect', onConnect);function onConnect(socket){ // sending to the client socket.emit('hello', 'can you hear me?', 1, 2, 'abc'); // sending to all clients except sender socket.broadcas
socket.io
2. 구글 코드랩: https://codelabs.developers.google.com/codelabs/webrtc-web/#0
Real time communication with WebRTC
A complete version of this step is in the step-2 folder. RTCPeerConnection is an API for making WebRTC calls to stream video and audio, and exchange data. This example sets up a connection between two RTCPeerConnection objects (known as peers) on the same
codelabs.developers.google.com
3. coturn 프로젝트: https://github.com/coturn/coturn
coturn/coturn
coturn TURN server project. Contribute to coturn/coturn development by creating an account on GitHub.
github.com
3-1 coturn tutorial : https://ourcodeworld.com/articles/read/1175/how-to-create-and-configure-your-own-stun-turn-server-with-coturn-in-ubuntu-18-04
How to create and configure your own STUN/TURN server with coturn in Ubuntu 18.04
Learn how to configure your own stun/turn server using coturn in Ubuntu 18.04 from scratch.
ourcodeworld.com
3-2 singaling server pm2 : https://pm2.keymetrics.io/
4. MDN web docs: https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API
WebRTC API
WebRTC (Web Real-Time Communication) is a technology which enables Web applications and sites to capture and optionally stream audio and/or video media, as well as to exchange arbitrary data between browsers without requiring an intermediary.
developer.mozilla.org
WebRTC에 대한 상세한 정보
튜토리얼, 함수들, 가이드
4-1 MDN web docs: https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Signaling_and_video_calling
Signaling and video calling
WebRTC allows real-time, peer-to-peer, media exchange between two devices. A connection is established through a discovery and negotiation process called signaling. This tutorial will guide you through building a two-way video-call.
developer.mozilla.org
5. W3C editor's draft: https://w3c.github.io/mediacapture-main/getusermedia.html#intro
Media Capture and Streams
Initial Author of this Specification was Ian Hickson, Google Inc., with the following copyright statement: © Copyright 2004-2011 Apple Computer, Inc., Mozilla Foundation, and Opera Software ASA. You are granted a license to use, reproduce and create deriv
w3c.github.io
webRTC 함수들에 대한 상세한 설명
6. adapater.js: https://github.com/webrtc/adapter
webrtc/adapter
READ ONLY FORK: Shim to insulate apps from spec changes and prefix differences. Latest adapter.js release: - webrtc/adapter
github.com
6. apprtc: https://github.com/webrtc/apprtc
webrtc/apprtc
The video chat demo app based on WebRTC. This project is currently on HOLD with minimal maintenance. - webrtc/apprtc
github.com
Perfect negotiation in webrtc: https://blog.mozilla.org/webrtc/perfect-negotiation-in-webrtc/
Perfect negotiation in WebRTC - Advancing WebRTC
New preface: What if you could add and remove media to and from a live WebRTC connection, without having to worry about state, glare (signaling collisions), role (what side you’re on), or what condition the connection is in? You’d simply call pc.addTra
blog.mozilla.org
Real-time communication with WebRTC: Google I/O 2013
슬라이드 노트: http://io13webrtc.appspot.com/#35
http://io13webrtc.appspot.com/#35
On one hand, we have STUN, which is cheap, but doesn't always work. And on the other, we have TURN, which always works, but has an operational cost. Fortunately, we can get the best of both worlds. WebRTC uses a mechanism called ICE, Interactive_Connectivi
io13webrtc.appspot.com
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