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所在平台: Udemy |
课程主页: https://www.udemy.com/course/webrtc-by-projects/
课程评论：没有评论
**Coursera 课程总结：WebRTC 实战项目** 本课程是一门理论与实践相结合的 WebRTC 入门及进阶课程，特别适合希望学习如何将 WebRTC 与 Canvas 结合使用的开发者。课程将通过一步步构建项目，深入讲解 WebRTC 的核心概念。 **核心内容概览：** * **MediaStream：** 什么是 MediaStream？如何获取？ * **RTCPeerConnection：** 什么是 RTCPeerConnection 对象？信令（signaling）过程的各个步骤。 * **ICE Candidate & NAT 穿越：** 什么是 ICE Candidate？STUN 和 TURN 服务器的作用，以及 ICE 机制如何收集和选择网络候选者，实现端到端（peer-to-peer）通信，即使在 NAT 网络下。 **课程结构：** * **第一部分：服务器搭建与部署** * 使用 Node.js 和 Socket.IO 搭建信令服务器。 * 演示如何通过 GitHub 将项目部署到 Heroku。 * **第二部分：实时视频通信项目** * 构建第一个 WebRTC 项目，实现基础的实时视频通话功能。 * **第三部分：Canvas 实时互动视频项目** * 构建第二个项目，在实时视频基础上增加 Canvas 互动功能。用户可以在视频画面上进行绘制（通过鼠标），并将 Canvas 的内容通过 WebRTC 发送，而非直接发送原始视频流。
This course is a mixte of theory and practice. This course is designed for beginners and intermediate Webrtc developers and also all who want to know how to use Webrtc with Canvas.We will build projects step by step to understand all the basics of WebrtcYou will find easy explanations of questions like, what is a MediaStream? how to get this object? what is an RTCPeerConnection object and what are all step of a signaling process.Also you will understand what is Ice Candidate, why we need STUN and TURN servers and what is ICE mechanism to collect and select the right network candidates to have a peer to peer communication even through NAT NetworkIn this course we have 3 sections:- In First section we will create a nodejs server with Socket io (for signaling process) and will show you how to deploy to Heroku using Github- In section 2, we create the first project, a realtime video communication using Webrtc- in section 3: we build our second projects, is a realtime video communication but using canvas, a User will draw with a mousse on a live video and send the contexte of the canvas using Webrtc instead of sending directly the live video from media source
