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所在平台: Udemy |
课程主页: https://www.udemy.com/course/webrtcunveiled/
课程评论：没有评论
课程名称：深入掌握WebRTC：使用SocketIO构建视频通话 课程概述：解锁WebRTC的强大功能，深入实时通信的世界！无论您是开发者、工程师还是技术爱好者，本课程将引导您逐步了解WebRTC的复杂工作原理，展示如何准备演示，并详细说明背后发生的事情。从理解基础知识到分解实现实时通信所需的关键协议，您将获得自信地在自己的项目中实施和排除WebRTC技术故障的技能。 您将准备的演示包括： - 从浏览器获取摄像头和麦克风访问权限。 - 使用RTCPeerConnection在两个浏览器标签之间进行视频通话。 - 使用RTCDataChannel在两个标签之间发送和接收消息。 - 使用SocketIO和Node.JS准备点对点视频通话应用程序。 - 与GenAI助手进行视频通话。 您将学习到的内容： - 理解WebRTC技术的核心概念以及实时通信的工作原理。 - 探索WebRTC架构，包括点对点连接、信令和ICE（交互式连接建立）。 - 深入了解支持WebRTC的协议，如SDP（会话描述协议）、RTP、RTCP和SRTP。 - 发现WebRTC中如何使用STUN和TURN服务器进行NAT穿透。 - 学习媒体路径，包括媒体编码、数据包化和安全机制。 - 了解在WebRTC中使用的各种媒体服务器架构，如SFU、MCU。 - 理解适合您WebRTC应用程序的媒体服务器架构。 适合人群： - 希望为应用程序添加实时通信功能的Web开发者。 - 希望了解WebRTC技术内部工作原理的软件工程师。 - 对现代浏览器如何处理实时媒体通信感兴趣的技术爱好者。 - 寻求在WebRTC领域发展的工程师。 课程要求：具备基本的网络和Web开发概念，了解HTML、CSS和JavaScript将有帮助。 更多内容：想了解并部署您自己的TURNSERVER？请关注我的TURNSERVER课程。在该课程中，您将学习NAT存在的原因、STUN和TURN协议，并为您的应用程序部署自己的TURNSERVER，然后验证其功能。
Unlock the power of WebRTC and dive deep into the world of real-time communication! Whether you're a developer, engineer, or technology enthusiast, this course will guide you through the intricate workings of WebRTC, showing you how to prepare demos and telling you exactly what happens under the hood. From understanding the fundamentals to breaking down the key protocols that make real-time communication possible, you'll gain the skills to confidently implement and troubleshoot WebRTC technology in your own projects.Prepare Demo for:Getting Access of Camera and Mic from browser.Video call between two tabs in browser using RTCPeerConnection. Send and Receive messages between two tabs in browser using RTCDataChannel.Prepare Peer-to-Peer Video Call app with SocketIO and Node.JS.Video Call with GenAI Assistant.What You Will Learn:Understand the core concepts behind WebRTC technology and how real-time communication works.Explore the WebRTC architecture, including peer-to-peer connections, signaling, and ICE (Interactive Connectivity Establishment).Deep dive into the protocols that power WebRTC, such as SDP (Session Description Protocol), RTP, RTCP and SRTP.Discover how NAT traversal works in WebRTC using STUN and TURN servers.Learn about the media path, including media encoding, packetization, and security mechanisms.Learn about various Media Server architecture such as SFU, MCU which are used in WebRTC.Understand which Media Server architecture is suitable for your WebRTC Application.Who This Course Is For:Web Developers looking to add real-time communication capabilities to their applications.Software Engineers who want to understand the inner workings of WebRTC technology.Technology Enthusiasts interested in how modern web browsers handle real-time media communication.Engineers seeking career in WebRTC. Requirements:Basic understanding of networking and web development concepts HTML, CSS and JavaScript is helpful.MORE: Interested in understanding and deploying your own TURNSERVER. Head on to my course on TURNSERVER. Their you will learn why NATs exist, what is STUN and TURN Protocol. Will deploy your own TURNSERVER for your application and then verify it too for its functionality.
