WebRTC Cookbook

Andrii Sergiienko

  • 出版商: Packt Publishing
  • 出版日期: 2015-02-28
  • 售價: $1,870
  • 貴賓價: 9.5$1,777
  • 語言: 英文
  • 頁數: 212
  • 裝訂: Paperback
  • ISBN: 1783284455
  • ISBN-13: 9781783284450
  • 下單後立即進貨 (約3~4週)

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商品描述

Get to grips with advanced real-time communication applications and services on WebRTC with practical, hands-on recipes

About This Book

  • Explore task-based recipes on integrating your WebRTC application with systems such as Asterisk and Freeswitch
  • Set up cutting-edge communicating networks by understanding the fundamentals of debugging, security, integration, attendant services, and more
  • Gain advanced knowledge of WebRTC with these incredibly effective recipes

Who This Book Is For

If you are a JavaScript developer with a basic knowledge of WebRTC and software development, but want to explore how to use it in more depth, this book is for you.

What You Will Learn

  • Understand the concepts of WebRTC while setting up rich web applications
  • Create a signaling server using Java or Erlang
  • Secure your WebRTC application by configuring your own STUN/TURN server
  • Integrate with VoIP platforms such as Asterisk and FreeSWITCH and abstract the complexities of communications protocols
  • Get acquainted with advanced debugging and troubleshooting techniques
  • Process videos using HTML5 filters
  • Develop native mobile applications using WebRTC and integrate them with web services
  • Discover advanced WebRTC techniques and solutions

In Detail

Using WebRTC, it is easy to develop in-browser applications and web services with extended multimedia features such as audio/video calls, VoIP, screen casting, peer-to-peer file transferring and more, without installing any third-party components/plugins on the client. In this book, you'll explore practical recipes covering the core aspects of application and service development with WebRTC. After a recap of the basics of developing rich media web applications and services, you will delve into core functionalities such as security and SSL, integration with other technologies such as VoIP and SIP, debugging clients and servers, working with filters, native applications for mobile platforms, and using third-party components.

By the end of this book, you will be able set up advanced, real-time, communicating networks with a full understanding of core WebRTC features.

商品描述(中文翻譯)

深入了解WebRTC的高級即時通訊應用和服務,並通過實際的實踐和操作指南來掌握它。

關於本書:
- 通過集成WebRTC應用程序與Asterisk和Freeswitch等系統的任務導向的操作指南,探索相關的實踐方法。
- 通過理解調試、安全性、集成、服務等基礎知識,建立尖端的通訊網絡。
- 通過這些極其有效的操作指南,獲得WebRTC的高級知識。

本書適合對WebRTC和軟件開發有基本了解的JavaScript開發人員,並希望深入探索如何更好地使用它的人。

你將學到什麼:
- 在設置豐富的Web應用程序時,了解WebRTC的概念。
- 使用Java或Erlang創建一個信令服務器。
- 通過配置自己的STUN/TURN服務器,保護你的WebRTC應用程序。
- 與Asterisk和FreeSWITCH等VoIP平台集成,抽象出通訊協議的複雜性。
- 熟悉高級調試和故障排除技術。
- 使用HTML5濾鏡處理視頻。
- 使用WebRTC開發原生移動應用程序,並將其與Web服務集成。
- 探索高級WebRTC技術和解決方案。

詳細內容:
使用WebRTC,可以輕鬆開發瀏覽器應用程序和Web服務,具有擴展的多媒體功能,如音頻/視頻通話、VoIP、屏幕投射、點對點文件傳輸等,而無需在客戶端安裝任何第三方組件/插件。在本書中,您將探索實踐操作指南,涵蓋WebRTC應用程序和服務開發的核心方面。在回顧開發豐富媒體Web應用程序和服務的基礎知識後,您將深入研究核心功能,如安全性和SSL、與VoIP和SIP等其他技術的集成、客戶端和服務器的調試、使用濾鏡、移動平台的原生應用程序以及使用第三方組件。

通過閱讀本書,您將能夠建立高級的實時通訊網絡,全面了解WebRTC的核心功能。