第三版WebRTC-APIs and RTCWEB Protocols of the HTML5 Real-Time Web原版完整.docx

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1、TABLE OF CONTENTSPreface1. Introduction to Web Real-Time Communications 2. How to Use WebRTC3. Local Media4. Signaling5. Peer-to-Peer Media6. Peer Connection and Offer/Answer Negotiation 7. Data Channel8. W3C Documents9. NAT and Firewall Traversal 10. Protocols11. IETF Documents12. IETF Related RFC

2、Documents 13. Security and Privacy14. Implementations and Uses A. The W3C Standards ProcessB. The IETF Standards Process C. GlossaryD. Supplementary Reading and Sources About the AuthorsAlan B. JohnstonDaniel C. BurnettThird EditionC:Digital Codex LLC 1 Copyright 2014 Digital Codex LLCP.O. Box 9131,

3、 St. Louis, MO 63117,USA Kindle eBook EditionAll rights reserved.All trademarks are the property of their respective holders.ISBN-13: 978-0-98597887-7ISBN-10: 0985978872DEDICATIONFor Aidan & Nora, and Craig & FionaACKNOWLEDGMENTSWe would like to thank our technical reviewers Alex Agranovsky, Carol D

4、avids, Emil Ivov, David Kemp, Henry Sinnreich, Harvey Waxman, and Dan York. We would also like to thank Marina Burnett and Chris Comfort for their proofreading and comments. We would also like to thank our families for their encouragement and support.And finally, we would like to acknowledge our col

5、leagues in W3C and the IETF who are working incredibly hard at creating the WebRTC standards.PREFACEThe changes in the world of WebRTC are amazing, but they sometimes feels schizophrenic. At times, it is moving forward rapidly, making major strides. At other times, it seems slow and almost glacial i

6、n its progress. Interoperability today is superb, and making our demo application work between browsers (Chrome and Firefox) and platforms (Windows and Mac) with voice, video, and the data channel has been a breeze. On the other hand, there are some interoperability clouds on the horizon with potent

7、ial video interop issues and even alternative APIs.Time will tell which personality will dominate going forward. Certainly, the standards and implementations are maturing and converging, and this makes our code writing and book writing much easier than it was back in the days of the first edition.Th

8、is edition has new content in the form of an enhanced demo application which now shows the use of the data channel for realtime text sent directly between browsers. Also, a full description of the browser media negotiation process including actual SDP session descriptions from Firefox and Chrome. Hi

9、nts on how to use Wireshark to monitor WebRTC protocols, and example captures are also included. TURN server support for NAT and firewall traversal is also new.This edition also features a step-by-step introduction to WebRTC, with concepts such as local media, signaling, and the Peer Connection intr

10、oduced through separate runnable demos. As always, all our code is available for download at and is up and running at for you to try.eBook Note: Code and lists will look best with the smallest font size of your device.We hope this new edition will be helpful to you in your WebRTC development and int

11、egration.PREFACE TO SECOND EDITIONWebRTC continues to evolve and grow in the handful of months since we published the first edition of WebRTC: APIs and Protocols of the HTML5 RealTime Web.There has been real progress in many areas in the IETF and W3C standards, although much work remains. Eventing,

12、stream representation at the protocol level, and even the syntax for some callbacks are all still very much under discussion, while details such as how multiple video tracks within a single MediaStream work and what should happen when a MediaStream attached to an HTML element has tracks added or rem

13、oved are only now beginning to be considered. Nevertheless, the core APIs are firming up. On the usage side, conferences, meetups, and startups have sprung up and are growing rapidly in size, with the World seeking to understand WebRTCs impact and opportunities.Other trends have become clear as well

14、. The disagreement about codecs, especially video codecs is escalating from a fight, to a war, to Mutually Assured Patent Destruction. The authors sincerely hope that the standards and industry come to agreement quickly on the mandatory to implement video codec very soon.While the basic parts of Web

15、RTC are working well in demos and applications today, many of the concerns and issues with WebRTC revolve around security and the signaling channel, which have complete new chapters in this second edition. Additionally, the sample code out there on the Web today is often either too complex or insuff

16、iciently explained, motivating the complete running and completely explained example in this new edition. This demo code, running on both Chrome and Firefox, is also on our website at .We hope that this new updated edition keeps you excited and informed about WebRTC. Happy reading!PREFACE TO FIRST E

17、DITIONThe Internet and the World Wide Web have changed our world. When the history of this period is written, much will be said about the impact of these technologies on life in the late 20th and early 21st centuries. The web has changed the way we receive information, interact with others, work, an

18、d play. Now, the web is about to dramatically change the way we communicate using voice and video. This book gives an up-to-the-minute snapshot of the standards and industry effort known as WebRTC, which is short for Web RealTime Communications. This technology, along with other advances in HTML5 br

19、owsers, has the potential to revolutionize the way we all communicate, in both personal and business spheres.The authors have been involved in the Internet Communications industry for many years, and have seen the advances and impact of the Internet on voice and video communications. We have worked

20、on signaling protocols such as Session Initiation Protocol (SIP), Session Description Protocol (SDP), and security protocols such as ZRTP for voice and video communication systems that will form the basis of what will inevitably replace the telephone system (called the Public Switched Telephone Netw

21、ork or PSTN). These Internet Communications technologies have brought an amazing wave of disruption, but we believe WebRTC has the potential to create even greater disruption.This book provides information for web developers and telephony developers who want to catch this new wave while it is still

22、building. The standards and protocols needed for WebRTC are still being developed and invented. Browsers are starting to support WebRTC functionality, little by little. However, the authors have seen the need for a book to explain this still- developing technology. This book will explain the technic

23、al goals, architectures, protocols, and Application Programming Interfaces (APIs) of WebRTC. In a publishing experiment, we plan to produce frequent editions of this book, perhaps as often as three per year, and focus on digital delivery and on-demand publishing to keep costs down and for maximum hy

24、per linking usefulness. For information on the latest edition and for a list of updates and changes, visit .This book begins with an introduction to WebRTC and discusses what isnew about it. The unique aspects of WebRTC peer-to-peer media flows are explored, and Network Address Translation (NAT) tra

25、versal explained. We then discuss the working documents and finalized documents that together comprise the WebRTC standards-in-progress in both the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF). Each chapter ends with a references section, listing the various standar

26、ds documents. References of the form RFC are IETF Request For Comments documents. References of the form draft- are IETF Internet-Drafts, working documents, whose content may have been updated or changed since the publication of this book. The hyperlinks provided will, in most cases, retrieve the mo

27、st recent version of the document. References to W3C drafts include a link to the latest public working draft, and also a link to the latest editors draft. For those of you unfamiliar with the standardization processes of W3C and the IETF, we have provided a reference in Appendices A and B. Finally,

28、 we discuss the current state of deployment in popular browsers.If you are a web developer, welcome to the world of InternetCommunications! Your users will greatly enjoy the ability to interact with each other using your applications realtime voice and video capabilities. To understand our descripti

29、ons of APIs, you will need a working knowledge of HTML and JavaScript, and some experience in web applications. See Appendix D for some useful references for this.If you are a VoIP or telephony developer, welcome to the web world! Your users will enjoy the capabilities of high quality audio and vide

30、o communication, and rich, web-powered user interfaces. To understand our descriptions of the on-the-wire protocols for transporting voice, video, and data, you will need a basic understanding of the Internet. Knowledge of another Internet Communication signaling protocol such as SIP or Jingle is al

31、so useful. See Appendix D for some additional useful reference reading.In many ways, WebRTC is a merging of worlds between the web and telephony. To help bridge the gap between the web and telephony world, we have also included a Glossary in Appendix C to briefly explain some common terms and concep

32、ts from each world.The authors look forward to participating in the next wave of disruption and innovation that WebRTC will likely unleash.We would love to hear from you and interact with you on Twitter (alanbjohnston and danielcburnett) or on Google+(alanbjohnston and danielcburnett).1 INTRODUCTION

33、 TO WEB REALTIME COMMUNICATIONSWeb RealTime Communications (RTC), or WebRTC, adds new functionality to the web browser. For the first time, browsers will interact directly with other browsers, resulting in a number of architectures including a triangle and trapezoid model. The media capabilities of

34、WebRTC are state-of-the-art, with many new features. The underlying standards of WebRTC are being developed by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF).1.1 WebRTC IntroductionWebRTC is an industry and standards effort to put realtime communications capabilit

35、ies into all browsers and make these capabilities accessible to web developers via standard HTML5 tags and JavaScript APIs (Application Programming Interfaces). For example, consider functionality similar to that offered by Skype SKYPE but without having to install any software or plugins. For a web

36、site or web application to work regardless of which browser is used, standards are required. Also, standards are required so that browsers can communicate with non-browsers, including enterprise and service provider telephony and communications equipment.1.1.1 The Web Browsing ModelThe basic model o

37、f web applications is shown in Figure 1.1. Transport of information between the browser and the web server is provided by the Hyper-Text Transport Protocol, HTTP (Section 10.2.1), which runs over Transmission Control Protocol, TCP (Section 10.2.9), or in some new implementations, over the WebSocket

38、protocol (see Section 10.2.2). The content or application is carried in Hyper-Text Markup Language, HTML, which typically includes JavaScript and Cascading Style Sheets CSS. In the simple case, the browser sends an HTTP request to the web server for content, and the web server sends a response conta

39、ining the document or image or other information requested. In the more complex case, the server sends JavaScript which runs on the browser, interacting with the browser through APIs and with the user through clicks and selects. The browser exchanges information with the server through an open HTTP

40、or WebSockets channel.Figure 1.1 Web Browser ModelIn figures in this book, we will show an arrow between the web browser and the web server to indicate the web session between them. Since WebRTC can utilize any web transport, the details of this connection, and whether it is HTTP or WebSockets, is n

41、ot discussed.1.1.2 The RealTime Communication Function in the BrowserFigure 1.2 shows the browser model and the role of the realtime communication function. The lighter block called “Browser RTC Function” is the focus of this book. The unique nature and requirements of realtime communications means

42、that adding and standardizing this block is non-trivial. The RTC function interacts with the web application using standard APIs. It communicates with the Operating System using the browser. A new aspect of WebRTC is the interaction that occurs browser-to-browser, known as a “Peer Connection”, where

43、 the RTC Function in one browser communicates using on- the-wire standard protocols (not HTTP) with the RTC Function in another browser or Voice over IP (VoIP) or video application. While web traffic uses TCP for transport, the on-the-wire protocol between browsers can use other transport protocols

44、such as User Datagram Protocol, UDP. Also new is the Signaling Server, which provides the signaling channel between the browser and the other end of the Peer Connection.Figure 1.2 RealTime Communication in the Browser1.1.3 Elements of a WebRTC SystemFigure 1.3 shows a typical set of elements in a We

45、bRTC system. This includes web servers, browsers running various operating systems on various devicesincluding desktop PCs, tablets, and mobile phones, and other servers. Additional elements include gateways to the Public Switched Telephone Network (PSTN) and other Internet communication endpoints s

46、uch as Session Initiation Protocol (SIP) phones and clients or Jingle clients. WebRTC enables communication among all these devices. The figures in this book will use these icons and elements as examples.Figure 1.3 Elements in a WebRTC Environment1.1.4 The WebRTC TriangleInitially, the most common s

47、cenario is likely to be where both browsers are running the same WebRTC web application, downloaded from the same website. This produces the WebRTC “Triangle” shown in Figure 1.4. This arrangement is called a triangle due to the shape of the signaling (sides of triangle) and media or data flows (bas

48、e of triangle) between the three elements. A Peer Connection establishes the transport for voice and video media and data channel flows directly between the browsers.Figure 1.4 The WebRTC TriangleNote that while we sometimes refer to the connection between the browser and server as signaling, it is not really signaling as used in telephony systems. Signaling is not standardized in WebRTC as it is just considered part of the application. This signal