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

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1、TABLE OF CONTENTSPreface1.Introduction to Web Real-Time Communications2.How to Use WebRTC3.Local Media4.Signaling5.Peer-to-Peer Media6.Peer Connection and Offer/Answer Negotiation7.Data Channel8.W3C Documents9.NAT and Firewall Traversal10.Protocols11.IETF Documents 12.IETF Related RFC Documents13.Se

2、curity and Privacy14.Implementations and UsesA.The W3C Standards Process B.The IETF Standards ProcessC.GlossaryD.Supplementary Reading and SourcesAbout the Authors Alan B.Johnston Daniel C.BurnettThird EditionC:Digital Codex LLC 1 Copyright 2014 Digital Codex LLCP.O.Box 9131,St.Louis,MO 63117,USA ht

3、tp:/Kindle eBook Edition All rights reserved.All trademarks are the property of their respective holders.ISBN-13:978-0-98597887-7ISBN-10:0985978872DEDICATIONFor Aidan&Nora,and Craig&Fiona ACKNOWLEDGMENTS We would like to thank our technical reviewers Alex Agranovsky,CarolDavids,Emil Ivov,David Kemp,

4、Henry Sinnreich,Harvey Waxman,and DanYork.We would also like to thank Marina Burnett and Chris Comfort for theirproofreading and comments.We would also like to thank our families for theirencouragement and support.And finally,we would like to acknowledge our colleagues in W3C and theIETF who are wor

5、king incredibly hard at creating the WebRTC standards.PREFACE The changes in the world of WebRTC are amazing,but they sometimes feelsschizophrenic.At times,it is moving forward rapidly,making major strides.Atother times,it seems slow and almost glacial in its progress.Interoperabilitytoday is superb

6、,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 someinteroperability clouds on the horizon with potential video interop issues andeven alternative APIs.Time wil

7、l tell which personality will dominate going forward.Certainly,thestandards and implementations are maturing and converging,and this makesour code writing and book writing much easier than it was back in the days ofthe first edition.This edition has new content in the form of an enhanced demo applic

8、ationwhich now shows the use of the data channel for realtime text sent directlybetween browsers.Also,a full description of the browser media negotiationprocess including actual SDP session descriptions from Firefox and Chrome.Hints on how to use Wireshark to monitor WebRTC protocols,and examplecapt

9、ures are also included.TURN server support for NAT and firewalltraversal is also new.This edition also features a step-by-step introduction to WebRTC,withconcepts such as local media,signaling,and the Peer Connection introducedthrough separate runnable demos.As always,all our code is available fordo

10、wnload at http:/ and is up and running athttp:/ for you to try.eBook Note:Code and lists will look best with the smallest font size of yourdevice.We hope this new edition will be helpful to you in your WebRTCdevelopment and integration.PREFACE TO SECOND EDITION WebRTC continues to evolve and grow in

11、 the handful of months since wepublished the first edition of WebRTC:APIs and Protocols of the HTML5RealTime Web.There has been real progress in many areas in the IETF and W3C standards,although much work remains.Eventing,stream representation at the protocollevel,and even the syntax for some callba

12、cks are all still very much underdiscussion,while details such as how multiple video tracks within a singleMediaStream work and what should happen when a MediaStream attached toan HTML element has tracks added or removed are only now beginning to beconsidered.Nevertheless,the core APIs are firming u

13、p.On the usage side,conferences,meetups,and startups have sprung up and are growing rapidly insize,with the World seeking to understand WebRTCs impact andopportunities.Other trends have become clear as well.The disagreement about codecs,especially video codecs is escalating from a fight,to a war,to

14、MutuallyAssured Patent Destruction.The authors sincerely hope that the standards andindustry come to agreement quickly on the mandatory to implement videocodec very soon.While the basic parts of WebRTC are working well in demos andapplications today,many of the concerns and issues with WebRTC revolv

15、earound security and the signaling channel,which have complete new chaptersin this second edition.Additionally,the sample code out there on the Webtoday is often either too complex or insufficiently explained,motivating thecomplete running and completely explained example in this new edition.Thisdem

16、o code,running on both Chrome and Firefox,is also on our website athttp:/.We hope that this new updated edition keeps you excited and informed aboutWebRTC.Happy reading!PREFACE TO FIRST EDITION The Internet and the World Wide Web have changed our world.When thehistory of this period is written,much

17、will be said about the impact of thesetechnologies on life in the late 20th and early 21st centuries.The web haschanged the way we receive information,interact with others,work,and play.Now,the web is about to dramatically change the way we communicate usingvoice and video.This book gives an up-to-t

18、he-minute snapshot of the standardsand industry effort known as WebRTC,which is short for Web RealTimeCommunications.This technology,along with other advances in HTML5browsers,has the potential to revolutionize the way we all communicate,inboth personal and business spheres.The authors have been inv

19、olved in the Internet Communications industry formany years,and have seen the advances and impact of the Internet on voice andvideo communications.We have worked on signaling protocols such asSession Initiation Protocol(SIP),Session Description Protocol(SDP),andsecurity protocols such as ZRTP for vo

20、ice and video communication systemsthat will form the basis of what will inevitably replace the telephone system(called the Public Switched Telephone Network or PSTN).These InternetCommunications technologies have brought an amazing wave of disruption,butwe believe WebRTC has the potential to create

21、 even greater disruption.This book provides information for web developers and telephonydevelopers who want to catch this new wave while it is still building.Thestandards and protocols needed for WebRTC are still being developed andinvented.Browsers are starting to support WebRTC functionality,littl

22、e bylittle.However,the authors have seen the need for a book to explain this still-developing technology.This book will explain the technical goals,architectures,protocols,and Application Programming Interfaces(APIs)ofWebRTC.In a publishing experiment,we plan to produce frequent editions ofthis book

23、,perhaps as often as three per year,and focus on digital delivery andon-demand publishing to keep costs down and for maximum hyper linkingusefulness.For information on the latest edition and for a list of updates andchanges,visit http:/.This book begins with an introduction to WebRTC and discusses w

24、hat isnew about it.The unique aspects of WebRTC peer-to-peer media flows areexplored,and Network Address Translation(NAT)traversal explained.Wethen discuss the working documents and finalized documents that togethercomprise the WebRTC standards-in-progress in both the World Wide WebConsortium(W3C)an

25、d the Internet Engineering Task Force(IETF).Eachchapter ends with a references section,listing the various standardsdocuments.References of the form RFC are IETF Request For Commentsdocuments.References of the form draft-are IETF Internet-Drafts,working documents,whose content may have been updated

26、or changed sincethe publication of this book.The hyperlinks provided will,in most cases,retrieve the most recent version of the document.References to W3C draftsinclude a link to the latest public working draft,and also a link to the latesteditors draft.For those of you unfamiliar with the standardi

27、zation processesof W3C and the IETF,we have provided a reference in Appendices A and B.Finally,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 eachother

28、 using your applications realtime voice and video capabilities.Tounderstand our descriptions of APIs,you will need a working knowledge ofHTML and JavaScript,and some experience in web applications.SeeAppendix D for some useful references for this.If you are a V oIP or telephony developer,welcome to

29、the web world!Yourusers will enjoy the capabilities of high quality audio and videocommunication,and rich,web-powered user interfaces.To understand ourdescriptions of the on-the-wire protocols for transporting voice,video,anddata,you will need a basic understanding of the Internet.Knowledge of anoth

30、erInternet Communication signaling protocol such as SIP or Jingle is also useful.See Appendix D for some additional useful reference reading.In many ways,WebRTC is a merging of worlds between the web andtelephony.To help bridge the gap between the web and telephony world,wehave also included a Gloss

31、ary in Appendix C to briefly explain some commonterms and concepts from each world.The authors look forward to participating in the next wave of disruption andinnovation that WebRTC will likely unleash.We would love to hear from you and interact with you on Twitter(alanbjohnston and danielcburnett)o

32、r on Google+( and ).1 INTRODUCTION TO WEB REALTIMECOMMUNICATIONS Web RealTime Communications(RTC),or WebRTC,adds new functionality tothe web browser.For the first time,browsers will interact directly with otherbrowsers,resulting in a number of architectures including a triangle andtrapezoid model.Th

33、e media capabilities of WebRTC are state-of-the-art,withmany new features.The underlying standards of WebRTC are being developedby the World Wide Web Consortium(W3C)and the Internet Engineering TaskForce(IETF).1.1 WebRTC IntroductionWebRTC is an industry and standards effort to put realtime communic

34、ationscapabilities into all browsers and make these capabilities accessible to webdevelopers via standard HTML5 tags and JavaScript APIs(ApplicationProgramming Interfaces).For example,consider functionality similar to thatoffered by Skype SKYPE but without having to install any software orplugins.Fo

35、r a website or web application to work regardless of whichbrowser is used,standards are required.Also,standards are required so thatbrowsers can communicate with non-browsers,including enterprise andservice provider telephony and communications equipment.1.1.1 The Web Browsing ModelThe basic model o

36、f web applications is shown in Figure 1.1.Transport ofinformation between the browser and the web server is provided by theHyper-Text Transport Protocol,HTTP(Section 10.2.1),which runs overTransmission Control Protocol,TCP(Section 10.2.9),or in some newimplementations,over the WebSocket protocol(see

37、 Section 10.2.2).Thecontent or application is carried in Hyper-Text Markup Language,HTML,which typically includes JavaScript and Cascading Style Sheets CSS.In thesimple case,the browser sends an HTTP request to the web server for content,and the web server sends a response containing the document or

38、 image or otherinformation requested.In the more complex case,the server sends JavaScriptwhich runs on the browser,interacting with the browser through APIs and withthe user through clicks and selects.The browser exchanges information withthe server through an open HTTP or WebSockets channel.Figure

39、1.1 Web Browser Model In figures in this book,we will show an arrow between the web browserand the web server to indicate the web session between them.Since WebRTCcan utilize any web transport,the details of this connection,and whether it isHTTP or WebSockets,is not discussed.1.1.2 The RealTime Comm

40、unication Function in the BrowserFigure 1.2 shows the browser model and the role of the realtimecommunication function.The lighter block called“Browser RTC Function”isthe focus of this book.The unique nature and requirements of realtimecommunications means that adding and standardizing this block is

41、 non-trivial.The RTC function interacts with the web application using standard APIs.Itcommunicates with the Operating System using the browser.A new aspect ofWebRTC is the interaction that occurs browser-to-browser,known as a“PeerConnection”,where the RTC Function in one browser communicates using

42、on-the-wire standard protocols(not HTTP)with the RTC Function in anotherbrowser or V oice over IP(V oIP)or video application.While web traffic usesTCP for transport,the on-the-wire protocol between browsers can use othertransport protocols such as User Datagram Protocol,UDP.Also new is theSignaling

43、Server,which provides the signaling channel between the browserand the other end of the Peer Connection.Figure 1.2 RealTime Communication in the Browser 1.1.3 Elements of a WebRTC SystemFigure 1.3 shows a typical set of elements in a WebRTC system.This includesweb servers,browsers running various op

44、erating systems on various devicesincluding desktop PCs,tablets,and mobile phones,and other servers.Additional elements include gateways to the Public Switched TelephoneNetwork(PSTN)and other Internet communication endpoints such as SessionInitiation Protocol(SIP)phones and clients or Jingle clients

45、.WebRTC enablescommunication among all these devices.The figures in this book will use theseicons and elements as examples.Figure 1.3 Elements in a WebRTC Environment 1.1.4 The WebRTC TriangleInitially,the most common scenario is likely to be where both browsers arerunning the same WebRTC web applic

46、ation,downloaded from the samewebsite.This produces the WebRTC“Triangle”shown in Figure 1.4.Thisarrangement is called a triangle due to the shape of the signaling(sides oftriangle)and media or data flows(base of triangle)between the threeelements.A Peer Connection establishes the transport for voice

47、 and videomedia and data channel flows directly between the browsers.Figure 1.4 The WebRTC Triangle Note that while we sometimes refer to the connection between the browserand server as signaling,it is not really signaling as used in telephony systems.Signaling is not standardized in WebRTC as it is

48、 just considered part of theapplication.This signaling may run over HTTP or WebSockets to the sameweb server that serves HTML pages to the browser,or to a completelydifferent web server that just handles the signaling.1.1.5 The WebRTC TrapezoidFigure 1.5 shows the WebRTC Trapezoid draft-ietf-rtcweb-

49、overview,basedon the SIP Trapezoid RFC3261.The two web servers are showncommunicating using a standard signaling protocol such as Session InitiationProtocol(SIP),used by many V oIP and video conferencing systems,or JingleXEP-0166,used to add voice and video capability to Jabber RFC6120instant messag

50、ing and presence systems.Alternatively,a proprietary signalingprotocol could be used.Note that in these more complicated cases,the mediamay not flow directly between the two browsers,but may go through mediarelays and other elements,as discussed in Chapter 5.Figure 1.5 The WebRTC Trapezoid 1.1.6 Web