基于WebRTC的实时视音频通信研究综述

doi:10.11896/j.issn.1002-137X.2015.02.001

Abstract

Abstract: WebRTC (Web real-time communication) can achieve real-time video and audio communication on the Web without installation of any plugin or client software.It is accepted as W3C draft,because of low development cost and wide application range.WebRTC is bringing us an innovation era on multimedia communications.The overall framework of WebRTC-based real-time video and audio communication was introduced first,and then the related key technologies of WebRTC were elaborated,including signaling mechanism,Web app and WebRTC of low layer of browser.The latter also includes video codec module,audio codec module and transmission module.Next,implementation details of WebRTC real-time video and audio communication applications were compared from the point of view of two users and multiple users.Finally,open issues that are worthy of further study were discussed.

Key words: WebRTC,P2P,WebSocket,Video and audio communication,Video conference

ZHANG Xiang-hui, HUANG Jia-qing, WU Kang-heng and LEI Zhi-bin. Survey on Real-time Video and Audio Communication Based on WebRTC[J].Computer Science, 2015, 42(2): 1-6.

References

[1] Jennings C,Hardie T,Westerlund M.Real-time commu-nications for the web[J].Communications Magazine,IEEE,2013,51(4):20-26
[2] Bergkvist A,Burnett D C,Jennings C,et al.WebRTC 1.0:Real-time communication between browsers,W3C Working Draft 10 September 2013 [S/OL].http://www.w3.org/TR/webrtc/
[3] Burnett D C,Bergkvist A,Jennings C,et al.Media Capture and Streams,W3C Working Draft 03 September 2013[S/OL].http://www.w3.org/TR/mediacapture-streams/
[4] Singh K,Krishnaswamy V.A Case for SIP in Java-Script[J].Communications Magazine,IEEE,2013,51(4):28-33
[5] Amirante A,Castaldi T,Miniero L,et al.On the seamless interaction between WebRTC browsers and SIP-based conferencing systems[J].Communications Magazine,IEEE,2013,51(4):42-47
[6] 刘亚杰.P2P流媒体内容分发关键技术研究 [D].长沙:国防科学技术大学,2005
[7] 张国印,李军,王向辉,等.一种基于移动 P2P 改进的 Gossip 算法[J].计算机科学,2013,40(9):103-105
[8] Uberti J,Jennings C.Javascript Session Establishment Protocol,draft-ietf-rtcweb-jsep-03,August 29 2013[S/OL].https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-03
[9] Handley M,Perkins C,Jacobson V.RFC 4566:SDP:session description protocol[S].IETF,July 2006
[10] Ciurana E.Developing with Google App Engine[M].Apress,2009:11-32
[11] Fette I,Melnikov A.RFC 6455:The WebSocket Proto-col[S].IETF,December 2011
[12] Hickson I.The WebSocket API,W3C Working Draft 09 August 2012[S/OL].http://www.w3.org/TR/2012/WD-websockets-20120809
[13] 孙清国,朱玮,刘华军.Web应用中的服务器推送技术研究综述[J].计算机系统应用,2008,7(11):116-120
[14] Garrett J J.Ajax:A new approach to web applica-tions[EB/OL].2005.http://www.adaptivepath.com/ideas/ajax-new-approach-web-applications/
[15] Void B Y.Node.js开发指南[M].北京:人民邮电出版社,2012:80-97
[16] Ornbo G.Sams Teach Yourself Node.js in 24 Hours[M].傅强,陈宗斌,译.北京:人民邮电出版,2013:148-185
[17] Jones P,Shabestary T,Alvestrand H,et al.RTP Payload Format for the iSAC Codec[S/OL].2013.https://tools.ietf.org/html/draft-ietf-avt-rtp-isac-04
[18] Duric A,Andersen S.RFC 3952:Real-time Transport Protocol (RTP) Payload Format for internet Low Bit Rate Codec (iLBC) Speech[S].IETF,2004 (下转第32页)(上接第6页)
[19] Andersen S,Duric A,Astrom H,et al.RFC 3951:In-ternet low bit rate codec (iLBC)[S].IETF,2004
[20] Valin J M,Vos K,Terriberry T.RFC 6716:Definition of theopus audio codec[S].IETF,2012
[21] Bankoski J,Wilkins P,Xu Y.Technical overview of VP8,an open source video codec for the Web [C]∥2011 IEEE International Conference on Multimedia and Expo (ICME).Barcelona,2011:1-6
[22] Rosenberg J.RFC 5245:Interactive connectivity estab-lishment(ICE):A protocol for network address translator (NAT) traversal for offer/answer protocols[S].IETF,2010
[23] Petit-Huguenin M.RFC 5928:Traversal Using Relays around NAT (TURN) Resolution Mechanism[S].IETF,2010
[24] Rosenberg J,Mahy R,Matthews P.RFC 5389:Session traversal utilities for NAT (STUN)[S].IETF,2008
[25] 竹洪涛.一种基于SIP和WebRTC的实时可视对讲方案设计[D].成都:西南交通大学,2013
[26] Mahy R,Matthews P,Rosenberg J.RFC 5766:Traversal using relays around NAT (TURN):relay extensions to session traversal utilities for NAT (STUN)[S].IETF,2010
[27] Perkins C,Westerlund M,Ott J.Web Real-Time Com-munication (WebRTC):Media Transport and Use of RTP[S/OL].2012.https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-06
[28] 郭远威.P2P流媒体关键算法的研究[D].长沙:中南大学,2012
[29] Becke M,Rathgeb E P,Werner S,et al.DataChannel considerations for RTCWeb[J].Communications Mag-azine,IEEE,2013,51(4):34-41
[30] 张春红,裘晓峰,弭伟,等.P2P 技术全面解析[M].北京:人民邮电出版社,2010:149-164
[31] Pouwelse J,Garbacki P,Epema D,et al.The BitTorrent P2Pfile-sharing system:Measurements and analy-sis[M]∥Peer-to-Peer Systems IV.Springer Berlin Hei-delberg,2005:205-216
[32] Johnston A B,Burnett D C.WebRTC:APIs and RTCWEB Protocols of the HTML5 Real-Time Web[M].Digital Codex LLC,2012:9-11
[33] Johnston A,Yoakum J,Singh K.Taking on WebRTC in an Enterprise[J].Communications Magazine,IEEE,2013,51(4):48-54
[34] Nurminen J K,Meyn A J R,Jalonen E,et al.P2P media streaming with HTML5 and WebRTC[C]∥IEEE International Conference on Computer Communications.Turin:IEEE,2013:63-64
[35] Xu Y,Yu C,Li J,et al.Video telephony for end-consumers:Measurement study of Google+,iChat,and Skype[C]∥Proceedings of the 2012 ACM conference on Internet measurement conference.ACM,2012:371-384

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