SIP流媒体广播应用服务器体系结构的研究
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摘要
随着IP技术的发展和日趋成熟,以VoIP(Voice over Internet Protocol)为代表的NGN(Next Generation Network)通信作为一种典型的宽带应用正面临着前所未有的发展机遇。SIP(Session Initiation Protocol)协议作为VoIP中核心控制协议之一,为语音、视频、数据业务的融合提供了一个综合的开放平台。在这个平台上,IP电话、IP传真、IP会议电视、流媒体广播等领域不断取得新的应用。
基于SIP的流媒体业务就是其中一项新应用,它是将传统的网络流媒体技术应用到IP通信领域,并与PSTN网、3G网相衔接,真正实现流媒体业务统一到IP网上进行。本文将从流媒体服务入手,深入研究相关的体系结构和实现所需的技术要素,探讨基于SIP的流媒体广播应用服务器的实现技术。
本文的要点包括:第一,介绍了研究的背景和意义、流媒体技术的原理、SIP及其相关协议和SIP/PSTN互通;第二,根据B2BUA(Back-to-Back User Agent)服务器的工作方式,提出了SIP流媒体广播应用服务器的工作模式和体系结构;第三,详细介绍了服务器的各个功能模块以及逻辑处理关系;第四,概要介绍了服务器各个模块的实现细节及各个模块处理的一些重要数据结构,并且探讨了系统实现中的一些相关的实现细节,其中信令方面主要采用SIP协议栈进行处理,媒体方面,本文主要采用H.264标准进行编解码,并采用中间格式技术进行转码,传输方面,采用UDP协议实现对RTP/RTCP网络传输的支持;第五,在负载均衡方面,本文认为负载均衡是一个系统的工程,需要在从文件来源到用户的播放终端这条路径上的各个环节进行研究。依据SIP协议的特性,使用SIP Proxy服务器作为负载平衡的服务器,根据不同的拨号规划,对负载进行调节;第六,根据广播服务器的特点,对解码和编码端采用位流和管道缓冲机制。
通过对服务器原型系统的测试,该系统基本实现了流媒体广播服务的功能,在此基础上只要对原型系统进一步优化和完善,就可以产生更大的实际应用价值。
With the development and maturity of IP technologies,as a typical application of Broad Band, NGN communication on behalf of VoIP is facing unprecedented opportunity for development. As one of the core control protocols in VoIP, SIP provides a comprehensive and open platform to integrate the audio, video and data. On this platform, fields such as IP telephone,IP fax,IP video conference, streaming media broadcast have made new progress in application unceasingly.
SIP-based streaming media business is one of these new applications. It applies the traditional network streaming media technology to IP network, and connects with PSTN network and 3G network, which really realizes streaming media business being unified in progress to IP network. Beginning with streaming media service, this article is aimed at thoroughly studying the related system and the architecture of the server as well as the technical elements to implement it. And the discussion of implementing SIP-based streaming media application server is also included.
The article includes the following main points. First, it introduces the background and significance of the research, the technology theory of streaming media, SIP and interraleted protocol, SIP/PSTN exchange; Second, according to B2BUA server mechanics, the article introduces work model and architecture of SIP-based streaming media application server; Third,each function module of the server and mutual logic treatment relation have been introduced in detail; Fourth, the modules’implementation details and some important data structures are given briefly and the related implementation details in the prototype system of the server are discussed, and these mainly include the use of the SIP stack for signaling processing, the H.264 standard for the media processing of encoding and decoding, the intermediate format technology for transfering the code,and the UDP protocol to realize the RTP/RTCP network transmission ; Fifth, load balancing is a systematic project in the field of load balance. In the road link from the media resources to the user terminal, the load balancing should be considered. Based on the characteristics of SIP, use the SIP Proxy Server as the load balancing server to balance the broadcasting server’s load through the dial plan management of SIP Proxy server; Sixth, according to the characteristics of the broadcasting server, apply bit-stream and pipe cushioning mechanism to encoding and decoding.
Through the testing of the server prototype system, it has basically realized the streaming media broadcast service function, after the further optimizations based on this prototype system, it may have more practical use.
基于SIP的流媒体业务就是其中一项新应用,它是将传统的网络流媒体技术应用到IP通信领域,并与PSTN网、3G网相衔接,真正实现流媒体业务统一到IP网上进行。本文将从流媒体服务入手,深入研究相关的体系结构和实现所需的技术要素,探讨基于SIP的流媒体广播应用服务器的实现技术。
本文的要点包括:第一,介绍了研究的背景和意义、流媒体技术的原理、SIP及其相关协议和SIP/PSTN互通;第二,根据B2BUA(Back-to-Back User Agent)服务器的工作方式,提出了SIP流媒体广播应用服务器的工作模式和体系结构;第三,详细介绍了服务器的各个功能模块以及逻辑处理关系;第四,概要介绍了服务器各个模块的实现细节及各个模块处理的一些重要数据结构,并且探讨了系统实现中的一些相关的实现细节,其中信令方面主要采用SIP协议栈进行处理,媒体方面,本文主要采用H.264标准进行编解码,并采用中间格式技术进行转码,传输方面,采用UDP协议实现对RTP/RTCP网络传输的支持;第五,在负载均衡方面,本文认为负载均衡是一个系统的工程,需要在从文件来源到用户的播放终端这条路径上的各个环节进行研究。依据SIP协议的特性,使用SIP Proxy服务器作为负载平衡的服务器,根据不同的拨号规划,对负载进行调节;第六,根据广播服务器的特点,对解码和编码端采用位流和管道缓冲机制。
通过对服务器原型系统的测试,该系统基本实现了流媒体广播服务的功能,在此基础上只要对原型系统进一步优化和完善,就可以产生更大的实际应用价值。
With the development and maturity of IP technologies,as a typical application of Broad Band, NGN communication on behalf of VoIP is facing unprecedented opportunity for development. As one of the core control protocols in VoIP, SIP provides a comprehensive and open platform to integrate the audio, video and data. On this platform, fields such as IP telephone,IP fax,IP video conference, streaming media broadcast have made new progress in application unceasingly.
SIP-based streaming media business is one of these new applications. It applies the traditional network streaming media technology to IP network, and connects with PSTN network and 3G network, which really realizes streaming media business being unified in progress to IP network. Beginning with streaming media service, this article is aimed at thoroughly studying the related system and the architecture of the server as well as the technical elements to implement it. And the discussion of implementing SIP-based streaming media application server is also included.
The article includes the following main points. First, it introduces the background and significance of the research, the technology theory of streaming media, SIP and interraleted protocol, SIP/PSTN exchange; Second, according to B2BUA server mechanics, the article introduces work model and architecture of SIP-based streaming media application server; Third,each function module of the server and mutual logic treatment relation have been introduced in detail; Fourth, the modules’implementation details and some important data structures are given briefly and the related implementation details in the prototype system of the server are discussed, and these mainly include the use of the SIP stack for signaling processing, the H.264 standard for the media processing of encoding and decoding, the intermediate format technology for transfering the code,and the UDP protocol to realize the RTP/RTCP network transmission ; Fifth, load balancing is a systematic project in the field of load balance. In the road link from the media resources to the user terminal, the load balancing should be considered. Based on the characteristics of SIP, use the SIP Proxy Server as the load balancing server to balance the broadcasting server’s load through the dial plan management of SIP Proxy server; Sixth, according to the characteristics of the broadcasting server, apply bit-stream and pipe cushioning mechanism to encoding and decoding.
Through the testing of the server prototype system, it has basically realized the streaming media broadcast service function, after the further optimizations based on this prototype system, it may have more practical use.
引文
[1] 韦乐平.电信网的发展与演进趋势[J].电信技术, 2001.
[2] J.Rosenberg,H.Schulzrinne,G..Camarillo,etc.SIP:Session Initiation Protocol [RFC3 261] [S]. June 2002.
[3] Daniel Collins. Carrier Grade Voice over IP (VoIP 技术与应用) [M].人民邮电出版社, April 2001.
[4] 司瑞峰,韩心慧,龙勤,潘爱民. SIP 标准中的核心技术与研究进展[J].软件学报, 2004.
[5] David Austerberry. The technology of Video&Audio streaming (second edition) [M], US: Focal Press, 2005.
[6] 叶永沛.网站流媒体视频的制作与应用[J] .中国有线电视, 2005.
[7] 刘炎芬.流媒体技术及其应用[J] .山西科技, 2005.
[8] 钟玉琢,白哲,沈洪.流媒体和视频服务器[M] .北京:清华大学出版社. 2003.
[9] http://www.ietf.org/rfc/rfc2327.txt
[10] H.Schulzrinne, S.Casner, R.Frederick and V.Jacobson .RTP:A Transport Protocol for Real-Time Applications. [RFC1889][S], January1996.
[11] H.Schulzrinne,S.Casner,R.Frederick,etc. RTP:A Transport Protocol for Real-Time Applications [RFC3550] [S]. July 2003.
[12] H.Schulzrinne,A.Rao,R.Lanphier.Real Time Streaming Protocol (RTSP) [RFC23 26] [S], April 1998.
[13] R.Braden, L.Zhang, S.Berson, S.Herzog and S.Jamin. Resource Reservation Protocol (RSVP) [RFC2205] [S], September 1997.
[14] 田毅珂,卢朝阳,高西全.MPEG-4 视频标准及其关键算法[J].现代电子技术,2002(1):38-41.
[15] 吴子辉,石旭利,张兆扬.H.264 编解码器方案综述[J].电视技术, 2005.
[16] 邢竞帆,李胜利. SIP 协议的研究[J].计算机应用, 2001.
[17] 陈建亚,余浩. 软交换与下一代网络[M].北京:北京邮电大学出版社,2003.
[18] 王华颖.信令协议 SIP 的研究与 SIPUA 的实现:(硕士学位论文) [M].天津:天津大学, 2005.
[19] R. Sparks. The Session Initiation Protocol (SIP) Refer Method [RFC3515] [S]. April 2003.
[20] D.Croeker, P.Overell.RFC2234 Augmented BNF for Syntax Speeifieations:ABNF, 1997.
[21] 沈鑫刻等.多媒体传输网络与 VOIP 系统设计[M].北京:人民邮电出版社, 2005.
[22] Berners-Lee, R.Fielding, L. Masinter. Uniform Resource Identifiers (URI): Generic Syntax [RFC2396] [S]. August 1998.
[23] Joachim Posegga, Jan Seedorf: Voice Over IP:Unsafe at any Bandwidth [EB/OL/]. http://www.informatik.uni-hamburg.de/SVS/papers/Eurescom_VoIP-unsafe.pdf. Sep 2005.
[24] Douglas E.Conmer, David L.Stevens. Internetworking With TCP/IP Vol.2: Design, Implementation and Internals Second Edition[M]. 北京:清华大学出版社, 1998.
[25] J.Rosenberg, H.Schulzrinne, R.Mahy, Ed. An INVITE Inititiated Dialog Event Package for the Session Initiation Protocol. draft-ietf-sipping-dialog-package-03, October 2003.
[26] 陈华林等著. SIP 协议中的媒体协商[J]. 广西通信技术, 2005. 8.
[27] http://www.scit.wlv.ac.uk/rfc/rfc30xx/RFC3016.html
[28] 孟凡蕊. 基于 RTP/RTCP 协议视频数据网络传输的实现[J]. 计算机工程与应用, 2004.
[29] 袁敏,夏勇.流媒体服务器技术与压缩编码方案[J].广科院互联网所. 2005.
[30] Stephan Wenger.H.264/AVC Over IP.IEEE Trans.on Circuits and Systems for Video Technology.2003,13:645～656.
[31] T.Stockhammer,M.Hannuksela,S.Wenger.H.26L/JVT Coding Network Abstraction Layer and IP-based Transport.IEEE ICIP,Rochester,NewYork,2002.
[32] Dawen Song,YiJun Mo, Furong Wang: Architecture of multiparty conferencing using SIP[Z]. US: Wireless Communications Page 1361- 1364 Sep 2005.
[33] 陈登伟,鲁智勇.网络动态负载均衡算法分析[J].现代电子技术, 2003 年第 21期.
[34] 罗坤,许先斌.安全高效的流媒体服务器集群系统[J].吉林大学学报(信息科学版) , 2006.
[35] Gonzalo.Camarillo 著.白建军,彭晖,田敏 译. SIP 揭密[M].北京:人民邮电出版社, 2003.
[36] Olivier.Hersent, David.Gurle, Jean-Pierre.Petit 著.邝坚,戴志涛 译. IP 电话—基于分组的多媒体通信系统[M].北京:人民邮电出版社, 2000.
[37] 覃少华,李子木等.一种新的基于代理缓存的流媒体分送策略[J].计算机工程, 2005.
[38] R.Zopf. Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN) [RFC3389][S]. September 2002.
[2] J.Rosenberg,H.Schulzrinne,G..Camarillo,etc.SIP:Session Initiation Protocol [RFC3 261] [S]. June 2002.
[3] Daniel Collins. Carrier Grade Voice over IP (VoIP 技术与应用) [M].人民邮电出版社, April 2001.
[4] 司瑞峰,韩心慧,龙勤,潘爱民. SIP 标准中的核心技术与研究进展[J].软件学报, 2004.
[5] David Austerberry. The technology of Video&Audio streaming (second edition) [M], US: Focal Press, 2005.
[6] 叶永沛.网站流媒体视频的制作与应用[J] .中国有线电视, 2005.
[7] 刘炎芬.流媒体技术及其应用[J] .山西科技, 2005.
[8] 钟玉琢,白哲,沈洪.流媒体和视频服务器[M] .北京:清华大学出版社. 2003.
[9] http://www.ietf.org/rfc/rfc2327.txt
[10] H.Schulzrinne, S.Casner, R.Frederick and V.Jacobson .RTP:A Transport Protocol for Real-Time Applications. [RFC1889][S], January1996.
[11] H.Schulzrinne,S.Casner,R.Frederick,etc. RTP:A Transport Protocol for Real-Time Applications [RFC3550] [S]. July 2003.
[12] H.Schulzrinne,A.Rao,R.Lanphier.Real Time Streaming Protocol (RTSP) [RFC23 26] [S], April 1998.
[13] R.Braden, L.Zhang, S.Berson, S.Herzog and S.Jamin. Resource Reservation Protocol (RSVP) [RFC2205] [S], September 1997.
[14] 田毅珂,卢朝阳,高西全.MPEG-4 视频标准及其关键算法[J].现代电子技术,2002(1):38-41.
[15] 吴子辉,石旭利,张兆扬.H.264 编解码器方案综述[J].电视技术, 2005.
[16] 邢竞帆,李胜利. SIP 协议的研究[J].计算机应用, 2001.
[17] 陈建亚,余浩. 软交换与下一代网络[M].北京:北京邮电大学出版社,2003.
[18] 王华颖.信令协议 SIP 的研究与 SIPUA 的实现:(硕士学位论文) [M].天津:天津大学, 2005.
[19] R. Sparks. The Session Initiation Protocol (SIP) Refer Method [RFC3515] [S]. April 2003.
[20] D.Croeker, P.Overell.RFC2234 Augmented BNF for Syntax Speeifieations:ABNF, 1997.
[21] 沈鑫刻等.多媒体传输网络与 VOIP 系统设计[M].北京:人民邮电出版社, 2005.
[22] Berners-Lee, R.Fielding, L. Masinter. Uniform Resource Identifiers (URI): Generic Syntax [RFC2396] [S]. August 1998.
[23] Joachim Posegga, Jan Seedorf: Voice Over IP:Unsafe at any Bandwidth [EB/OL/]. http://www.informatik.uni-hamburg.de/SVS/papers/Eurescom_VoIP-unsafe.pdf. Sep 2005.
[24] Douglas E.Conmer, David L.Stevens. Internetworking With TCP/IP Vol.2: Design, Implementation and Internals Second Edition[M]. 北京:清华大学出版社, 1998.
[25] J.Rosenberg, H.Schulzrinne, R.Mahy, Ed. An INVITE Inititiated Dialog Event Package for the Session Initiation Protocol. draft-ietf-sipping-dialog-package-03, October 2003.
[26] 陈华林等著. SIP 协议中的媒体协商[J]. 广西通信技术, 2005. 8.
[27] http://www.scit.wlv.ac.uk/rfc/rfc30xx/RFC3016.html
[28] 孟凡蕊. 基于 RTP/RTCP 协议视频数据网络传输的实现[J]. 计算机工程与应用, 2004.
[29] 袁敏,夏勇.流媒体服务器技术与压缩编码方案[J].广科院互联网所. 2005.
[30] Stephan Wenger.H.264/AVC Over IP.IEEE Trans.on Circuits and Systems for Video Technology.2003,13:645～656.
[31] T.Stockhammer,M.Hannuksela,S.Wenger.H.26L/JVT Coding Network Abstraction Layer and IP-based Transport.IEEE ICIP,Rochester,NewYork,2002.
[32] Dawen Song,YiJun Mo, Furong Wang: Architecture of multiparty conferencing using SIP[Z]. US: Wireless Communications Page 1361- 1364 Sep 2005.
[33] 陈登伟,鲁智勇.网络动态负载均衡算法分析[J].现代电子技术, 2003 年第 21期.
[34] 罗坤,许先斌.安全高效的流媒体服务器集群系统[J].吉林大学学报(信息科学版) , 2006.
[35] Gonzalo.Camarillo 著.白建军,彭晖,田敏 译. SIP 揭密[M].北京:人民邮电出版社, 2003.
[36] Olivier.Hersent, David.Gurle, Jean-Pierre.Petit 著.邝坚,戴志涛 译. IP 电话—基于分组的多媒体通信系统[M].北京:人民邮电出版社, 2000.
[37] 覃少华,李子木等.一种新的基于代理缓存的流媒体分送策略[J].计算机工程, 2005.
[38] R.Zopf. Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN) [RFC3389][S]. September 2002.