基于SIP的移动VoIP的研究与实现
摘要
随着无线网络技术的快速发展,移动设备功能的不断强大,传统的多媒体业务已不能满足人们的需求。如何在现有的异构网络环境之上发展更多的移动多媒体增值业务,成了当前研究的热门。其中典型的应用为VoIP技术。实现VoIP的一个关键的技术就是信令技术,目前广泛应用的有两种信令体系：H.323和SIP。H.323功能强大但却非常复杂,它规定了完整多媒体应用架构,要求较高,实现起来比较困难。SIP协议是一个简单的、易扩展和实现的,基于文本的信令控制协议。更重要的是,它是一个底层接入无关的应用层信令协议。基于以上原因,研究SIP对移动VoIP的支持与应用,对发展接入无关的移动多媒体应用有着重要的意义和价值。
本文设计实现了基于SIP的网络服务组件,通过该系统基于SIP的移动VoIP用户可以很简单的进行定位和建立多媒体会话,进而完成语音交互过程。本系统遵循RFC3261规范,并基于MJSIP协议栈,对SIP协议栈进行了重新设计和改进,实现了包括：传输层、事务层和事务用户层。然后基于此协议栈设计实现了包括代理服务器、注册服务器和重定向服务器等SIP网元,构建了一个基于SIP的网络融合平台。同时,本文又对新兴的开源手机操作系统Android进行了研究,将SIP协议栈成功应用于Android平台上,实现了模拟终端到服务器的通信。
在完成以上功能的同时,本文重点研究了SIP对移动性的支持,提出了一种动态分层注册模型,用于实现移动设备域内移动时切换延迟问题。由于SIP协议本身缺少安全机制保护,故在SIP应用中,安全性问题也是考虑的重点,本文也对SIP的安全性进行了相关研究,并实现了Http摘要认证方案。
本系统在模拟网络环境上进行了多次测试,验证了SIP服务器的正确性和稳定性。本文通过构建的SIP网络,验证了SIP移动多媒体应用的可行性和方便性,下一阶段,本项目将重点研究基于SIP网络的多媒体应用系统,譬如：多方会议系统等。
With the rapid development of wireless network technology and the continually strong mobile device features, the traditional media businesses have been unable to meet people's needs. How to develop the mobile media with more value-added services in the existing heterogeneous network environment has become the most popular of current research. One of the most typical applications is VoIP technology. A key realization achieving VoIP technology is the signaling technology. There are two wildly used signaling systems:H.323 and SIP. H.323 is powerful but complex. It provides a complete framework with higher requirement and is achieved more difficultly. SIP protocol is a signaling control protocol which is simply expensing and achieving, based on text. More important is that it is unrelated to the low-level access. Above the reasons, the research based on the support and application of SIP for mobile VoIP has more important significance and value for developing access unrelated mobile multimedia applications.
This thesis design enables network service components of SIP. Through this system, mobile VoIP users based on SIP are easily able to locate and establish multi-media session, then complete the voice interaction. Follows RFC3261 specification, this system redesigns the SIP protocol stack based on the MJSIP protocol stack and achieves transport layer, transaction layer and transaction user layer. It achieves SIP element location, including proxy servers, registration servers and redirect servers and builds a SIP-based platform for network convergence. Meanwhile, this paper has researched on emerging open-source Android mobile operating system which uses the SIP protocol stack on the Android platform to achieve a simulated end-to-server communication.
While completing the above functions, this thesis focuses on the support of SIP for mobility and raises a dynamic hierarchical registration model, used to implement the switching delays within the domain for mobile device.Because the SIP protocol is lack of security mechanisms, in the SIP applications, security issues are valued to concern. This thesis also carried out the safety of SIP-related research and realized Http summary of certification program.
This system carries out a number of tests to verify the correctness and stability of the SIP server in a simulated network environment. This article only provides the basic functions of SIP network and verifies feasibility and conveniences of SIP mobile multimedia. Next stage, we will focus on web development SIP-based multimedia applications, such as:multi-party conferencing systems.
本文设计实现了基于SIP的网络服务组件,通过该系统基于SIP的移动VoIP用户可以很简单的进行定位和建立多媒体会话,进而完成语音交互过程。本系统遵循RFC3261规范,并基于MJSIP协议栈,对SIP协议栈进行了重新设计和改进,实现了包括：传输层、事务层和事务用户层。然后基于此协议栈设计实现了包括代理服务器、注册服务器和重定向服务器等SIP网元,构建了一个基于SIP的网络融合平台。同时,本文又对新兴的开源手机操作系统Android进行了研究,将SIP协议栈成功应用于Android平台上,实现了模拟终端到服务器的通信。
在完成以上功能的同时,本文重点研究了SIP对移动性的支持,提出了一种动态分层注册模型,用于实现移动设备域内移动时切换延迟问题。由于SIP协议本身缺少安全机制保护,故在SIP应用中,安全性问题也是考虑的重点,本文也对SIP的安全性进行了相关研究,并实现了Http摘要认证方案。
本系统在模拟网络环境上进行了多次测试,验证了SIP服务器的正确性和稳定性。本文通过构建的SIP网络,验证了SIP移动多媒体应用的可行性和方便性,下一阶段,本项目将重点研究基于SIP网络的多媒体应用系统,譬如：多方会议系统等。
With the rapid development of wireless network technology and the continually strong mobile device features, the traditional media businesses have been unable to meet people's needs. How to develop the mobile media with more value-added services in the existing heterogeneous network environment has become the most popular of current research. One of the most typical applications is VoIP technology. A key realization achieving VoIP technology is the signaling technology. There are two wildly used signaling systems:H.323 and SIP. H.323 is powerful but complex. It provides a complete framework with higher requirement and is achieved more difficultly. SIP protocol is a signaling control protocol which is simply expensing and achieving, based on text. More important is that it is unrelated to the low-level access. Above the reasons, the research based on the support and application of SIP for mobile VoIP has more important significance and value for developing access unrelated mobile multimedia applications.
This thesis design enables network service components of SIP. Through this system, mobile VoIP users based on SIP are easily able to locate and establish multi-media session, then complete the voice interaction. Follows RFC3261 specification, this system redesigns the SIP protocol stack based on the MJSIP protocol stack and achieves transport layer, transaction layer and transaction user layer. It achieves SIP element location, including proxy servers, registration servers and redirect servers and builds a SIP-based platform for network convergence. Meanwhile, this paper has researched on emerging open-source Android mobile operating system which uses the SIP protocol stack on the Android platform to achieve a simulated end-to-server communication.
While completing the above functions, this thesis focuses on the support of SIP for mobility and raises a dynamic hierarchical registration model, used to implement the switching delays within the domain for mobile device.Because the SIP protocol is lack of security mechanisms, in the SIP applications, security issues are valued to concern. This thesis also carried out the safety of SIP-related research and realized Http summary of certification program.
This system carries out a number of tests to verify the correctness and stability of the SIP server in a simulated network environment. This article only provides the basic functions of SIP network and verifies feasibility and conveniences of SIP mobile multimedia. Next stage, we will focus on web development SIP-based multimedia applications, such as:multi-party conferencing systems.
引文
[1]IETF.SIP:RFC 3261.2002
[2]Kin C,Choi B.Design and implement of the VoIP supporting the differentiated call processing.ICO IN 2002.
[3]陈建亚等.软交换与下一代网络.北京邮电大学出版社.2003
[4]桂海源等.IP电话技术与软交换.北京邮电大学出版社.2004
[5]周海华等.SIP原理与应用.机械工业出版社.2006
[6]张智江等.SIP协议及其应用.电子工业出版社.2005
[7]Georg Mayer,Hisham Khartabil等.IMS:IP多媒体概念和服务.机械工业出版社.2007
[8]IETF.RTCP:RFC 3605,Real Time Control Protocol attribute in Session Description Protocol.2003
[9]Hierarchical Mobility Management for VoIP Traffic. Archan Misra Subir Das Anthony J McAuley Telcordia Technologies,IEEE 2001
[10]3GPP TS23.228V5.6.0.IP Multimedia Subsystem (IMS):Stage2.2002
[11]IETF.SDP:RFC 2327, Session Description Protocol.1998
[12]Aytunc Durlanik.et, Ibrahim Sogukpinar. SIP Authentication Scheme using ECDH.2005
[13]Crocker, D. and P. Overell, "Augmented BNF for Syntax Specifications:ABNF", RFC 5234, 2008.
[14]徐鹏等.SIP协议控制多媒体会话的研究与应用.计算机工程.2006：196-198
[15]沈鑫剡,多媒体传输网络与VoIP系统设计.人民邮电出版社.2005
[16]Mohammad Sabbir Alam.et, Articulated Narrowcasting for Privacy and Awareness in Multimedia Conferencing Systems and Design for Implementation within a SIP Framework, Virtual Reality and Broadcasting.2008
[17]V. Miroslav et al., Performance Comparison of Secure and Insecure VoIP environments, Proc. KTTO'08,2008.
[18]S. McGann et al., "An Analysis of Security Threats and Tools in SIP-Based VoIP Systems," in Second VoIP Security Workshop,2005.
[19]樊自甫等.基于S/MIME的SIP安全性方案.计算机工程.2009
[20]孟岩.Android组件模型评析.程序员.2008
[21]余厚全等.SIP协议栈中消息收发的研究.光通信研究.2004
[22]A.Dutta, B.Kim, T.Zhang, S.Baba, K.Taniuchi, Y.Ohba, H.Schulzrinne "Experimental Analysis of Multi Interface Mobility Management with SIP and MIP", IEEE Wirelesscom 2005.
[23]Jennings, C. and R. Mahy, "Managing Client Initiated Connections in the Session Initiation Protocol (SIP)", draft-ietf-sip-outbound-15.txt (work in progress),2008
[24]Tonesi, D.S.;Salgarelli, L.; Yan Sun; La Porta, T.F.Evaluation of Signaling Loads in 3GPP et works,EEE Wireless Communications.2008
[25]Wenchao Mad Yuguang Fang Dynamic Hierarchical Mobility Management Strategy for Mobile IP Networks. IEEE 2004
[26]IETF.RTP:RFC 3550, a Transport Protocol for Real-Time Applications.2003
[27]赵臣兵,刘立柱.基于RTP协议的视频实时采集与传输的研究.微计算机信息.2006第22卷6-1期
[28]S. Ehlert, G. Zhang, D. Geneiatakis,G.Kambourakis, T. Dagiuklas,J. Markl, and D. Sisalem, "Two layer Denial of Service prevention on SIP VoIP infrastructures," Computer Communications,2008.
[29]D. Yon and G Camarillo, "TCP-Based Media Transport in the Session Description Protocol (SDP)", IETF RFC 4145, September 2005.
[30]Stefano Salsano,et,Seamless vertical handover of VoIP calls based on SIP Session Border Controllers,IEEE 2006.
[31]D. Simeonidou et al., "Dynamic Optical-Network Architectures and Technologies for Existing and Emerging Grid Services,"IEEE J. Lightwave Tech., vol.23, no.10,2005
[32]E. Nahum et al., "Evaluating SIP Proxy Server Performance," in Proc.NOSSDAV'07,2007.
[33]J. Humberto, R. State, and et al, "KiF:A stateful SIP Fuzzer," in Proc of IPTComm-Principles, Systems and Applications of IP Telecommuni-cations, New York, USA,2007.
[34]D. Sisalem, J. Kuthan, and et al, "DoS Attacks Targeting a SIP VoIP Infrastructure:Attack Scenarios and Prevention Mechanisms," IEEENetwork, vol.20, no.5, pp.26-31,2006.
[35]C. Pavlovski, "Service Delivery Platforms in Practice," Comm. Mag.,IEEE, vol.45, no.3, pp. 114-121,2007.
[36]Yacine Rebahi, Jordi Jaen Pallares et.al.,Performance Analysis of Identity Management in the Session Initiation protocol (SIP). The sixth AICCSA-08 Conference,2008.
[37]Ge Zhang, Towards Secure SIP Signalling Service for VoIP applications, Karlstad University Studies.2009
[38]李婧,基于SIP的安全认证机制的研究及改进.计算机工程.2009
[39]3GPP, "3GPP System Architecture Evolution (SAE):Report on Technical Options and Conclusions," TR23.882 v1.9.0, Apr.2007.
[40]F. Persson, "Voice over IP Realized for the 3GPP Long Term Evolution," IEEE VTC Fall'07, Sept.2007, pp.1436-40.
[41]H. Zhai, J. Wang, and Y.Fang, "Providing Statistical QOS Guarantee for Voice over IP in the IEEE 802.11 Wireless LANs," IEEE Wireless Commun, vol.13,no.1,Feb.2006, pp.36-43
[42]朱亚楠等.高可移植SIP终端的设计与实现.计算应用与软件.2009
[43]Emanuel B. Rodrigues and Francisco Rodrigo P. Cavalcanti, QOS-Driven Adaptive Congestion Control for Voice over IP in Multiservice Wireless Cellular Networks.IEEE 2008
[44]Hemanth Srinivasan and Kamil Sarac, A SIP Security Testing Framework.IEEE 2009
[45]J.Rosenberg,H.Schulzrinne,RFC 3264,An Offer/Answer Model with the Session Description Protocol,2002
[46]Michael Juntao Yuan,Kevin Sharp.Series 40可扩展应用程序开发.人民邮电出版社.2006
[47]徐晓宇,张惠民.SIP会话协议在第三代移动网络中关键问题研究.数据通信.2004
[2]Kin C,Choi B.Design and implement of the VoIP supporting the differentiated call processing.ICO IN 2002.
[3]陈建亚等.软交换与下一代网络.北京邮电大学出版社.2003
[4]桂海源等.IP电话技术与软交换.北京邮电大学出版社.2004
[5]周海华等.SIP原理与应用.机械工业出版社.2006
[6]张智江等.SIP协议及其应用.电子工业出版社.2005
[7]Georg Mayer,Hisham Khartabil等.IMS:IP多媒体概念和服务.机械工业出版社.2007
[8]IETF.RTCP:RFC 3605,Real Time Control Protocol attribute in Session Description Protocol.2003
[9]Hierarchical Mobility Management for VoIP Traffic. Archan Misra Subir Das Anthony J McAuley Telcordia Technologies,IEEE 2001
[10]3GPP TS23.228V5.6.0.IP Multimedia Subsystem (IMS):Stage2.2002
[11]IETF.SDP:RFC 2327, Session Description Protocol.1998
[12]Aytunc Durlanik.et, Ibrahim Sogukpinar. SIP Authentication Scheme using ECDH.2005
[13]Crocker, D. and P. Overell, "Augmented BNF for Syntax Specifications:ABNF", RFC 5234, 2008.
[14]徐鹏等.SIP协议控制多媒体会话的研究与应用.计算机工程.2006：196-198
[15]沈鑫剡,多媒体传输网络与VoIP系统设计.人民邮电出版社.2005
[16]Mohammad Sabbir Alam.et, Articulated Narrowcasting for Privacy and Awareness in Multimedia Conferencing Systems and Design for Implementation within a SIP Framework, Virtual Reality and Broadcasting.2008
[17]V. Miroslav et al., Performance Comparison of Secure and Insecure VoIP environments, Proc. KTTO'08,2008.
[18]S. McGann et al., "An Analysis of Security Threats and Tools in SIP-Based VoIP Systems," in Second VoIP Security Workshop,2005.
[19]樊自甫等.基于S/MIME的SIP安全性方案.计算机工程.2009
[20]孟岩.Android组件模型评析.程序员.2008
[21]余厚全等.SIP协议栈中消息收发的研究.光通信研究.2004
[22]A.Dutta, B.Kim, T.Zhang, S.Baba, K.Taniuchi, Y.Ohba, H.Schulzrinne "Experimental Analysis of Multi Interface Mobility Management with SIP and MIP", IEEE Wirelesscom 2005.
[23]Jennings, C. and R. Mahy, "Managing Client Initiated Connections in the Session Initiation Protocol (SIP)", draft-ietf-sip-outbound-15.txt (work in progress),2008
[24]Tonesi, D.S.;Salgarelli, L.; Yan Sun; La Porta, T.F.Evaluation of Signaling Loads in 3GPP et works,EEE Wireless Communications.2008
[25]Wenchao Mad Yuguang Fang Dynamic Hierarchical Mobility Management Strategy for Mobile IP Networks. IEEE 2004
[26]IETF.RTP:RFC 3550, a Transport Protocol for Real-Time Applications.2003
[27]赵臣兵,刘立柱.基于RTP协议的视频实时采集与传输的研究.微计算机信息.2006第22卷6-1期
[28]S. Ehlert, G. Zhang, D. Geneiatakis,G.Kambourakis, T. Dagiuklas,J. Markl, and D. Sisalem, "Two layer Denial of Service prevention on SIP VoIP infrastructures," Computer Communications,2008.
[29]D. Yon and G Camarillo, "TCP-Based Media Transport in the Session Description Protocol (SDP)", IETF RFC 4145, September 2005.
[30]Stefano Salsano,et,Seamless vertical handover of VoIP calls based on SIP Session Border Controllers,IEEE 2006.
[31]D. Simeonidou et al., "Dynamic Optical-Network Architectures and Technologies for Existing and Emerging Grid Services,"IEEE J. Lightwave Tech., vol.23, no.10,2005
[32]E. Nahum et al., "Evaluating SIP Proxy Server Performance," in Proc.NOSSDAV'07,2007.
[33]J. Humberto, R. State, and et al, "KiF:A stateful SIP Fuzzer," in Proc of IPTComm-Principles, Systems and Applications of IP Telecommuni-cations, New York, USA,2007.
[34]D. Sisalem, J. Kuthan, and et al, "DoS Attacks Targeting a SIP VoIP Infrastructure:Attack Scenarios and Prevention Mechanisms," IEEENetwork, vol.20, no.5, pp.26-31,2006.
[35]C. Pavlovski, "Service Delivery Platforms in Practice," Comm. Mag.,IEEE, vol.45, no.3, pp. 114-121,2007.
[36]Yacine Rebahi, Jordi Jaen Pallares et.al.,Performance Analysis of Identity Management in the Session Initiation protocol (SIP). The sixth AICCSA-08 Conference,2008.
[37]Ge Zhang, Towards Secure SIP Signalling Service for VoIP applications, Karlstad University Studies.2009
[38]李婧,基于SIP的安全认证机制的研究及改进.计算机工程.2009
[39]3GPP, "3GPP System Architecture Evolution (SAE):Report on Technical Options and Conclusions," TR23.882 v1.9.0, Apr.2007.
[40]F. Persson, "Voice over IP Realized for the 3GPP Long Term Evolution," IEEE VTC Fall'07, Sept.2007, pp.1436-40.
[41]H. Zhai, J. Wang, and Y.Fang, "Providing Statistical QOS Guarantee for Voice over IP in the IEEE 802.11 Wireless LANs," IEEE Wireless Commun, vol.13,no.1,Feb.2006, pp.36-43
[42]朱亚楠等.高可移植SIP终端的设计与实现.计算应用与软件.2009
[43]Emanuel B. Rodrigues and Francisco Rodrigo P. Cavalcanti, QOS-Driven Adaptive Congestion Control for Voice over IP in Multiservice Wireless Cellular Networks.IEEE 2008
[44]Hemanth Srinivasan and Kamil Sarac, A SIP Security Testing Framework.IEEE 2009
[45]J.Rosenberg,H.Schulzrinne,RFC 3264,An Offer/Answer Model with the Session Description Protocol,2002
[46]Michael Juntao Yuan,Kevin Sharp.Series 40可扩展应用程序开发.人民邮电出版社.2006
[47]徐晓宇,张惠民.SIP会话协议在第三代移动网络中关键问题研究.数据通信.2004