基于SIP协议的VoIP网络研究及终端实现
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摘要
VoIP(Voice over IP)是一种利用IP网络作为传输载体实现计算机——计算机、普通电话——普通电话以及计算机——普通电话之间实时语音通信，并提供相应的增值业务的技术。
论文通过对现有的两种信令控制协议——H．323协议和会话初始化协议(SIP)进行分析比较，提出了在局域网VoIP系统中更适合采用会话初始化协议的观点。并设计实现了具有SIP用户代理功能的VoIP终端设备，利用该设备在局域网中构建VoIP实验系统，从工程上探讨和验证VoIP网络理论及实时传输协议对于语音数据传输的保障作用。同时也验证了SIP协议在建立呼叫及保证会话中的有效性。论文也对VoIP网络结构、VoIP协议层、VoIP系统设备、软件、协议等要素进行了广泛的讨论，并根据下一代IP网络融合的概念，提出了一种语音与数据集成的网络模型。
此外，论文对于实时语音数据网络的服务质量QoS保障进行了阐述，对于正在发展的IP网络QoS保障协议包括资源预留协议、区分服务模式和多标签交换协议进行了探讨，并结合语音的IP网络特点，提出在整个系统中实现端到端的有保障QoS的实现方式。
Voice over IP(VoIP) is the technology which uses the IP network as the carrier of real time vocal dataflow, to make a communication connection between computers, traditional telephones and both of them. It can also provide the value increment service based on the basic structure of VoIP.
In this thesis, research has been carried on in several aspects such as network structure of VoIP, protocol layers of VoIP, signaling control protocols and the LAN VoIP system based on SIP. And then, the VoIP terminal device is designed and tested in the environment of the experimental LAN VoIP system.Research has been focused on the two different signaling protocols: H.323 and SIP, and then a solution based on SIP is provided, and proved to be effective in the experimental LAN VoIP system.
Also the very important factor Quality of Service(QoS) is investigated, the different QoS supports in the IP packet switch network are given and analyzed, including Intr-Serv mode and related RSVP and Diff-Serv QoS mode. The architecture to guarantee the end-to-end QoS over the different protocols and different network domains in transmitting the real time vocal data flow is provided. Then the synergic solution to provide the better QoS over the currently Best-Effort network is given and evaluated.
论文通过对现有的两种信令控制协议——H．323协议和会话初始化协议(SIP)进行分析比较，提出了在局域网VoIP系统中更适合采用会话初始化协议的观点。并设计实现了具有SIP用户代理功能的VoIP终端设备，利用该设备在局域网中构建VoIP实验系统，从工程上探讨和验证VoIP网络理论及实时传输协议对于语音数据传输的保障作用。同时也验证了SIP协议在建立呼叫及保证会话中的有效性。论文也对VoIP网络结构、VoIP协议层、VoIP系统设备、软件、协议等要素进行了广泛的讨论，并根据下一代IP网络融合的概念，提出了一种语音与数据集成的网络模型。
此外，论文对于实时语音数据网络的服务质量QoS保障进行了阐述，对于正在发展的IP网络QoS保障协议包括资源预留协议、区分服务模式和多标签交换协议进行了探讨，并结合语音的IP网络特点，提出在整个系统中实现端到端的有保障QoS的实现方式。
Voice over IP(VoIP) is the technology which uses the IP network as the carrier of real time vocal dataflow, to make a communication connection between computers, traditional telephones and both of them. It can also provide the value increment service based on the basic structure of VoIP.
In this thesis, research has been carried on in several aspects such as network structure of VoIP, protocol layers of VoIP, signaling control protocols and the LAN VoIP system based on SIP. And then, the VoIP terminal device is designed and tested in the environment of the experimental LAN VoIP system.Research has been focused on the two different signaling protocols: H.323 and SIP, and then a solution based on SIP is provided, and proved to be effective in the experimental LAN VoIP system.
Also the very important factor Quality of Service(QoS) is investigated, the different QoS supports in the IP packet switch network are given and analyzed, including Intr-Serv mode and related RSVP and Diff-Serv QoS mode. The architecture to guarantee the end-to-end QoS over the different protocols and different network domains in transmitting the real time vocal data flow is provided. Then the synergic solution to provide the better QoS over the currently Best-Effort network is given and evaluated.
引文
[1] Daniel Collins, "Carrier Grade Voice over IP", the People's Posts and Telecommunications Publishing House, Beijing 2001．
[2] 叶华等，“IP电话／传真技术”，人民邮电出版社，北京，2000．
[3] Postel, J, "Internet Protocol", RFC 791, September 1981．
[4] Postel, J, "Transmission Control Protocol", RFC 793, September 1981．
[5] Postel, J, "User Datagram Protocol", RFC 768, August, 1980．
[6] Schulzrinne, H. et al, "RTP: A Transport Protocol for Real-Time Applications", RFC1889, January 1996．
[7] Schulzrinne, H, "RTP Profile for Audio and Video Conferences with Minimal Control", RFC1890, January 1996．
[8] ITU-T, "Pulse code modulation (PCM) of voice frequencies", Recommendation G.711, August 1988．
[9] ITU-T, "Dual rate speech code for multimedia communications transmitting at 5． 3 and 6． 3 kbps", Recommendation G.723． 1, March 1996．
[10] ITU-T, "Coding of speech at 8 kbps using Conjugate-Structure Algebraic-Code-Excited Linear Prediction(CS-ACELP)", Recommendation G.729, March 1996．
[11] Jerry,D,Gibson,“多媒体数字压缩原理与标准”，电子工业出版社，北京， 2000．
[12] ITU-T, "Packet-based multimedia communications systems", Recommendation H.323, February 1998．
[13] Internet Engineering Task Force, Resource Reservation Protocol, RFC 2205,1997．
[14] Internet Engineering Task Force, An Architecture for Differentiated Services, RFC2475, 1999．
[15] Jacobson, V., Nichols, K. and Poduri, K. "An Expedited Forwarding PHB", RFC 2598, June 1999．
[16] Wroclawski, J, "The Use of RSVP with IETF Integrated Services", RFC 2210, September 1997．
[17] Shenker, S. and Partridge, C. and Guerin, R, "Specification of Guaranteed Quality of Service", RFC 2212, September 1997．
[18] Awduche, D, "Requirements for Traffic Engineering Over MPLS", RFC 2702, September 1999．
[19] Handley, M. et al, "SIP: Session Initiation Protocol", RFC 2543, March 1999．
[20] Handley, M. and Jacobson, V, "SDP: Session Description Protocol", RFC 2327, April 1998．
[21] Schulzrinne, H. et al, "Real Time Streaming Protocol(RTSP)", RFC 2326, April 1998.
[22] Singh, K. and Schulzrinne, H, "Interworking Between SIP/SDP and H.323", work in progress.
[23] Susan Love, Joe Nowalk, Nancy Weiss, Keri Then "A Case Study in Campus VoIP Deployment: Hype vs. Reality", 1999. http://198.11.21.25/capstoneTest/Students/Papers/docs/VOIPcampus38173.pdf.
[24] 黄伟峰，"SIP——构筑下一代SoftSwitch的协议"，计算机世界网，2001．2．
[25] 何畏，虞水俊，李俊，"局域网IP电话系统信令控制协议研究"，计算机世界网，2002．1．
[26] 蔡皖东，"计算机网络"，西安电子科技大学出版社，西安，2000．
[27] Scott Keagy, "Integrating Voice and Data Networks", Publishing House of Electronics Industry, Beijing, 2002.
[28] Robert, P. et al, "Cisco AWID and IP telephony Design and Implementation", Publishing House of Electronics Industry, Beijing, 2002.
[29] Nathan, J, Muller, "IP Convergence: The Next Revolution in Telecommunications", Publishing House of Electronics Industry, Beijing. 2001.
[30] 吴伟，"下一代IP网络技术保障——多协议标签交换"，清华大学出版社，北京，2002．
[31] 侯自强，"宽带IP技术发展"，人民邮电出版社，北京，2001．
[32] 张雄伟曹铁勇，"DSP芯片的原理与开发应用"，电子工业出版社，2000．9．
[33] 侯伯亨等，"VHDL硬件描述语言与数字逻辑'，西安电子科技大学出版社，西安，1997。
[34] Ti, "TMS320c20x User's Guide", Texas Instruments, 1999.
[35] Uyless Black, "Voice Over IP", China Machine Press, Beijing, 1997.
[36] 张中山，隆克平，程时端，"MPLS——支持IP组播中标签分配的新机制"，北京邮电大学学报，2001．3。
[37] Bill Douskalis，"IP电话技术稳定的VOIP服务集成"，机械工业出版社，北京，2001．
[38] 熬志刚，"现代高速交换局域网及其应用"，国防工业出版社，北京，2001．
[39] Uyless Black，"现代通信最新技术"，清华大学出版社，北京，2000．
[2] 叶华等，“IP电话／传真技术”，人民邮电出版社，北京，2000．
[3] Postel, J, "Internet Protocol", RFC 791, September 1981．
[4] Postel, J, "Transmission Control Protocol", RFC 793, September 1981．
[5] Postel, J, "User Datagram Protocol", RFC 768, August, 1980．
[6] Schulzrinne, H. et al, "RTP: A Transport Protocol for Real-Time Applications", RFC1889, January 1996．
[7] Schulzrinne, H, "RTP Profile for Audio and Video Conferences with Minimal Control", RFC1890, January 1996．
[8] ITU-T, "Pulse code modulation (PCM) of voice frequencies", Recommendation G.711, August 1988．
[9] ITU-T, "Dual rate speech code for multimedia communications transmitting at 5． 3 and 6． 3 kbps", Recommendation G.723． 1, March 1996．
[10] ITU-T, "Coding of speech at 8 kbps using Conjugate-Structure Algebraic-Code-Excited Linear Prediction(CS-ACELP)", Recommendation G.729, March 1996．
[11] Jerry,D,Gibson,“多媒体数字压缩原理与标准”，电子工业出版社，北京， 2000．
[12] ITU-T, "Packet-based multimedia communications systems", Recommendation H.323, February 1998．
[13] Internet Engineering Task Force, Resource Reservation Protocol, RFC 2205,1997．
[14] Internet Engineering Task Force, An Architecture for Differentiated Services, RFC2475, 1999．
[15] Jacobson, V., Nichols, K. and Poduri, K. "An Expedited Forwarding PHB", RFC 2598, June 1999．
[16] Wroclawski, J, "The Use of RSVP with IETF Integrated Services", RFC 2210, September 1997．
[17] Shenker, S. and Partridge, C. and Guerin, R, "Specification of Guaranteed Quality of Service", RFC 2212, September 1997．
[18] Awduche, D, "Requirements for Traffic Engineering Over MPLS", RFC 2702, September 1999．
[19] Handley, M. et al, "SIP: Session Initiation Protocol", RFC 2543, March 1999．
[20] Handley, M. and Jacobson, V, "SDP: Session Description Protocol", RFC 2327, April 1998．
[21] Schulzrinne, H. et al, "Real Time Streaming Protocol(RTSP)", RFC 2326, April 1998.
[22] Singh, K. and Schulzrinne, H, "Interworking Between SIP/SDP and H.323", work in progress.
[23] Susan Love, Joe Nowalk, Nancy Weiss, Keri Then "A Case Study in Campus VoIP Deployment: Hype vs. Reality", 1999. http://198.11.21.25/capstoneTest/Students/Papers/docs/VOIPcampus38173.pdf.
[24] 黄伟峰，"SIP——构筑下一代SoftSwitch的协议"，计算机世界网，2001．2．
[25] 何畏，虞水俊，李俊，"局域网IP电话系统信令控制协议研究"，计算机世界网，2002．1．
[26] 蔡皖东，"计算机网络"，西安电子科技大学出版社，西安，2000．
[27] Scott Keagy, "Integrating Voice and Data Networks", Publishing House of Electronics Industry, Beijing, 2002.
[28] Robert, P. et al, "Cisco AWID and IP telephony Design and Implementation", Publishing House of Electronics Industry, Beijing, 2002.
[29] Nathan, J, Muller, "IP Convergence: The Next Revolution in Telecommunications", Publishing House of Electronics Industry, Beijing. 2001.
[30] 吴伟，"下一代IP网络技术保障——多协议标签交换"，清华大学出版社，北京，2002．
[31] 侯自强，"宽带IP技术发展"，人民邮电出版社，北京，2001．
[32] 张雄伟曹铁勇，"DSP芯片的原理与开发应用"，电子工业出版社，2000．9．
[33] 侯伯亨等，"VHDL硬件描述语言与数字逻辑'，西安电子科技大学出版社，西安，1997。
[34] Ti, "TMS320c20x User's Guide", Texas Instruments, 1999.
[35] Uyless Black, "Voice Over IP", China Machine Press, Beijing, 1997.
[36] 张中山，隆克平，程时端，"MPLS——支持IP组播中标签分配的新机制"，北京邮电大学学报，2001．3。
[37] Bill Douskalis，"IP电话技术稳定的VOIP服务集成"，机械工业出版社，北京，2001．
[38] 熬志刚，"现代高速交换局域网及其应用"，国防工业出版社，北京，2001．
[39] Uyless Black，"现代通信最新技术"，清华大学出版社，北京，2000．