基于SIP的嵌入式VoIP终端的设计与实现
摘要
近几年,随IP语音通信技术和宽带业务的发展,VoIP电话技术已逐步取代传统PSTN(Public Switch Telephone Network公共电话交换网络)电话,成为语音通信技术发展的新的趋势,并且其形态也由最初的基与PC的应用软件发展为基于IP网络的嵌入式SoC(System on Chip片上系统)智能终端。
SIP(Session Initiation Protocol会话初始协议)是由IETF(Internet Engineering Task Force英特网工程任务组)提出的一个应用层协议,用于创建、修改和终止一个或多个参与者参加的会话。该协议通过借鉴HTTP和SMTP协议风格及其成功经验,继承了HTTP协议的一些头域、编码规则、协议流程和认证机制,以其简单、开放、灵活、可扩展等显著优点,成为为业界所广泛接受,应用最为广泛的VoIP信令协议。
在分析了IP语音通信技术的基本原理、关键技术和发展现状之后,着重讨论了语音通信通常用到的一些重要协议和技术,包括用于信令部分的SIP协议,用于媒体参数协商的SDP协议,用于语音数据实时传输的RTP协议等。然后从整体上介绍了本终端的系统结构和设计过程,最终选用Broadcom 1100系列语音通信专用SoC平台,基与VxWorks实时操作系统,采用消息队列机制控制系统事务状态机运作与转换予以实现。
经过测试,终端实现了RFC3261所规定的基本要求,具有稳定、操作简单等优点,并有良好的互通性。
With the development of IP-based telecommunication technology and Internet wideband service, VoIP telephone technology superseded the conventional PSTN technology step by step in recent years, and becoming the development trend of voice communication technology. On the other hand, the form of which developed from Applications based on personal computer to system-on-chip intelligent terminal embedded system based on IP networks.
The Session Initiation Protocol is an application-layer signaling protocol distributed by IETF for creating, modifying, and terminating session with one or more participants. By learning lessons of HTTP and SMTP protocols, reusing many of the header field, encoding rule, protocol procedures and authentication mechanism of HTTP, as a result of its far lower complexity, rich extensibility and better scalability, it is accept by the industry and becoming the best wide-used protocol, also, it is the core protocol of the NGN project.
This thesis decribes the principle, core technology and development trend of IP communication technology in the first part. After that, it analyses the wide-used protocols of IP phone technology, including SIP used for signaling control, SDP for media parameters negotiation, and RTP for voice data real-time transporting. In the following charpters, it details the architecture and processing of design and implementation this terminal. It is implemented on the Broadcom 1100 series voice communication dedicated SoC platform, based on VxWorks real time OS, and controlled the system state machine by message queue mechanism.
At last, this terminal passed the performance and compatibility test, complying with the RFC3261. And it is considered as stable, easily used and high compatible.
SIP(Session Initiation Protocol会话初始协议)是由IETF(Internet Engineering Task Force英特网工程任务组)提出的一个应用层协议,用于创建、修改和终止一个或多个参与者参加的会话。该协议通过借鉴HTTP和SMTP协议风格及其成功经验,继承了HTTP协议的一些头域、编码规则、协议流程和认证机制,以其简单、开放、灵活、可扩展等显著优点,成为为业界所广泛接受,应用最为广泛的VoIP信令协议。
在分析了IP语音通信技术的基本原理、关键技术和发展现状之后,着重讨论了语音通信通常用到的一些重要协议和技术,包括用于信令部分的SIP协议,用于媒体参数协商的SDP协议,用于语音数据实时传输的RTP协议等。然后从整体上介绍了本终端的系统结构和设计过程,最终选用Broadcom 1100系列语音通信专用SoC平台,基与VxWorks实时操作系统,采用消息队列机制控制系统事务状态机运作与转换予以实现。
经过测试,终端实现了RFC3261所规定的基本要求,具有稳定、操作简单等优点,并有良好的互通性。
With the development of IP-based telecommunication technology and Internet wideband service, VoIP telephone technology superseded the conventional PSTN technology step by step in recent years, and becoming the development trend of voice communication technology. On the other hand, the form of which developed from Applications based on personal computer to system-on-chip intelligent terminal embedded system based on IP networks.
The Session Initiation Protocol is an application-layer signaling protocol distributed by IETF for creating, modifying, and terminating session with one or more participants. By learning lessons of HTTP and SMTP protocols, reusing many of the header field, encoding rule, protocol procedures and authentication mechanism of HTTP, as a result of its far lower complexity, rich extensibility and better scalability, it is accept by the industry and becoming the best wide-used protocol, also, it is the core protocol of the NGN project.
This thesis decribes the principle, core technology and development trend of IP communication technology in the first part. After that, it analyses the wide-used protocols of IP phone technology, including SIP used for signaling control, SDP for media parameters negotiation, and RTP for voice data real-time transporting. In the following charpters, it details the architecture and processing of design and implementation this terminal. It is implemented on the Broadcom 1100 series voice communication dedicated SoC platform, based on VxWorks real time OS, and controlled the system state machine by message queue mechanism.
At last, this terminal passed the performance and compatibility test, complying with the RFC3261. And it is considered as stable, easily used and high compatible.
引文
[1]张登银,孙精科. VoIP技术分析与系统设计.北京:人民邮电出版社, 2003: 15-32
[2] D Collins. VoIP技术与应用.舒华英.北京:人民邮电出版社, 2003: 37-46
[3] G Camarillo. SIP揭密.白建军.北京:人民邮电出版社. 2003: 21-29
[4] A B. Johnston. Understanding the Session Initiation Protocol 2nd Edition. Artech House. 2004: 46-57
[5] H Sinnreich, A B. Johnston. Communications Using SIP Delivering VoIP and Multimedia Services with Session Initiation Protocol. Wiley, 2006: 18-25
[6]张传福. IP电话协议及其比较.当代通信, 2003, 12(6): 50-52
[7]虞志文,史济建.基于SIP的Soft Phone的设计与实现.计算机应用, 2003, 9(5): 99-100
[8] O Hersent, D Gurle. IP电话-基于分组的多媒体通信系统.北京:人民邮电出版社, 2002: 91-101
[9]王瑞刚,李燕. IP电话终端设备-原理、电路及应用.西安:西安电子科技大学出版社, 2004: 69-83
[10]申敏. DSP原理及其在移动通信中的应用.北京:人民邮电出版社,2002:103-106
[11]杨昆. IP电话及其增值业务技术.北京:人民邮电出版社,2003: 99-103
[12]赵慧玲,叶华.以软交换为核心的下一代网络技术.北京:人民邮电出版社,2002: 35-94
[13]陈智育,温彦军. VxWorks程序开发实践.北京:人民邮电出版社, 2004: 66-123
[14] PNilanjan Banerjee, Arup Acharya, S K. Das. Enabling SIP-based sessions in ad hoc networks. Wireless Networks. 2007: (13)4: 26-27
[15] D Geneiatakis, G Kambourakis, C Lambrinoudaki. A framework for protecting a SIP-based infrastructure against malformed message attacks. Computer Networks: The International Journal of Computer and Telecommunications Networking. 2007: (51)10: 45-47
[16] Sean Walberg. Expose VoIP problems with wireshark. Linux Journal. 2007: 155: 89-90
[17] Sean Walberg. How to configure SIP and NAT. Linux Journal. 2007: 155: 98-99
[18] M J. Steinmann. Unified communications with SIP. Queue. 2007: (5)2: 24-26
[19] Kundan Singh, H Schulzrinne. Failover, load sharing and server architecture in SIP telephony. Computer Communications. 2007: (30)5: 90-92
[20] Dongmei Jiang, PRamiro Liscano, Luigi Logrippo. Personalization of internet telephony services for presence with SIP and extended CPL. Computer Communications. 2006: (29)18: 21-23
[21] PMario Kolberg, E H. Magill. Managing feature interactions between distributed SIP call control services. Computer Networks: The International Journal of Computer and Telecommunications Networking. 2007: (51)2: 47-48
[22] Xiaotao Wu, Henning Schulzrinne. Handling feature interactions in the language for end system services. Computer Networks: The International Journal of Computer and Telecommunications Networking. 2007: (51)2: 51-53
[23] Aameek Singh, Arup Acharya, PPriya Mahadevan. SPLAT: a unified SIP services platform for VoIP applications: Research Articles. International Journal of Communication Systems. 2006: (19)4: 66
[24] Hyejeong Lee, Jee-Young Song, PSun-Ho Lee. Integrated Mobility Management Methods for Mobile IP and SIP in IP based Wireless Data Networks. Wireless Personal Communications: An International Journal.2005: (35)3
[25]涂继辉,钟宝荣.基于SIP协议点到点软电话的设计与实现.长江大学学报.2005: (12)5: 21-23
[26]车进. IP电话技术研究及其实现.计算机通信. 2006: (7)4: 56-57
[27]刘丽,戴洪波.浅谈SIP的系统结构.电脑知识与技术. 2006: (29)10: 27-29
[28]赵雪梅. IPSec网络安全与实现.计算机应用. 2003: (15)9: 26-27
[29]彭春峰,阮方.基于ARM9的多功能IP电话机的解决方案.计算机应用. 2002: 22(11): 70-71
[30] R. V Cox. Three new speech coders from the ITU cover a range of applications. IEEE Communication Magzine. 1997: 35(9): 47
[31] M. E Pekins, K Pascal. Charactering the subjective performance of the ITU-T 8kb/s speech coding algorithm G.729. IEEE Communications Magazine. 1997: 35(9): 76-81
[32]袁占亭,张秋余,孙为.基于VoIP的电话终端设备的研究和设计.微计算机信息.2003: 19(9): 87
[33]李琳. SIP协议在开放的VOIP模型中的实现.计算机工程. 2002: 2(1):223-227
[34]何永,龙林浒.一种SIP NAT应用网关的设计与实现.通讯世界. 2002: 2(8): 39-42
[35]罗斯青. H.323和SIP的比较.通讯世界. 2002: 2(5): 36-38
[36]杜吉友,张艳霞,董德存.穿越防火墙/NAT的SIP通信研究.中国数据通信. 2004: 4(4):72-75
[37]刘建新,王芙蓉.软交换中SIP与H.323电话网互通的实现.通信技术. 2003: 2(1): 57-60
[38]孟正东,糜正昆.软交换网络体系结构和SIP协议.现代电信科技. 2002: 6(7): 14-18
[39] A. Stephens, P. J. Cordell. SIP and H.323 --- Interworking VoIP Networks. BT Technology Journal. 2001: (19)2: 11-12
[40] PD. R. Wisely. SIP and Conversational Internet Applications. BT Technology Journal. 2001: (19)2: 17-19
[41] PG. Reid, F. J. Aparicio, PA. L. Low. SIP mediated services. BT Technology Journal. 2007: (25)2
[42] Erich M. Nahum, John Tracey, C P. Wright. Evaluating SIP server performance. ACM SIGMETRICS Performance Evaluation Review. 2007: (35)1:28-30
[43] Aameek Singh, Arup Acharya. Multiplayer networked gaming with the session initiation protocol. Computer Networks: The International Journal of Computer and Telecommunications Networking. 2005: (49)1: 32-34
[44] Maurizio Molina, PJurgen Quittek, P Brunner. Scalable and efficient QoS support for SIP-signalled voice calls. International Journal of Communication Systems. 2006: (19)4: 17-18
[2] D Collins. VoIP技术与应用.舒华英.北京:人民邮电出版社, 2003: 37-46
[3] G Camarillo. SIP揭密.白建军.北京:人民邮电出版社. 2003: 21-29
[4] A B. Johnston. Understanding the Session Initiation Protocol 2nd Edition. Artech House. 2004: 46-57
[5] H Sinnreich, A B. Johnston. Communications Using SIP Delivering VoIP and Multimedia Services with Session Initiation Protocol. Wiley, 2006: 18-25
[6]张传福. IP电话协议及其比较.当代通信, 2003, 12(6): 50-52
[7]虞志文,史济建.基于SIP的Soft Phone的设计与实现.计算机应用, 2003, 9(5): 99-100
[8] O Hersent, D Gurle. IP电话-基于分组的多媒体通信系统.北京:人民邮电出版社, 2002: 91-101
[9]王瑞刚,李燕. IP电话终端设备-原理、电路及应用.西安:西安电子科技大学出版社, 2004: 69-83
[10]申敏. DSP原理及其在移动通信中的应用.北京:人民邮电出版社,2002:103-106
[11]杨昆. IP电话及其增值业务技术.北京:人民邮电出版社,2003: 99-103
[12]赵慧玲,叶华.以软交换为核心的下一代网络技术.北京:人民邮电出版社,2002: 35-94
[13]陈智育,温彦军. VxWorks程序开发实践.北京:人民邮电出版社, 2004: 66-123
[14] PNilanjan Banerjee, Arup Acharya, S K. Das. Enabling SIP-based sessions in ad hoc networks. Wireless Networks. 2007: (13)4: 26-27
[15] D Geneiatakis, G Kambourakis, C Lambrinoudaki. A framework for protecting a SIP-based infrastructure against malformed message attacks. Computer Networks: The International Journal of Computer and Telecommunications Networking. 2007: (51)10: 45-47
[16] Sean Walberg. Expose VoIP problems with wireshark. Linux Journal. 2007: 155: 89-90
[17] Sean Walberg. How to configure SIP and NAT. Linux Journal. 2007: 155: 98-99
[18] M J. Steinmann. Unified communications with SIP. Queue. 2007: (5)2: 24-26
[19] Kundan Singh, H Schulzrinne. Failover, load sharing and server architecture in SIP telephony. Computer Communications. 2007: (30)5: 90-92
[20] Dongmei Jiang, PRamiro Liscano, Luigi Logrippo. Personalization of internet telephony services for presence with SIP and extended CPL. Computer Communications. 2006: (29)18: 21-23
[21] PMario Kolberg, E H. Magill. Managing feature interactions between distributed SIP call control services. Computer Networks: The International Journal of Computer and Telecommunications Networking. 2007: (51)2: 47-48
[22] Xiaotao Wu, Henning Schulzrinne. Handling feature interactions in the language for end system services. Computer Networks: The International Journal of Computer and Telecommunications Networking. 2007: (51)2: 51-53
[23] Aameek Singh, Arup Acharya, PPriya Mahadevan. SPLAT: a unified SIP services platform for VoIP applications: Research Articles. International Journal of Communication Systems. 2006: (19)4: 66
[24] Hyejeong Lee, Jee-Young Song, PSun-Ho Lee. Integrated Mobility Management Methods for Mobile IP and SIP in IP based Wireless Data Networks. Wireless Personal Communications: An International Journal.2005: (35)3
[25]涂继辉,钟宝荣.基于SIP协议点到点软电话的设计与实现.长江大学学报.2005: (12)5: 21-23
[26]车进. IP电话技术研究及其实现.计算机通信. 2006: (7)4: 56-57
[27]刘丽,戴洪波.浅谈SIP的系统结构.电脑知识与技术. 2006: (29)10: 27-29
[28]赵雪梅. IPSec网络安全与实现.计算机应用. 2003: (15)9: 26-27
[29]彭春峰,阮方.基于ARM9的多功能IP电话机的解决方案.计算机应用. 2002: 22(11): 70-71
[30] R. V Cox. Three new speech coders from the ITU cover a range of applications. IEEE Communication Magzine. 1997: 35(9): 47
[31] M. E Pekins, K Pascal. Charactering the subjective performance of the ITU-T 8kb/s speech coding algorithm G.729. IEEE Communications Magazine. 1997: 35(9): 76-81
[32]袁占亭,张秋余,孙为.基于VoIP的电话终端设备的研究和设计.微计算机信息.2003: 19(9): 87
[33]李琳. SIP协议在开放的VOIP模型中的实现.计算机工程. 2002: 2(1):223-227
[34]何永,龙林浒.一种SIP NAT应用网关的设计与实现.通讯世界. 2002: 2(8): 39-42
[35]罗斯青. H.323和SIP的比较.通讯世界. 2002: 2(5): 36-38
[36]杜吉友,张艳霞,董德存.穿越防火墙/NAT的SIP通信研究.中国数据通信. 2004: 4(4):72-75
[37]刘建新,王芙蓉.软交换中SIP与H.323电话网互通的实现.通信技术. 2003: 2(1): 57-60
[38]孟正东,糜正昆.软交换网络体系结构和SIP协议.现代电信科技. 2002: 6(7): 14-18
[39] A. Stephens, P. J. Cordell. SIP and H.323 --- Interworking VoIP Networks. BT Technology Journal. 2001: (19)2: 11-12
[40] PD. R. Wisely. SIP and Conversational Internet Applications. BT Technology Journal. 2001: (19)2: 17-19
[41] PG. Reid, F. J. Aparicio, PA. L. Low. SIP mediated services. BT Technology Journal. 2007: (25)2
[42] Erich M. Nahum, John Tracey, C P. Wright. Evaluating SIP server performance. ACM SIGMETRICS Performance Evaluation Review. 2007: (35)1:28-30
[43] Aameek Singh, Arup Acharya. Multiplayer networked gaming with the session initiation protocol. Computer Networks: The International Journal of Computer and Telecommunications Networking. 2005: (49)1: 32-34
[44] Maurizio Molina, PJurgen Quittek, P Brunner. Scalable and efficient QoS support for SIP-signalled voice calls. International Journal of Communication Systems. 2006: (19)4: 17-18