μclinux下基于SIP协议语音网关的软件实现
摘要
SIP协议因其简单、灵活的技术特点,正在逐步代替原有的H.323协议成为IP语音技术(VoIP)的信令核心协议,基于SIP的语音网关研究也随之成为一个热点。然而,目前大多数终端的研究都集中在软终端上,对于嵌入式终端的研究还存在SIP信令控制复杂、硬件相关性较大和通用性不好的问题。为此,本文在研究的基础上,设计并实现了一种信令控制简单、通用性较好的嵌入式语音网关,以适应SIP终端的广泛应用。
本文研究的终端系统可以使位于不同物理位置的用户,使用不同的设备通过互联网进行实时语音通信。它主要采用基于SIP的信令技术来建立语音会话,利用事务状态机模型进行信令控制,使用SDP媒体协商技术进行语音协商,结合RTP实时传输技术,实现了端到端的语音实时传输。
论文首先着重研究了基于SIP协议的语音通信技术,详细的描述了SIP协议及其用于信令呼叫控制技术,以及实时传输协议RTP和媒体协商协议SDP的包的格式及具体应用;从整体上介绍了软件实现的系统平台,即语音网关的软硬体系统结构的组成,硬件部分包括语音网关的硬件实现、各主要芯片的功能,语音网关对语音信号处理流程,软件部分包括嵌入式操作系统μclinux的简介,BootLoader移植,开发环境的建立,内核的精简;重点阐述了语音网关应用软件的设计思想和具体实现,其中包括其应用定位、运行状态设计及任务划分和各任务的功能,强调了各模块的接口设计、各模块处理流程设计、数据结构的定义。在信令控制部分,实现了一种基于消息映射事务状态机模型转换,解决了信令控制复杂的问题。在语音处理部分,定义了语音传输功能的相关接口,完成了实时语音传输的工作流程。经过系统测试,语音网关的SIP信令流程符合RFC3261标准,语音清晰。
该课题的研究对于实现语音与数据在网络边缘的融合,从而实现现有网络向下一代网络演进具有十分重要的现实意义。
With the simple and flexible advantages,SIP protocol has been gradually taking the place of the original H.323 protocol and becomes the core protocol in VoIP,thus,the research on voice gateway based on SIP becomes a hot issue.However,most recent research is put on soft terminals,and embedded terminals also have the complexity problems on the signaling control and extensive use,so this thesis designs and implements a kind of embedded voice gateway that simplifies the signaling control for the extensive use of the sip terminal.
The terminal this thesis research on can make users from different locations communicate with others in voice over IP.It uses sip signal technology to set up the voice session and complete the signal control,with the SDP and RTP transport technology,the voice gateway achieves the real-time voice communication from end to end.
This thesis firstly research on the voice communication technique based on SIP,which include SIP and sip signal call, real time transport protocol and media negotiation SDP as well as its packet format and application described in detail.It introduces the architecture of voice gateway.Hardware implementation,the main chip function,the process of voice signal are outlined in the hardware;System software simpley introduce embedded operating systemμclinux and emphasize on the transplanting ofμclinux's bootloader and core.The voice gateway application software design thinking and concrete realization are discussed on detail,which include its application location,running state design,task partitioning and function. The module's interface design,task process design,data structure definition are emphasized on.The transaction state machine based on message mapping is designed to solve the complexity problem on the signaling control;in the part of media session,the interface and work flow about voice processing is mainly introduced.The voice gateway SIP signal process conforms to RFC3261,and its voice quality is clear by system test.
The study of subject promote to the integration of voice and data at the network edge,and realizing the existing network to the next generation network evolution is of great practical significance.
本文研究的终端系统可以使位于不同物理位置的用户,使用不同的设备通过互联网进行实时语音通信。它主要采用基于SIP的信令技术来建立语音会话,利用事务状态机模型进行信令控制,使用SDP媒体协商技术进行语音协商,结合RTP实时传输技术,实现了端到端的语音实时传输。
论文首先着重研究了基于SIP协议的语音通信技术,详细的描述了SIP协议及其用于信令呼叫控制技术,以及实时传输协议RTP和媒体协商协议SDP的包的格式及具体应用;从整体上介绍了软件实现的系统平台,即语音网关的软硬体系统结构的组成,硬件部分包括语音网关的硬件实现、各主要芯片的功能,语音网关对语音信号处理流程,软件部分包括嵌入式操作系统μclinux的简介,BootLoader移植,开发环境的建立,内核的精简;重点阐述了语音网关应用软件的设计思想和具体实现,其中包括其应用定位、运行状态设计及任务划分和各任务的功能,强调了各模块的接口设计、各模块处理流程设计、数据结构的定义。在信令控制部分,实现了一种基于消息映射事务状态机模型转换,解决了信令控制复杂的问题。在语音处理部分,定义了语音传输功能的相关接口,完成了实时语音传输的工作流程。经过系统测试,语音网关的SIP信令流程符合RFC3261标准,语音清晰。
该课题的研究对于实现语音与数据在网络边缘的融合,从而实现现有网络向下一代网络演进具有十分重要的现实意义。
With the simple and flexible advantages,SIP protocol has been gradually taking the place of the original H.323 protocol and becomes the core protocol in VoIP,thus,the research on voice gateway based on SIP becomes a hot issue.However,most recent research is put on soft terminals,and embedded terminals also have the complexity problems on the signaling control and extensive use,so this thesis designs and implements a kind of embedded voice gateway that simplifies the signaling control for the extensive use of the sip terminal.
The terminal this thesis research on can make users from different locations communicate with others in voice over IP.It uses sip signal technology to set up the voice session and complete the signal control,with the SDP and RTP transport technology,the voice gateway achieves the real-time voice communication from end to end.
This thesis firstly research on the voice communication technique based on SIP,which include SIP and sip signal call, real time transport protocol and media negotiation SDP as well as its packet format and application described in detail.It introduces the architecture of voice gateway.Hardware implementation,the main chip function,the process of voice signal are outlined in the hardware;System software simpley introduce embedded operating systemμclinux and emphasize on the transplanting ofμclinux's bootloader and core.The voice gateway application software design thinking and concrete realization are discussed on detail,which include its application location,running state design,task partitioning and function. The module's interface design,task process design,data structure definition are emphasized on.The transaction state machine based on message mapping is designed to solve the complexity problem on the signaling control;in the part of media session,the interface and work flow about voice processing is mainly introduced.The voice gateway SIP signal process conforms to RFC3261,and its voice quality is clear by system test.
The study of subject promote to the integration of voice and data at the network edge,and realizing the existing network to the next generation network evolution is of great practical significance.
引文
[1]欧建平,娄生强。网络与多媒体通信技术。北京:人民邮电出版社,2005
[2]Mehta.P,S.Udani."Voice over IP",IEEEPotentials,pp.36-40.October/November2001
[3]司端锋,韩心慧等.SIP标准中的核心技术与研究进展,软件学报。 2005,16
[4]CINEMA.Columbia.Internet.Extensive.Multimedia.Architecture,http://www.cs.columbia.edu/IRT/cinema/
[5]SER.http://www.iptel.org/ser.
[6]VOCAL.http://www.vovida.org.
[7]oSIP.http://www.gnu.org/software/osip.
[8]reciprocate,http://www.sipfoundry.org/reSIProcate.
[9]sipsys,http://www.sipsys.com.
[10]尚群.基于北京大学无线网的VOIP实现.CERNET 2006.
[11]王海波.基于ARM9的嵌入式SIP服务器的研究与设计.河海大学.2006
[12]张登银.VOIP技术分析与系统设计.北京:人民邮电出版社.2003年5月
[13]Recommendation,G.711,G.723.G.729,InternationalTelecommunic ation Union,The International Telegraph and Telephone Consultative Committee,Geneva,1988
[14]邢竞帆,李胜利.SIP协议的研究,计算机应用,2002,21(8):10-12
[15]糜正琨.IP网络电话技术.北京:人民邮电出版社.2000.6.
[16]J.Rosenberg.H.Sehulzrinne.G.Camarillo.A.Johnston.J.Peterso n,R.Sparks,M.Handley,E.Sehooler.SIP:SessionInitiationProt ocol[R]RFC3261,June2002.
[17]M.Handley,H.Sehulzrinne,E.Sehooler,J.Rosenberg,SIP:Session Initiation Protocol[R],RFC2543,Mareh 1999.
[18]白建军,彭晖译.SIP揭密.北京:人民邮电出版社.2003年6月
[19]LuanDang,DavidKelly."PracticalVoIPUsing.VOCAL"REILLY.July 2002
[20]H.Schulzrinne,S.Casner,R.Frederick,Vjacobson."RTP:A Transoport Protocol for Real-Time Application" RFC1889.Jannuary 1996
[21]Handley.Mand.V.Jacobson.SDP:Session,Description,Protocol.IETF RFC2327.April 1998
[22]W90N740 USER' S MANUAL[Z],WINBOND,2003.10.
[23]陆宝格,何锐.如何编译Culinux内核(J).单片机与嵌入式系统 应用,2002(11):76-80.
[24]陈莉君.深入分析Linux内核源代码[M].北京:人民邮电出版社,2002.
[25]张秋余,魏政.嵌入式μ clinux家庭网关[J].电子产品世界,2003(12):59-64.
[26]王学龙.嵌入式Linux系统设计与应用[M].北京:清华大学出版社,2001.
[27]曾宏安,齐尧,焦振强等.用U-BOTO构建嵌入式系统的引导装载程序[J].单片机与嵌入式系统应用,2003,10:34-37.
[28]魏忠,蔡勇,雷红卫.嵌入式开发详解[M].北京:电子工业出版社,2003
[31]W.Richard Stevens.UNIX network programming(volume 2):InterprocessCommunications(second edition).Beijing,China:Tsinghua university press.2002.126-136
[32]Jon Postel.Internet Protocol.IETF RFC 791.1981.1-31
[33]W.Richard Stevens.TCP/IP详解(卷1):协议.范建华,胥光辉,张涛等译.北京:机械工业出版社,2000.24-243
[34]W.Richard Stevens.UNIX环境高级编程.尤晋元等译.北京:机械工业出版社,2001.148-153
[35]Gary R.Wright W.Richard Stevens.TCP/IP详解(卷2):实现.陆雪莹,蒋慧等译.北京:机械工业出版社,2000.20-250
[36]Daniel P.Bovet,Marco Cesati.Understanding Linux Kernel.Reilly press.2001.103-154
[37]涂万青.基于RTP的IP网络交互视频终端QoS保证技术的研究:[硕士学位论文].广州:中山大学,2002.
[38]胡春筠.基于RTP协议的IP实时语音端到端传输的研究:[硕士学位论文]。广州:华南理工大学,2003.
[39]严蔚敏,吴伟民.数据结构.第二版.北京:清华大学出版社,1992.30-103
[40]J.Rosenberg,H.Schulzrinne,G.Camarilloand so on."Session initiation protocol".RFC3261,Internet Engineering Task Force.June 2002.
[41]H.Schulzrinne,S.Casner,R.Frederick,V.Jacobson."RTP:A Transport Protocolfor Real-Time Application" RFC1889.January 1996
[42]AudioCodes L td.AC4880xC3C Datasheet V 3.10[Z],2003.2.
[43]Radvision L td.SIP Stack Toolkit Programmer Guider Version 3.31 September.2004 pp15-30
[2]Mehta.P,S.Udani."Voice over IP",IEEEPotentials,pp.36-40.October/November2001
[3]司端锋,韩心慧等.SIP标准中的核心技术与研究进展,软件学报。 2005,16
[4]CINEMA.Columbia.Internet.Extensive.Multimedia.Architecture,http://www.cs.columbia.edu/IRT/cinema/
[5]SER.http://www.iptel.org/ser.
[6]VOCAL.http://www.vovida.org.
[7]oSIP.http://www.gnu.org/software/osip.
[8]reciprocate,http://www.sipfoundry.org/reSIProcate.
[9]sipsys,http://www.sipsys.com.
[10]尚群.基于北京大学无线网的VOIP实现.CERNET 2006.
[11]王海波.基于ARM9的嵌入式SIP服务器的研究与设计.河海大学.2006
[12]张登银.VOIP技术分析与系统设计.北京:人民邮电出版社.2003年5月
[13]Recommendation,G.711,G.723.G.729,InternationalTelecommunic ation Union,The International Telegraph and Telephone Consultative Committee,Geneva,1988
[14]邢竞帆,李胜利.SIP协议的研究,计算机应用,2002,21(8):10-12
[15]糜正琨.IP网络电话技术.北京:人民邮电出版社.2000.6.
[16]J.Rosenberg.H.Sehulzrinne.G.Camarillo.A.Johnston.J.Peterso n,R.Sparks,M.Handley,E.Sehooler.SIP:SessionInitiationProt ocol[R]RFC3261,June2002.
[17]M.Handley,H.Sehulzrinne,E.Sehooler,J.Rosenberg,SIP:Session Initiation Protocol[R],RFC2543,Mareh 1999.
[18]白建军,彭晖译.SIP揭密.北京:人民邮电出版社.2003年6月
[19]LuanDang,DavidKelly."PracticalVoIPUsing.VOCAL"REILLY.July 2002
[20]H.Schulzrinne,S.Casner,R.Frederick,Vjacobson."RTP:A Transoport Protocol for Real-Time Application" RFC1889.Jannuary 1996
[21]Handley.Mand.V.Jacobson.SDP:Session,Description,Protocol.IETF RFC2327.April 1998
[22]W90N740 USER' S MANUAL[Z],WINBOND,2003.10.
[23]陆宝格,何锐.如何编译Culinux内核(J).单片机与嵌入式系统 应用,2002(11):76-80.
[24]陈莉君.深入分析Linux内核源代码[M].北京:人民邮电出版社,2002.
[25]张秋余,魏政.嵌入式μ clinux家庭网关[J].电子产品世界,2003(12):59-64.
[26]王学龙.嵌入式Linux系统设计与应用[M].北京:清华大学出版社,2001.
[27]曾宏安,齐尧,焦振强等.用U-BOTO构建嵌入式系统的引导装载程序[J].单片机与嵌入式系统应用,2003,10:34-37.
[28]魏忠,蔡勇,雷红卫.嵌入式开发详解[M].北京:电子工业出版社,2003
[31]W.Richard Stevens.UNIX network programming(volume 2):InterprocessCommunications(second edition).Beijing,China:Tsinghua university press.2002.126-136
[32]Jon Postel.Internet Protocol.IETF RFC 791.1981.1-31
[33]W.Richard Stevens.TCP/IP详解(卷1):协议.范建华,胥光辉,张涛等译.北京:机械工业出版社,2000.24-243
[34]W.Richard Stevens.UNIX环境高级编程.尤晋元等译.北京:机械工业出版社,2001.148-153
[35]Gary R.Wright W.Richard Stevens.TCP/IP详解(卷2):实现.陆雪莹,蒋慧等译.北京:机械工业出版社,2000.20-250
[36]Daniel P.Bovet,Marco Cesati.Understanding Linux Kernel.Reilly press.2001.103-154
[37]涂万青.基于RTP的IP网络交互视频终端QoS保证技术的研究:[硕士学位论文].广州:中山大学,2002.
[38]胡春筠.基于RTP协议的IP实时语音端到端传输的研究:[硕士学位论文]。广州:华南理工大学,2003.
[39]严蔚敏,吴伟民.数据结构.第二版.北京:清华大学出版社,1992.30-103
[40]J.Rosenberg,H.Schulzrinne,G.Camarilloand so on."Session initiation protocol".RFC3261,Internet Engineering Task Force.June 2002.
[41]H.Schulzrinne,S.Casner,R.Frederick,V.Jacobson."RTP:A Transport Protocolfor Real-Time Application" RFC1889.January 1996
[42]AudioCodes L td.AC4880xC3C Datasheet V 3.10[Z],2003.2.
[43]Radvision L td.SIP Stack Toolkit Programmer Guider Version 3.31 September.2004 pp15-30