SIP协议在多媒体业务中的研究与实现
摘要
Internet的发展大大推动了基于IP协议的数据通信的发展,为集成语音、图像和数据的多媒体业务提供了有力的支持。H.323协议是较早出现的用于多媒体通信的信令协议,但其呼叫信令的建立受到传统电信网的封闭架构和低带宽的局限。而且,H.323协议完成多媒体通信会话的建立过程十分复杂,有些业务需要在多个协议之间交互,这样限制了多媒体业务的快速开发。因此,我们需要新的协议来改善这个局面。这个协议就是由IETF制定的会话初始协议(SIP,Session InitiationProtocol),它是对基于IP的多媒体通信系统进行控制的协议。SIP具有简单、可扩展的特性,因此通过SIP进行业务管理,可以更快速更灵活的支持多媒体业务。所以SIP在多媒体业务中的应用具有重要意义。
本文设计实现了基于SIP的多媒体终端系统,该系统利用SIP协议实现了多媒体呼叫控制,同时可以利用RTP协议支持视音频双向通信。具体工作包括:首先对SIP协议及其相关技术进行深入细致的研究,接着对SIP协议的体系结构及JAIN(Java APIs for Integrated Network)SIP协议栈进行了详细的分析。在此基础上,论文提出了系统的总体设计方案,系统主要由五个模块来实现,分别是主控模块、SIP模块、视音频处理模块、即时消息模块和界面模块。各模块的功能如下:主控模块完成了整个系统的初始化工作;而其中SIP模块是整个系统的核心模块,是以JAIN SIP为基础,具体实现了SIP通信中各通信实体的基本方法,从而完成了SIP信令功能;视音频处理模块是采用Java媒体框架(JMF,Java Media Framework)来实现的,它实现了视频和音频的采集、传输和播放,其中选用了GSM编码格式的音频和H.263编码格式的视频;此外,还对该系统进行了优化和完善,加入了MESSAGE方法,使得能够在视音频通信的同时完成即时消息的交互;而且该系统具有良好的操作界面,包括用户端设置、服务器端设置和用户昵称、端口号的设置等。
本系统经过在局域网上的多次测试,实现了UAC到UAS的直接通信和通过SIP服务器的通信,并通过SIPFlow协议分析软件分析出信令格式正确,而且视音频通信流畅,实现了设计时所期望实现的功能。
The development of Internet greatly promotes the development of data communication based on IP protocol, and provides strong support for multimedia services of integrated voice, image and data. H.323 is the earlier signaling protocol for multimedia communications, but its establishment of call signaling is limited by the closed structure of traditional telecommunication networks and low-bandwidth. Furthermore, the process that H.323 protocol realizes the establishment of multimedia communication sessions is complex, and some services need interaction between several protocols. In this way, it limites the rapid exploitation of multimedia services. So we need a new protocol to improve this situation. That is Session Initiation Protocol(SIP) which is formulated by IETF, and it's used to control the system of multimedia communication based on IP. SIP is simply structured and easy to be extended, thereby, it can accomplish multimedia services more quickly and neatly. So the application of SIP in the multimedia services possesses great significance.
This paper designs and implements the multimedia terminal system based on SIP, which uses SIP protocol to realize multimedia call control and uses RTP protocol to support two-way video and audio communication. The specific tasks include: First, do embed and meticulous researches on SIP and related techniques, then analyze the SIP system structure and JAIN(Java APIs for Integrated Network) SIP protocol stack in detail. On the base of frontal work, this paper gives a total design scheme of the system. The realization of the system is made up of five modules, which are main control module, SIP module, video&audio disposal module, instant message module and display module. The functions of each module are as follows: The main control module completes initialization of the whole system. SIP module is the core module of the system, which is based on the JAIN SIP protocol stack, and it implements the basic method of the communication entity in detail, so that the SIP functions are realized. The video&audio disposal module implemented by Java Media Framework (JMF) makes the function of capturing, transmission and play. GSM and H.263 are choosed as the compress formats of audio and video. Then to optimize and consummate this system, MESSAGE method is added into system, it can implement the alternation of instant message when media communication is in process. Furthermore, this system has a favorable operation interface, including user setting, server setting, user's nickname setting and port setting, etc.
This system is tested in the LAN for many times, and the results show that it can accomplish communication from UAC to UAS and communication through SIP server. Through SIPFlow protocol analysis software, the analytic results of the signalings' format are right. What's more, the video and audio communications are quite fluent. So the functions that are expected in the design are realized.
本文设计实现了基于SIP的多媒体终端系统,该系统利用SIP协议实现了多媒体呼叫控制,同时可以利用RTP协议支持视音频双向通信。具体工作包括:首先对SIP协议及其相关技术进行深入细致的研究,接着对SIP协议的体系结构及JAIN(Java APIs for Integrated Network)SIP协议栈进行了详细的分析。在此基础上,论文提出了系统的总体设计方案,系统主要由五个模块来实现,分别是主控模块、SIP模块、视音频处理模块、即时消息模块和界面模块。各模块的功能如下:主控模块完成了整个系统的初始化工作;而其中SIP模块是整个系统的核心模块,是以JAIN SIP为基础,具体实现了SIP通信中各通信实体的基本方法,从而完成了SIP信令功能;视音频处理模块是采用Java媒体框架(JMF,Java Media Framework)来实现的,它实现了视频和音频的采集、传输和播放,其中选用了GSM编码格式的音频和H.263编码格式的视频;此外,还对该系统进行了优化和完善,加入了MESSAGE方法,使得能够在视音频通信的同时完成即时消息的交互;而且该系统具有良好的操作界面,包括用户端设置、服务器端设置和用户昵称、端口号的设置等。
本系统经过在局域网上的多次测试,实现了UAC到UAS的直接通信和通过SIP服务器的通信,并通过SIPFlow协议分析软件分析出信令格式正确,而且视音频通信流畅,实现了设计时所期望实现的功能。
The development of Internet greatly promotes the development of data communication based on IP protocol, and provides strong support for multimedia services of integrated voice, image and data. H.323 is the earlier signaling protocol for multimedia communications, but its establishment of call signaling is limited by the closed structure of traditional telecommunication networks and low-bandwidth. Furthermore, the process that H.323 protocol realizes the establishment of multimedia communication sessions is complex, and some services need interaction between several protocols. In this way, it limites the rapid exploitation of multimedia services. So we need a new protocol to improve this situation. That is Session Initiation Protocol(SIP) which is formulated by IETF, and it's used to control the system of multimedia communication based on IP. SIP is simply structured and easy to be extended, thereby, it can accomplish multimedia services more quickly and neatly. So the application of SIP in the multimedia services possesses great significance.
This paper designs and implements the multimedia terminal system based on SIP, which uses SIP protocol to realize multimedia call control and uses RTP protocol to support two-way video and audio communication. The specific tasks include: First, do embed and meticulous researches on SIP and related techniques, then analyze the SIP system structure and JAIN(Java APIs for Integrated Network) SIP protocol stack in detail. On the base of frontal work, this paper gives a total design scheme of the system. The realization of the system is made up of five modules, which are main control module, SIP module, video&audio disposal module, instant message module and display module. The functions of each module are as follows: The main control module completes initialization of the whole system. SIP module is the core module of the system, which is based on the JAIN SIP protocol stack, and it implements the basic method of the communication entity in detail, so that the SIP functions are realized. The video&audio disposal module implemented by Java Media Framework (JMF) makes the function of capturing, transmission and play. GSM and H.263 are choosed as the compress formats of audio and video. Then to optimize and consummate this system, MESSAGE method is added into system, it can implement the alternation of instant message when media communication is in process. Furthermore, this system has a favorable operation interface, including user setting, server setting, user's nickname setting and port setting, etc.
This system is tested in the LAN for many times, and the results show that it can accomplish communication from UAC to UAS and communication through SIP server. Through SIPFlow protocol analysis software, the analytic results of the signalings' format are right. What's more, the video and audio communications are quite fluent. So the functions that are expected in the design are realized.
引文
[1]李旭东.基于SIP的多媒体通信系统的设计与实现[硕士学位论文].吉林大学.2007.6.1-2
[2]高伟.基于SIP协议的IP电话研究与实现[硕士学位论文].西南交通大学.2007.4.4-5
[3]黄永峰.IP网络多媒体通信技术.北京:人民邮电出版社.2002.6-15
[4]Daniel Collions.VOIP技术与应用.人民邮电出版社.2003.6.8-28.3-30
[5]沈波,张顺颐,沈苏彬.会话发起协议SIP的分析和研究.数据通信.2001(4):15-18
[6]胡凌凌,彭容修.SIP协议在一个IP电话模型中的实现.微机发展.2005年2月第15卷第2期.1-3
[7]周海华,边恩炯.SIP原理与应用.北京机械工业出版社.2006.8-20
[8]张智江,张云勇,刘韵沽.SIP协议及其应用.电子工业出版社.2005
[9]司端锋,韩心慧,龙勤,潘爱民.SIP标准中的核心技术与研究进展.软件学报.Vol16.No 2
[10]Gonzalo Camarillo.白建军,彭晖田敏等译.SIP揭秘.北京:人民邮电出版社.2003.63-70
[11]谢希仁.计算机网络.北京:电子工业出版社.1999
[12]J.Rosenberg,H.Schulzrirme,G.Camanilo,A.Johnston,J.Peterson,R.Sparks.SIP:Session initiation protocol.RFC 3261.June 2002.1-16
[13]陈银星,周利清.SIP协议在VOIP终端的设计和实现.现代电信科技.2006年第10期.11-13
[14]B.Campbell,J.Rosenberg.Session initiation protocol(SIP)extention for instant messaging.Intemet RFC3428.2002.2-10
[15]A.Mankin,S.Bradner,R.Mahy,etc.Change Process for the session Initiation Protocol(SIP).Internet RFC3427.2002.2-6
[16]J.Rosenberg,H.Schulzrinne.Reliability of provisional responses in the session initiation protocol(SIP).Internet RFC 3262.2002.2-8
[17]Roach AB.session initiation protocol(SIP)-specific event notification.Intemet RFC 3265.2002.3-9
[18]J.Rosenberg.The session initiation protocol(SIP)UPDATE method.Intemet RFC 3311.2002.2-8
[19]R.Sparks,The session initiation protocol(SIP)refer method.Intemet RFC 3515.2003.5-10
[20]S.Donovan.The SIP INFO method.Intemet RFC 2976.2000.2-7
[21]沈鑫剡.多媒体传输网络与VOIP系统设计.人民邮电出版社.2005.3.277-287
[22]M.Handley,V.Jacobson.SDP:Session Description Protocol.RFC 2327.April 1998.7-29
[23]魏春城.SIP协议的特点及应用.电信科学.2002年18卷9期.64-66
[24]孟旭东.H.323与SIP的比较.电信技术.2001年12期.15-18
[25]闭彬煊.SIP与H.323在Internet电话信令方面的比较,广西交通科技,2000年12月第25期增刊:155-159
[26]李爽,赵琛.H.323和SIP协议介绍及比较.中国数据通信.2001(12):35-39
[27]何彬,张国清.SIP可视电话系统的信令流分析.计算机工程与应用.2005年41卷15期.157-159
[28]雷正雄,朱晓民,廖建新.SIP协议栈的设计与实现.现代电信科技.2004.(13):17-20
[29]几种开源SIP协议栈对比:http://blog.csdn.net/da51e/archive.
[30]The JAIN~(TM)APIs:Integrated Network APIs for the Java~(TM)Platform.May 2002. http://java.sun.com/products/jain/
[31]沈凤译.Java网络编程与分布式计算.北京:机械工业出版社.2003.12-60
[32]姜利群.Java网络编程基础.北京:清华大学出版社.2003.16-22
[33]李芝兴.JAVA程序设计之网络编程.北京:清华大学出版社.2006.18-27
[34]周静,邹北骥.利用JMF实现RTP流的传输与播放[J].湖南大学学报(自然科学报)2003.30(3):186-188
[35]Java Media Framework[EB/OL].2004-02-10.http://java.sun.com/products/java-media/jmf/index.jsp
[36]贾谦,徐晨墒,刘增基.SIP协议扩展技术研究.现代电信科技.2003年第7期
[37]张桂元,贾燕枫.Eclipse开发入门与项目实践.北京:人民邮电出版社.2006.15-35
[38]J.Lewis,W.Loftus.王锦全译.Java程序设计基础[M].第3版.16-19
[39]HORTONI.李昭智,王哲明,王红广等译.Java2编程指南(SDK1.4版).北京:电子工业出版社,2003.16-20
[2]高伟.基于SIP协议的IP电话研究与实现[硕士学位论文].西南交通大学.2007.4.4-5
[3]黄永峰.IP网络多媒体通信技术.北京:人民邮电出版社.2002.6-15
[4]Daniel Collions.VOIP技术与应用.人民邮电出版社.2003.6.8-28.3-30
[5]沈波,张顺颐,沈苏彬.会话发起协议SIP的分析和研究.数据通信.2001(4):15-18
[6]胡凌凌,彭容修.SIP协议在一个IP电话模型中的实现.微机发展.2005年2月第15卷第2期.1-3
[7]周海华,边恩炯.SIP原理与应用.北京机械工业出版社.2006.8-20
[8]张智江,张云勇,刘韵沽.SIP协议及其应用.电子工业出版社.2005
[9]司端锋,韩心慧,龙勤,潘爱民.SIP标准中的核心技术与研究进展.软件学报.Vol16.No 2
[10]Gonzalo Camarillo.白建军,彭晖田敏等译.SIP揭秘.北京:人民邮电出版社.2003.63-70
[11]谢希仁.计算机网络.北京:电子工业出版社.1999
[12]J.Rosenberg,H.Schulzrirme,G.Camanilo,A.Johnston,J.Peterson,R.Sparks.SIP:Session initiation protocol.RFC 3261.June 2002.1-16
[13]陈银星,周利清.SIP协议在VOIP终端的设计和实现.现代电信科技.2006年第10期.11-13
[14]B.Campbell,J.Rosenberg.Session initiation protocol(SIP)extention for instant messaging.Intemet RFC3428.2002.2-10
[15]A.Mankin,S.Bradner,R.Mahy,etc.Change Process for the session Initiation Protocol(SIP).Internet RFC3427.2002.2-6
[16]J.Rosenberg,H.Schulzrinne.Reliability of provisional responses in the session initiation protocol(SIP).Internet RFC 3262.2002.2-8
[17]Roach AB.session initiation protocol(SIP)-specific event notification.Intemet RFC 3265.2002.3-9
[18]J.Rosenberg.The session initiation protocol(SIP)UPDATE method.Intemet RFC 3311.2002.2-8
[19]R.Sparks,The session initiation protocol(SIP)refer method.Intemet RFC 3515.2003.5-10
[20]S.Donovan.The SIP INFO method.Intemet RFC 2976.2000.2-7
[21]沈鑫剡.多媒体传输网络与VOIP系统设计.人民邮电出版社.2005.3.277-287
[22]M.Handley,V.Jacobson.SDP:Session Description Protocol.RFC 2327.April 1998.7-29
[23]魏春城.SIP协议的特点及应用.电信科学.2002年18卷9期.64-66
[24]孟旭东.H.323与SIP的比较.电信技术.2001年12期.15-18
[25]闭彬煊.SIP与H.323在Internet电话信令方面的比较,广西交通科技,2000年12月第25期增刊:155-159
[26]李爽,赵琛.H.323和SIP协议介绍及比较.中国数据通信.2001(12):35-39
[27]何彬,张国清.SIP可视电话系统的信令流分析.计算机工程与应用.2005年41卷15期.157-159
[28]雷正雄,朱晓民,廖建新.SIP协议栈的设计与实现.现代电信科技.2004.(13):17-20
[29]几种开源SIP协议栈对比:http://blog.csdn.net/da51e/archive.
[30]The JAIN~(TM)APIs:Integrated Network APIs for the Java~(TM)Platform.May 2002. http://java.sun.com/products/jain/
[31]沈凤译.Java网络编程与分布式计算.北京:机械工业出版社.2003.12-60
[32]姜利群.Java网络编程基础.北京:清华大学出版社.2003.16-22
[33]李芝兴.JAVA程序设计之网络编程.北京:清华大学出版社.2006.18-27
[34]周静,邹北骥.利用JMF实现RTP流的传输与播放[J].湖南大学学报(自然科学报)2003.30(3):186-188
[35]Java Media Framework[EB/OL].2004-02-10.http://java.sun.com/products/java-media/jmf/index.jsp
[36]贾谦,徐晨墒,刘增基.SIP协议扩展技术研究.现代电信科技.2003年第7期
[37]张桂元,贾燕枫.Eclipse开发入门与项目实践.北京:人民邮电出版社.2006.15-35
[38]J.Lewis,W.Loftus.王锦全译.Java程序设计基础[M].第3版.16-19
[39]HORTONI.李昭智,王哲明,王红广等译.Java2编程指南(SDK1.4版).北京:电子工业出版社,2003.16-20