无线IP移动终端VoIP技术的研究与实现
摘要
VoIP(Voice over Internet Protocol)指的是将模拟的声音讯号经过压缩与封包之后,以数据封包的形式在IP网络的环境进行语音讯号的传输,具有低成本,高扩展性的特点。目前,国内外许多公司都在积极开发基于VoIP的应用,VoIP技术相对来说已经比较成熟。但是由于IP网络对语音和视频等实时通信业务的服务质量(Quality of Service)QoS无法保障,VoIP QoS成为推广VoIP应用面临的一个技术瓶颈。本论文针对此技术瓶颈做了研究和探讨,并通过移动终端项目给出了相应的解决方案。论文基于的移动终端项目的整个开发过程是依据软件工程的开发流程进行的。
论文简单描述了VoIP技术的国内外研究现状及面临的QoS问题;介绍了VoIPQoS原理,重点分析了影响VoIP终端QoS的关键因素:语音编码方式、抖动缓冲控制技术、回声消除技术,并针对这几种因素提出了目前先进的终端解决方案:针对编码方式,基于带宽需求、平均观点分值、失真损害值、复杂性等指标,对各种编解码方式的性能和特点进行了评估;分析了当前主流的抖动缓冲算法各自的实现原理和性能优劣,提出了性能优化的Jitter缓冲自适应调整算法来解决抖动问题;介绍了回声消除技术原理,提出了用声学回声消除器解决声学回声问题;最后结合实际终端设计对理论部分进行了验证,给出了具体的实现流程及主要数据结构和算法;论文结束部分对论文内容做了简单总结并提出了论文中需要进一步研究的技术。
本论文基于的移动终端项目目前已经完成,产品已经投放市场。
VoIP is the abbreviated form of Voice over Internet Protocol, it means transmitting analog signals of voice on Internet after converting them into digital signals,so it has the characters of low cost and high expansibility. In recent years, more and more companies are trying to develop products based on VoIP,and their research results make VoIP technique relatively mature now. But a coming serious problem is that IP Internet can not promise the (Quality of Service)QoS of real-time communication signals like voice and video,so how to guarantee VoIP QoS become the bottleneck to popularize VoIP application. This thesis is just based on this bottleneck and it provide the relevant solutions.
The first part of the thesis describes the current research situation of VoIP at home and abroad and extracts the QoS problems now that we should face to. The middle part describes the theory of VoIP QoS, analyzes the key factors that affect the QoS: voice coding method, jitter buffer technique, echo cancellation, then put forward to relavent proposals: Evaluate the different voice encryption algorithm basing on bandwidth requirement, average viewpoint value, distortion damage value,complexity and other more marks; Introduce the jitter buffer algorithm and praise self-adapting jitter buffer algorithm is the best algorithm currently to resolve the jitter; Discribes echo cancellation and provide the method of echo canceller. The last part introsuces the mobile terminal project, emphasize the relevant methods that should be adopted to guarantee QoS, provide the concrete design process and the main data structure and algorithm; The latter end gives the conclusion of the whole thesis and provides the techniques should be further researched next step.
The project that this thesis bases on has been finished and the prouduct has been put into market.
论文简单描述了VoIP技术的国内外研究现状及面临的QoS问题;介绍了VoIPQoS原理,重点分析了影响VoIP终端QoS的关键因素:语音编码方式、抖动缓冲控制技术、回声消除技术,并针对这几种因素提出了目前先进的终端解决方案:针对编码方式,基于带宽需求、平均观点分值、失真损害值、复杂性等指标,对各种编解码方式的性能和特点进行了评估;分析了当前主流的抖动缓冲算法各自的实现原理和性能优劣,提出了性能优化的Jitter缓冲自适应调整算法来解决抖动问题;介绍了回声消除技术原理,提出了用声学回声消除器解决声学回声问题;最后结合实际终端设计对理论部分进行了验证,给出了具体的实现流程及主要数据结构和算法;论文结束部分对论文内容做了简单总结并提出了论文中需要进一步研究的技术。
本论文基于的移动终端项目目前已经完成,产品已经投放市场。
VoIP is the abbreviated form of Voice over Internet Protocol, it means transmitting analog signals of voice on Internet after converting them into digital signals,so it has the characters of low cost and high expansibility. In recent years, more and more companies are trying to develop products based on VoIP,and their research results make VoIP technique relatively mature now. But a coming serious problem is that IP Internet can not promise the (Quality of Service)QoS of real-time communication signals like voice and video,so how to guarantee VoIP QoS become the bottleneck to popularize VoIP application. This thesis is just based on this bottleneck and it provide the relevant solutions.
The first part of the thesis describes the current research situation of VoIP at home and abroad and extracts the QoS problems now that we should face to. The middle part describes the theory of VoIP QoS, analyzes the key factors that affect the QoS: voice coding method, jitter buffer technique, echo cancellation, then put forward to relavent proposals: Evaluate the different voice encryption algorithm basing on bandwidth requirement, average viewpoint value, distortion damage value,complexity and other more marks; Introduce the jitter buffer algorithm and praise self-adapting jitter buffer algorithm is the best algorithm currently to resolve the jitter; Discribes echo cancellation and provide the method of echo canceller. The last part introsuces the mobile terminal project, emphasize the relevant methods that should be adopted to guarantee QoS, provide the concrete design process and the main data structure and algorithm; The latter end gives the conclusion of the whole thesis and provides the techniques should be further researched next step.
The project that this thesis bases on has been finished and the prouduct has been put into market.
引文
[1]戴维森.高艳.VoIP技术构架.第2版.北京.人民邮电出版社.2008年4月.23-35页
[2]王琦,马跃.VoIP中为保证语音质量所采用的关键技术.中国数据通信.2002年.第2期.25-290页
[3]程时瑞.IP QoS技术的现状和发展趋势.中兴通讯技术.2003年.第4期.25-60页
[4]黄永峰,IP网络多媒体通信技术.人民邮电出版社.2003.50-80页
[5]张宇.VoIP若干关键技术的研究.浙江大学电子科学与技术系.2001.50-90页
[6]贺贵明.通信原理概论.华中科技大学出版社.2002.80-90页
[7]张磊.VoIP语音技术及应用.机械工业出版社.2000.70-79页
[8]李建东,盛敏,刘乃安.宽带无线IP技术与系统.中兴通信技术.2002年.第1期.45-48页
[9]杨帆,朱光喜.宽带无线IP系统中基于区分服务的QoS实现策略.通信技术.2002年.第9期.73-75页
[10]吴春雷,周忠丽,邓文红.基于软交换体系下的VoIP分析与实现.中国民航飞行学院学报.2005年.24-28页
[11]Gil Held.语音与数据网络组网.人民邮电出版社.2003.45-68页
[12]Danial Collins.舒华英,李勇.北京.人民邮电出版社.2003.50-80页
[13]FeliPeAlvarez-Cuevas,MiquelBerrtan,FrancescOller,randJosePhM.Seiga.Voicesynchronization in Packet switching networks.IEEENewtokrsMagazine.SePtember1993.20-25
[14]TadeusUhl.Qualityof ServieeinVoIPCommunieation.InternationalJournal of Electroniesand Communieations.2004.178-182
[15]Eustathia Ziouva,Theodore AntonakoPoulos.A Dynamieally Adaptable Polling Scheme for Voice Supporting.IEEES02.11Networks.ComputerCommunications.2003.129-142
[16]Geneva,Switzerland.ITU-T Rec.H.323.Visual telephone systems andequipment for local area networks which Provid e a non-guaranteed quality of serviee.1996.29-40
[17]Video codec fot audiovisual services at 64 kbit.ITU-T Recommendation H.261.1993.60-240
[18]Bill Douskalis.The integration of robust VOIP Service.Pearson Edueation India.1999.23-58
[19]S.Paskalis,A.Kaloxylos,E.Zervas,L.Merakos.AnefficientRSVP-Mobile IP Internet working scheme.2003.50-68
[20]C.Perkins.IP Encapusulation within IP.2003.34-68
[2]王琦,马跃.VoIP中为保证语音质量所采用的关键技术.中国数据通信.2002年.第2期.25-290页
[3]程时瑞.IP QoS技术的现状和发展趋势.中兴通讯技术.2003年.第4期.25-60页
[4]黄永峰,IP网络多媒体通信技术.人民邮电出版社.2003.50-80页
[5]张宇.VoIP若干关键技术的研究.浙江大学电子科学与技术系.2001.50-90页
[6]贺贵明.通信原理概论.华中科技大学出版社.2002.80-90页
[7]张磊.VoIP语音技术及应用.机械工业出版社.2000.70-79页
[8]李建东,盛敏,刘乃安.宽带无线IP技术与系统.中兴通信技术.2002年.第1期.45-48页
[9]杨帆,朱光喜.宽带无线IP系统中基于区分服务的QoS实现策略.通信技术.2002年.第9期.73-75页
[10]吴春雷,周忠丽,邓文红.基于软交换体系下的VoIP分析与实现.中国民航飞行学院学报.2005年.24-28页
[11]Gil Held.语音与数据网络组网.人民邮电出版社.2003.45-68页
[12]Danial Collins.舒华英,李勇.北京.人民邮电出版社.2003.50-80页
[13]FeliPeAlvarez-Cuevas,MiquelBerrtan,FrancescOller,randJosePhM.Seiga.Voicesynchronization in Packet switching networks.IEEENewtokrsMagazine.SePtember1993.20-25
[14]TadeusUhl.Qualityof ServieeinVoIPCommunieation.InternationalJournal of Electroniesand Communieations.2004.178-182
[15]Eustathia Ziouva,Theodore AntonakoPoulos.A Dynamieally Adaptable Polling Scheme for Voice Supporting.IEEES02.11Networks.ComputerCommunications.2003.129-142
[16]Geneva,Switzerland.ITU-T Rec.H.323.Visual telephone systems andequipment for local area networks which Provid e a non-guaranteed quality of serviee.1996.29-40
[17]Video codec fot audiovisual services at 64 kbit.ITU-T Recommendation H.261.1993.60-240
[18]Bill Douskalis.The integration of robust VOIP Service.Pearson Edueation India.1999.23-58
[19]S.Paskalis,A.Kaloxylos,E.Zervas,L.Merakos.AnefficientRSVP-Mobile IP Internet working scheme.2003.50-68
[20]C.Perkins.IP Encapusulation within IP.2003.34-68