End-to-end testing of IP QoS mechanisms

Voice quality provides a valid metric for testing the effectiveness of quality-of-service mechanisms in preserving the quality of voice streams in IP network traffic     

IPv6环境下的端到端QoS模型

IP网络将向着IPv6的方向发展,提供可靠的网络服务并保证服务质量是未来IPv6网络需要解决的关键问题.介绍了IPv6的QoS功能,在分析比较综合服务(integrated services,IntServ)、区分服务(differentiated services,DiffServ)和多协议标签交换(multiprotocol label switching,MPLS)等服务质量模型的基础上,提出了一个基于IPv6的端到端QoS模型,探讨了模型的功能组成和工作过程,指出了进一步研究的方向.     

ITRA: Inter-Tier Relationship Architecture for End-to-end QoS

The importance of guaranteed end-to-end quality of service (QoS) increases as business-to-business (B2B) interactions become increasingly sophisticated, lengthy, and involve more sites. The collaboration between tiers to supply functionality can be extended to provide QoS. In this paper, we focus on end-to-end availability. In multi-tier computation, support for transparent failovers is often required, even if failures occur in more than one tier at the same time. One of the obstacles in achieving a transparent failover is determining the status of outstanding operations, some of which span several tiers. Such determination of outstanding operations status upon failure requires collaboration of neighboring tiers. In this paper, we present inter-tier relationship architecture (ITRA). ITRA describes mechanisms, the role of each tier with respect to its predecessor and successor tiers, programming model and inter-tier relationship protocol. eServices following the ITRA architecture can collaborate to transparently recover from failures in multiple tiers, as well as better exploit mutual resources to provide the required availability and failover transparency aspects of an end-to-end QoS. We have exercised the architecture by mapping it to enterprise java beans (EJB) and implemented a prototype.     

基于WiMAX的VoIP的QoS改进

为了提高无线资源的利用率,使WiMAx系统更好地支持语音业务,基于IEEE802．16e协议的QoS调度体系和已有的信息流优先级（Traffic Priority）,引入新的信息流优先级乘数（Tra所cPriority Multiplier）概念,提出使用高优先级的BE服务流承载SIP信令,以取代目前普遍使用nrtPS承载SIP信令的做法。通过对优先级乘数的不同设置,可以使WiMAX运营商更加灵活地分配有限的无线资源,从而提供更好的网络和语音服务。     

基于QoS量化用户体验质量的评价模型

针对网络环境中业务供应商提供的业务不能很好的满足用户需求的问题,引入用户体验质量（QoE）并结合服务质量（QoS）参数,通过仿真网络得到相应数据,使用matlab工具分别建立用户体验质量与比特率、用户体验质量与丢包率的评价模型.运用统计分析方法对评价模型进行数据分析,并与史蒂文斯幂定律、韦伯-费希纳定律等模型进行对比验证.结果表明,构建的评价模型能更精确的体现用户体验质量与服务质量的关系,为供应商提供了重要的参考依据,并指明了服务方向.     

基于Wi—Fi的VoIP语音质量测试方案的分析与研究

着重分析在目前测试VoIP语音质量的基础上,如何应用已有方法,进行对基于Wi—Fi的网络语音电话的测试,并对这些测试方案进行对比,所研究出的客观的测试方案可以有效推动IP语音技术与无线网络技术的发展。     

提高IP电话QoS的综合去抖动同步策略研究

在提供IP电话服务时,时延抖动是一个重要的QoS参数。提出的一种去抖动同步策略综合考虑了如何能达到消除延时抖动的影响,同时又能保证良好的播放实时性。该策略将基于播放时间的同步去抖算法和基于接收缓存数据量控制的再同步算法有机结合并改进,从中获得最佳同步调整,同时还实现了对网络状况的自适应估计。实验证明接收方使用该综合去抖动同步策略能保证接收端语音流平稳连续播放,能使得IP电话的QoS得到很大改善。     

加密MANET端到端流的推断与识别研究

根据数据流的帧发送时序特性, 提出一种基于时序匹配和关联分析的端到端流推断算法。采用时序匹配、关联分析、流追踪与回溯等机制, 分析与追踪探测区域内的所有端到端流。该算法能够在无须解密报文的前提下进行流量分析, 推断出端到端流的源、目的节点、流路径、流开始时间和流持续时长等信息。仿真实验表明, 算法可以有效识别网络中的端到端流。     

Skype发送模型及其应用

为了提高Skype语音流的检测效率,在分析Skype流量特征和负载特征的基础上,提出了Skype语音源的发送模型.该模型体现了Skype发送端帧结构的特征,大大简化了接收端识别的过程.     

VoIP服务质量管理机制及原型系统

In this paper, we briefly introduce the architecture of VoIP application at the beginning. Then we focus on discussing QoS management mechanisms in data plane and control plane in detail. We explore the implementation of QoS management system by describing the prototype system QNet. At last, we discuss their future trends and present several issues remained to be studied by combining the experiences of our research.     

一种基于行为模式的Skype流量识别方法

Skype是一种基于P2P技术的VoIP客户端,其通讯协议不公开,且通讯内容加密,因此对Skype的流量识别不能采用传统的端口识别法及特征字检测法。首先对Skype的通信机制进行深入的探讨,并通过实际的数据包分析总结出Skype流量的行为模式,最后设计并实现了相应的识别模块对结论进行验证。     

Multipoint-to-point communications for SHE surveillance with QoS and QoE management

The bandwidth-hunger applications of SHE (Smart Home Environment) can take advantage of the multipoint-to-point (MPP) connections to aggregate more bandwidth to gain user-perceived Quality of Experience (QoE) and network Quality of Service (QoS). The receiver-centric transport-layer R²CP (Radial Reception Control Protocol) was proposed to resolve the incapability of the MPP communication in conventional TCP and UDP. However, R²CP has no consideration to discriminate the importance in a packet payload which is critical to QoE and brings an issue for critical data packets that may be dropped in great risk of network congestion. In this paper, we thus present P-R²CP (Prioritized R²CP) to effectively decrease the loss ratio of critical data packets in MPP video streaming while the network is congested. P-R²CP is a cross-layer protocol that considers both the transport-layer issues and the media content’s properties in application-layer. Then, an example on MPP-UVS (MPP ubiquitous video surveillance) is presented as UVS is now a very important Internet application that requires QoS/QoE management to protect lives and assets especially in SHE. Our experiments are conducted on different kinds of surveillance videos over MPP links with different bandwidth and packet loss inserted. The experimental results demonstrate that, as the loss of critical packets is decreased by an order and much less critical data packets are dropped, P-R²CP can highly guard not only QoS but also QoE of SHE surveillance video streaming.     

VoIP网关QoS模块的设计与实现

从如何改善VoIP网关语音服务质量出发,分析了目前可行的QoS方案,并提出了一种支持区分服务的解决方案。该方案结构清晰、扩展性强,解决了VoIP网关没有服务质量控制的问题,为VoIP网关QoS模块设计与实现提供了有效的参考。该方案目前已成功运用于基于嵌入式Linux平台的VoIP网关中,能够明显改善语音的服务质量。     

Skype安全性分析

Skype正越来越普及,并受到越来越多的关注。文章从通信安全的角度来分析Skype的安全特性,介绍了Skype通信加密的过程,讨论Skype使用加密的具体技术细节,在此基础上分析鉴定用户服务器身份以保护通信信息的安全,分析了几种常见攻击对Skype的可行性,对Skype的通信安全进行概括,并提出了使用中应注意的安全事项。     

支持VoIP电话紧急呼叫的QoS保障方案及其仿真

紧急呼叫在Internet上是一个非常重要的任务,在有限带宽情况下尽可能多地接受VoIP呼叫是必要的;但同时为紧急呼叫提供高质量的服务也是必须的。本文基于一种保证端到端的VoIPQoS服务模型,提出了新的方案。该方案基于DiffservAF服务模型、SIP信息段和一种通过观察VoIP数据流信息而设计的呼叫接纳信令,来为紧急VoIP提供高质量服务。最后,根据网络仿真结果性能指标,验证提出方案的合理性、可靠性。     

一种VoIP软交换系统QoS策略控制器的设计和实现

QoS策略控制器是VoIP软交换系统中的一个重要部件．它与目录服务器之间通过LDAP协议进行通信，访问目录服务器并作出决策．同时它又通过COPS协议与支持QoS的路由器建立联系，通知它所作出的决策．于是，在系统内实现了服务质量控制策略．本文详细地给出了该QoS策略控制器的设计以及具体的实现．     

广域专网中VoIP电话的QoS保障

简要分析了广域专网中的VoIP服务质量问题,对QoS的概念、功能、模型进行了阐述,提出了在广域专网中的Q-oS保证策略,给出了配置实例.     

Detailed Analysis of Skype Traffic

Skype is beyond any doubt the VoIP application in the current Internet application spectrum. Its amazing success has drawn the attention of telecom operators and the research community, both interested in knowing its internal mechanisms, characterizing its traffic, understanding its users' behavior. In this paper, we investigate the characteristics of traffic streams generated by voice and video communications, and the signaling traffic generated by Skype. Our approach is twofold, as we make use of both active and passive measurement techniques to gather a deep understanding on the traffic Skype generates. From extensive testbed experiments, we devise a source model which takes into account: i) the service type, i.e., SkypeOut calls or calls between two Skype clients, ii) the selected source Codec, iii) the adopted transport layer protocol, and iv) network conditions. Leveraging on the use of an accurate Skype classification engine that we recently proposed, we study and characterize Skype traffic based on extensive passive measurements collected from our campus LAN.     

基于UMTS的QoS结构及管理策略的研究

Along with the high speed development of mobile systems communications, the research of third generation of mobile systems are in progress rapidly, UMTS is one of the much representative schemes in the global.The paper discusses the architecture and server function of the UMTS and UMTS QoS. Based on this, the paper studies emphatically UMTS QoS architecture, Structure Components, mechanism and Management scheme of each part and arithmetic.