RSVP在数字视频网络传输QoS保障中的应用研究

随着网络数字视频的广泛应用，对实时性的要求也变得越来越紧迫。这些应用程序要求因特网协议(IP)能够提供有保证的带宽、较高的优先级以及较低的包丢失率。为了达到这一要求，Internet工程任务组(IETF)开发了一套用于因特网的综合服务的协议和标准。通过资源预留协议(RSVP)来管理网络资源分配以提供不同级别的服务。讨论了怎样将该协议与典型的数字视频网络传输方案相结合，以使现有支持服务质量(QoS)的网络可以提供更为稳定流畅的传输效果。     

A taxonomy of multicast protocols for Internet applications

The continuous growth in applications that require communications among a group of hosts or simultaneous dissemination of data to multiple sites on the Internet has led to considerable interest in multicast communication. These applications have different quality of service requirements such as deadlines to response time and tolerance to data loss. Internet multicast protocols provide a spectrum of services to cater for the needs of a wide array of multicast applications. The design alternatives of a multicast protocol for a particular application may lead to different performance characteristics, for example higher utilization of network bandwidth vs lower protocol processing requirements. Therefore a framework is required to various design choices available to design multicast protocols and evaluate the performance tradeoffs associated with these alternatives. Motivated by such a need and by the evolution of multicast applications over the Internet, this article presents a taxonomy of multicast protocols and a common framework to evaluate their performance.     

A Survey of Application Layer Techniques for Adaptive Streaming of Multimedia

Though the integrated services model and resource reservation protocol (RSVP) provide support for quality of service, in the current Internet only best-effort traffic is widely supported. New high-speed technologies such as ATM (asynchronous transfer mode), gigabit Ethernet, fast Ethernet, and frame relay, have spurred higher user expectations. These technologies are expected to support real-time applications such as video-on-demand, Internet telephony, distance education and video-broadcasting. Towards this end, networking methods such as service classes and quality of service models are being developed. Today's Internet is a heterogeneous networking environment. In such an environment, resources available to multimedia applications vary. To adapt to the changes in network conditions, both networking techniques and application layer techniques have been proposed. In this paper, we focus on the application level techniques, including methods based on compression algorithm features, layered encoding, rate shaping, adaptive error control, and bandwidth smoothing. We also discuss operating system methods to support adaptive multimedia. Throughout the paper, we discuss how feedback from lower networking layers can be used by these application-level adaptation schemes to deliver the highest quality content.     

基于混合遗传算法的多播路由多目标优化

互联网不断增长的多媒体应用引发人们研究如何满足这些应用的服务质量(QoS)约束.目前网络中多主机之间的多播通信一般需要严格的多个QoS保证,文中描述了一种适应于研究多播QoS路由多目标优化的网络模型,在此基础上提出了基于遗传算法和禁忌搜索混合策略的,具有多目标的多播路由QoS优化方法,以克服遗传算法的爬山能力差以及不成熟收敛等问题.此外还采用了改进的多播树编码方法及高效的遗传操作,同时还优化时延、丢包率和带宽利用率等不同的参数.实验结果表明,该算法为多播路由QoS多目标优化问题的求解提供了一种有效的新途径.     

基于多目标遗传算法的多播服务质量路由优化

互联网不断增长的多媒体应用引发人们研究如何满足这些应用的服务质量(QoS)约束，并导致提出了一些基于服务质量的体系结构．目前网络中多主机之间的多播通信一般需要严格的多个QoS保证．首先将满足不同约束的多播路由选择过程转化为一个多目标优化问题，然后使用一种基于多目标遗传算法的新型多播树计算方法，同时优化时延、丢包率和带宽利用率等不同的参数．实验结果表明，该方法能在有限进化代数内产生一组有效的非劣多播路由解，结合多目标优化的遗传算法克服了单目标路由优化的缺陷．     

一种TCP友好的Internet DTV组播发送速率控制机制

当前的Internet没有对流媒体应用提供足够的QoS的保证,同时Internet DTV的组播发送速率必须满足以下两点要求:1)能够自适应网络拥塞的变化;2)能够适应节目码率的要求,因此需要在服务器端进行组播发送速率控制.分析了Internet DTV组播的难点,提出了一种基于缓冲区管理的具有网络自适应特性的组播发送速率控制方法.通过合理地控制发送方的发送速率,既能自适应网络状况的变化,又能满足流媒体实时播放的需求.仿真实验结果表明,该算法在保证服务器端发送缓冲区不溢出前提下,与单纯采用TFMCC方法比较,减少了接收端的平均丢包率,提高了网络电视节目流的传输质量.     

一种基于种族分类进化的QoS异构组播路由机制

随着新型网络应用的大量涌现,传统的网络技术已无法满足当前应用在带宽、延迟及出错率等方面的需求,IP over DWDM光互联网以其独特性能优势成为研究热点。基于种族分类进化算法,提出了一种IP over DWDM光互联网服务质量QoS异构组播路由机制。具体地,利用概率论方法处理网络状态参数信息的不确定性;引入模糊数学方法,确定用户对QoS的需求并提供柔性QoS支持;综合考虑网络提供方和用户方的利益,设计公平的带宽定价方法。仿真结果表明,该路由机制获得了良好的综合性能指标,可以有效地解决IP over DWDM光互联网中的柔性QoS异构组播路由选择问题。     

区分服务网络中IP多播:问题与解决方案

在下一代因特网中,越来越多的应用将会需要网络提供一定的服务质量以及进行多播传输．因为区分服务体系结构提供了一种可扩展的QoS解决方案,而多播传输则提供了一种节约网络资源的有效方法,二者的集成成为必然趋势．但是,由于区分服务体系结构与多播树结构上的差异,二者的集成存在着一些问题．详细分析了区分服务网络中进行IP多播传输可能出现的问题,并且对近年来提出的各种解决方案进行了分类比较和深入剖析,同时指出了在这一领域中未来的研究方向．     

Bandwidth guaranteed multicast scheduling for virtual output queued packet switches

Multicast enables efficient data transmission from one source to multiple destinations, and has been playing an important role in Internet multimedia applications. Although several multicast scheduling schemes for packet switches have been proposed in the literature, they usually aim to achieve only short multicast latency and high throughput without considering bandwidth guarantees. However, fair bandwidth allocation is critical for the quality of service (QoS) of the network, and is necessary to support multicast applications requiring guaranteed performance services, such as online audio and video streaming. This paper addresses the issue of bandwidth guaranteed multicast scheduling on virtual output queued (VOQ) switches. We propose the Credit based Multicast Fair scheduling (CMF) algorithm, which aims at achieving not only short multicast latency but also fair bandwidth allocation. CMF uses a credit based strategy to guarantee the reserved bandwidth of an input port on each output port of the switch. It keeps track of the difference between the reserved bandwidth and actually received bandwidth, and minimizes the difference to ensure fairness. Moreover, in order to fully utilize the multicast capability provided by the switch, CMF lets a multicast packet simultaneously send transmission requests to multiple output ports. In this way, a multicast packet has more chances to be delivered to multiple destination output ports in the same time slot and thus to achieve short multicast latency. Extensive simulations are conducted to evaluate the performance of CMF, and the results demonstrate that CMF achieves the two design goals: fair bandwidth allocation and short multicast latency.     

安全服务质量(QoSS)的评价与控制模型研究

安全服务质量(QoSS)是评价安全系统能否满足用户安全功能需求的重要指标。论文首先给出了基于QoSS的安全服务向量的形式化定义,以方便用户表达和规范化各自的安全需求。然后,在此基础上,提出了一个安全服务质量的评价与控制模型,重点介绍了用户对安全服务满意程度和安全服务成本的计算方法,以及对QoSS进行控制的流程。     

一种分类预计算QoS路由算法

为了满足Internet上各种网络应用的传输服务质量的需求,Internet工程任务组(IETF)先后提出了集成服务/资源预留模型(integrated service/resource reservation protocol,简称IntServ/RSVP)、区分服务模型(differentiated service,简称DiffServ)以及从流量工程角度提出的多协议标记交换(multi protocol label switching,简称MPLS).这些服务模型都需要一个与之相适应的QoS路由机制     

基于流媒体相关的视频安全组播协议(MSMP)研究

随着无线网络的快速发展和Internet中流媒体视频的巨大成功,无线网络中的视频服务有望在不久的将来得到大规模部署,无线网络上的实时流媒体传输技术已成为研究热点,而其中视频安全组播协议是一个关键问题。但是,由于无线网络中有限的带宽和有限的存储空间,外部的攻击与自适应机制带来的安全性问题在无线流媒体视频中是不可避免的。一个精心设计的密钥管理算法不仅可以明显地提高流媒体视频的性能,还能够保证可靠的数据嵌入以及实时视频应用提供安全支持。如何设计一个高效的密钥管理算法,是当前流媒体视频应用中一个备受关注的问题。针对无线视频应用,针对一个处于开放和不安全的网络环境中的自适应视频应用的密钥管理算法进行了研究,并进行了算法评价。     

Real‐time multicast tree visualization and monitoring

The exponential growth of the Internet combined with the increasing popularity of streaming audio and video are pushing Internet bandwidth constraints to their limits. Methods of managing and more efficiently utilizing existing bandwidth are becoming increasingly vital. Using IP multicast to deliver content, especially streaming audio and video, can provide enormous bandwidth savings. A decade of effort at deploying multicast, combined with the rising need for better traffic management for bandwidth‐hungry applications has led to significant momentum for multicast use and deployment. One of the remaining barriers to widespread adoption is the lack of multicast monitoring and debugging tools. To address this need we introduce MHealth, a graphical, near real‐time multicast monitoring tool. MHealth utilizes existing tools to collect comprehensive data about Realtime Transport Protocol (RTP) based streaming audio/video sessions. By using a combination of application‐level protocol data for participant information and a multicast route tracing tool for topology information, MHealth is able to present a multicast tree's topology and information about the quality of received data. In this paper we describe the design and implementation of MHealth and include an example analysis of multicast tree statistics. Copyright © 2000 John Wiley & Sons, Ltd.     

Effective distributed service architecture for ubiquitous video surveillance

Video surveillance systems are playing an important role to protect lives and assets of individuals, enterprises and governments. Due to the prevalence of wired and wireless access to Internet, it would be a trend to integrate present isolated video surveillance systems by applying distributed computing environment and to further gestate diversified multimedia intelligent surveillance (MIS) applications in ubiquity. In this paper, we propose a distributed and secure architecture for ubiquitous video surveillance (UVS) services over Internet and error-prone wireless networks with scalability, ubiquity and privacy. As cloud computing, users consume UVS related resources as a service and do not need to own the physical infrastructure, platform, or software. To protect the service privacy, preserve the service scalability and provide reliable UVS video streaming for end users, we apply the AES security mechanism, multicast overlay network and forward error correction (FEC), respectively. Different value-added services can be created and added to this architecture without introducing much traffic load and degrading service quality. Besides, we construct an experimental test-bed for UVS system with three kinds of services to detect fire and fall-incident features and record the captured video at the same time. Experimental results showed that the proposed distributed service architecture is effective and numbers of services on different multicast islands were successfully connected without influencing the playback quality. The average sending rate and the receiving rates of these services are quite similar, and the surveillance video is smoothly played.     

安全组播体系结构的研究进展

伴随着互联网带宽的增加和组播技术的日渐成熟,利用组播技术进行网络视频会议、股市行情分发等大规模应用已成为现实。文中介绍了IP组播体系结构需要提供的安全服务。接着通过介绍几种代表性的组播体系结构,说明了目前组播体系结构的研究进展。最后通过分析这些体系结构,提出了建设性的观点。     

Scalable solutions for secure group communications

With the advent of digital technologies and widening Internet bandwidth in recent years there has been a marked rise in new multimedia services, including teleconferencing, pay-per-view TV, interactive simulations, software updates and real-time delivery of stock market information. Multicast data distribution has been used in controlled environments to deliver such services. However, the lack of secure, accountable multicast data distribution has prevented its use in general Internet environments. Proposals for multicast security solutions so far are complex and often require trust in intermediate components or are inefficient. A secure multicast protocol suite must provide data confidentiality and multicast packet source authentication. In this paper we present a robust, simple and efficient multicast key management protocol based on proxy encryption and a multicast data source authentication mechanism based on symmetric message authentication codes. The solutions are analyzed and compared to previously published schemes. The results show that the proposed schemes are efficient and scalable relative to existing schemes.     

Low-cost,delay-bounded point-to-multipoint communication to support multicasting over WDM networks

The confluence of technical advances and multimedia service needs is intensifying the need for high throughput and low latency. Future communication networks will face an increase in traffic driven by multimedia requirements with stringent delay and jitter requirements. Wavelength division multiplexing (WDM) optical networks have the potential for meeting these goals by offering unprecedented high bandwidth and low latency. One very important aspect of the emerging Internet services is the need to support multicasting. This is crucial if WDM networks were to play an efficient role in the next generation Internet.Multicasting in WDM networks supporting multimedia applications can be viewed as the process of taking a group communication request and selecting a multicast tree that satisfies the quality of service requirements, in terms of bandwidth and end-to-end delay, of the underlying application. In this paper, we present a new class of low-cost, bounded-delay multicast heuristics for WDM networks. The heuristics use various techniques to establish a tree of semi-lightpaths between a source and a group of destination nodes. The unique feature of these heuristics is that they decouple the cost of establishing the multicast tree from the delay incurred by data transmission due to light-wave conversion and processing at intermediate nodes along the transmission path. A simulation study shows the performance of the proposed heuristics.     

Seamless handover between unicast and multicast multimedia streams

With the deployment of heterogeneous networks, mobile users are expecting ubiquitous connectivity when using applications. For bandwidth-intensive applications such as Interact Protocol Television （IPTV）, multimedia contents are typical- ly transmitted using a multicast delivery method due to its bandwidth efficiency. However, not all networks support multicasting. Multicasting alone could lead to service disruption when the users move from a multicast-capable network to a non-multicast network. In this paper, we propose a handover scheme called application layer seamless switching （ALSS） to provide smooth real-time multimedia delivery across unicast and multicast networks. ALSS adopts a soft handover to achieve seamless playback during the handover period. A real-time streaming testbed is implemented to investigate the overall handover performance, espe- cially the overlapping period where both network interfaces are receiving audio and video packets. Both the quality of service （QoS） and objective-mapped quality of experience （QoE） metrics are measured. Experimental results show that the overlapping period takes a minimum of 56 and 4 ms for multicast-to-unicast （M2U） and unicast-to-multicast （U2M） handover, respectively. The measured peak signal-to-noise ratio （PSNR） confirms that the frame-by-frame quality of the streamed video during the handover is at least 33 dB, which is categorized as good based on ITU-T recommendations. The estimated mean opinion score （MOS） in terms of video playback smoothness is also at a satisfactory level.     

A Scalable Overlay Multicast Architecture for Large-Scale Applications

In this paper, we propose a two-tier overlay multicast architecture (TOMA) to provide scalable and efficient multicast support for various group communication applications. In TOMA, multicast service overlay network (MSON) is advocated as the backbone service domain, while end users in access domains form a number of small clusters, in which an application-layer multicast protocol is used for the communication between the clustered end users. TOMA is able to provide efficient resource utilization with less control overhead, especially for large-scale applications. It also alleviates the state scalability problem and simplifies multicast tree construction and maintenance when there are large numbers of groups in the network. To help MSON providers efficiently plan backbone service overlay, we suggest several provisioning algorithms to locate proxies, select overlay links, and allocate link bandwidth. Extensive simulation studies demonstrate the promising performance of TOMA     

一种TCP-friendly主动分层组播拥塞控制机制

针对Internet多媒体业务的快速发展对组播拥塞控制提出的要求,在对现有分层组播算法存在问题进行分析的基础上,提出了一种接收端快速自适应的TCP-Friendly主动分层组播拥塞控制机制ALMCC.它采用主动标记分层,并在接收端根据分组延时,快速的自适应网络带宽.仿真实验表明,ALMCC算法提高了分层组播拥塞控制性能,具有拥塞响应速度快、丢包率低和TCP-Friendly特性.