区分服务网络基于测量的接纳控制方案的设计与应用

提出了一种分布式可扩展的接纳控制方案，其目的是为区分服务网络提供端到端服务质量（Quality of Services,QoS)保证，该方案主要由以下部分构成：（1）连接接纳控制协议，主要负责主机和网络节点以及网络节点和带宽代理之间的信息传送，实现对连接请求的串行操作；（2）位于网络核心节点的可用带宽估计算法；（3）位于网络边缘节点的接纳控制算法；给出了该方案在视频传输方面的应用实例；利用多种网络拓扑结构和QoS指标评价该方案的性能，实验结果表明该方案能准确地控制可接纳区域和提高网络资源的利用率。     

一种基于测量的动态公平接纳控制算法

接纳控制算法是接纳控制的核心,在基于参数和基于测量的这两类接纳控制算法中,基于测量的接纳控制无需知道应用的流量模型,其次它能动态适应网络的负载变化,提高网络资源的利用率;然而最新的研究表明,仅仅追求高接纳率的接纳控制算法会造成对不同带宽QoS需求的连接的接纳非公平性,并且历经多跳的流更不易被接纳;文章的目的是解决基于测量的接纳控制算法的非公平性问题,文中分析了基于测量的接纳控制的基本思想和非公平性产生的原因,并在此基础上提出和实现了一种基于测量的HR接纳控制算法;仿真结果表明,该算法在以降低很小的接纳率的代价下,能有效地克服基于测量类接纳控制算法的非公平性问题。     

基于QoS的分布式接纳控制算法

随着人们对移动自组网(ad hoc网络)研究的深入,服务质量(QoS)已经成为了当前的研究热点之一.为了在ad hoc特殊的网络中提供对QoS的支持,必须设计一种针对流的高效的接纳控制算法.通过分析无线信道的特点,提出了估算本地可用带宽和流可用带宽的方法,同时对流的带宽耗费进行了推导,解决了实现接纳控制必须解决的关键问题,提出了一种基于QoS的分布式接纳控制算法.仿真实验证明,接纳控制算法应用到传统的路由协议(比如AODV)中,性能将会大大提高,并且更适合于实时业务的传输.     

WiMAX网络中信噪比与带宽结合的接纳控制算法

呼叫接纳控制（CAC）是宽带无线接入网络无线资源管理中一个重要的组成部分,其中策略设计的好坏直接影响着网络的性能和资源利用率。针对现有基于WiMAX的接纳控制算法仅考虑了带宽作为接纳控制的标准,提出一种结合带宽和信噪比的接纳控制算法,在考虑小区带宽因素的同时能兼顾考虑接纳后业务的信噪比情况。仿真结果表明,结合带宽和信噪比的接纳控制算法可以有效的降低系统中呼叫的掉线率,提高系统性能。     

基于IEEE 802.16的接纳控制机制

分析IEEE 802.16的QoS机制,针对其没有详细规定接纳控制算法的情况,结合协议中定义的业务类型QoS特性,在信令交互机制下,提出一种基于优先级的动态带宽分配接纳控制机制。仿真结果表明,与预留带宽的固定带宽分配机制相比,采用该机制后,系统的切换连接掉线率、新增连接的阻塞率和带宽利用率得到改善。     

QoS接纳控制算法的研究与比较

在下一代因特网中,网络服务质量的提供将成为人们关注的热点。要高效、高质量地传输各类业务,就必须通过一个有效的接纳控制机制来保证。本文介绍目前存在的两类QoS接纳控制算法：基于模型的接纳控制算法(Model Based Connection Admission Control)和基于测量的接纳控制算法(Measurement-Based Connection Admission Control),分析比较了它们的优缺点,并在此基础上提出了一个基于测量的接纳控制算法基本框架。     

一种可分片预留接纳控制算法研究

接纳控制算法的好坏直接影响分布式计算中资源提前预留机制的总体性能.针对现有灵活资源预留接纳控制算法的优缺点,提出了一种可分片预留接纳控制算法.当无法实现固定资源预留时,该算法在保证最大分片间隔的前提下,允许对资源进行分片预留;在各分片中,若存在剩余资源量小于请求预留资源量的时隙,允许用最小资源量进行预留.通过与3种可拓展预留接纳控制算法(缩短持续时间,增大预留带宽(shorten the duration and increase the reserved bandwidth,SDIB);减小预留带宽,延长持续时间(reduce the reserved bandwidth and extend the duration,RBED);改变预留的开始时间(change the reserved start time,CST))的对比实验,从接纳率和有效资源利用率方面进行了评估.实验结果表明,可分片预留接纳控制算法能有效减少资源碎片,具有更优的综合性能.     

AAL2分组话音复接器连接允许控制算法的研究

研究了AAL2分组话音复接器连接允许控制算法,得出结论：对于无比特丢充的AAL2分组话音复接器,当ATMVC输出速率较高时,按平均速率分配带宽基本上可以满足分组话音业务服务质量的要求;适当降低ATM VC的带宽利用率（例如：令ρ=0.9),则可以进一步提高话音质量,获得令人满意的话音;而当ATM VC的带宽利用率ρ=0.8时,可以获得较高质量的话音。计算机仿真证实上述结论是正确的。最后,提出了一种AAL2分组话音复接器连接允许地控制算法,按此算法,在保证分组许音业务服务质量要求得到满足的前提下,可以方便地控制控入AAL2分组话音复接器的话音源个数。     

WLAN中基于混合模式的接纳控制算法

针对WLAN标准IEEE 802.11e中 EDCA不能提供定量的服务质量(QoS)的问题,提出了一种混合模式(基于模型和测量)的接纳控制算法。通过建立退避实例的各个状态转移的Markov模型,利用贝塞尔削减规则得出网络性能指标的解析表达式,并根据测量的信道的实时状况预测新业务流可获得的吞吐量,最后提出了一种基于吞吐量的接纳控制算法。实验结果表明,该算法保证了已经接入业务流的服务质量,同时接纳了更多的新业务流,提高了网络的吞吐量。     

基于冲突概率调控的无线局域网接纳控制

本文针对基于EDCA协议的无线局域网实时多媒体系统,提出一种基于用户冲突概率调控的接纳控制算法.并在实际802.11系统参数下进行仿真,仿真结果表明该接纳控制算法可以在统计意义上保证各类业务的服务质量要求,并有效利用系统资源.     

Admission control for media on demand services

Admission control software is used to make accept or deny decisions about incoming service requests to avoid overload. Existing media streaming software includes only limited support for admission control by allowing for predefined static rules. Such rules limit for example the number of requests that are allowed to enter the system during a certain time or define thresholds concerning the utilization level of a single resource such as network bandwidth. In media streaming applications, however, the bottleneck resource (CPU, Disk I/O, network bandwidth, etc.) might change over time depending on the current demand for different types of audio or video files. This paper proposes a model for adaptive admission control in the presence of multiple scarce resources. Opportunity costs for a service request are determined at the moment of an incoming request and compared to the revenue of a request in order to make an accept/deny decision. Opportunity costs are based on resource utilization, service resource requirements, expected future demand for services, and the revenue per accepted service. The model allows rejection of service requests early to reserve capacity required to perform future service requests with higher revenues. We describe a number of experiments to illustrate the benefits of adaptive admission control models over static admission control rules.     

Enhancing aggregate QoS for video streaming

The IETF's Differentiated Services architecture is designed to provide different types or levels of service for Internet traffic. One of its key features is that traffic flows are aggregated so that routers in the core network only need to distinguish a relatively small number of aggregate flows, even if those flows consist of hundreds or thousands of individual flows. However, network-level QoS differentiation may not satisfy the requirements of many QoS-sensitive applications such as live video streaming, where end-to-end delay and reliability must be guaranteed.

This paper investigates application-level service differentiation for MPEG video streaming in a Diff-Serv-aware MPLS network infrastructure, along with routing support at the network layer. Based on the fact that MPEG video has become one of the most popular formats for Internet (wired and wireless) users, our approach enhances aggregate QoS for video streaming by employing existing application-level knowledge of the MPEG video structure; therefore it requires neither a new video compression algorithm nor additional bandwidth. Several MPEG video dispersion models are proposed and analyzed. Simulation results show that two such strategies are superior to the other tested. In addition, a new routing scheme is proposed to support searching the ‘best’ paths for efficient multi-path video streaming. Through extensive simulations, we demonstrate that our approaches improve the aggregate QoS of MPEG video streams, as well as overall network efficiency.     

iGridMedia: The system to provide low delay peer-to-peer live streaming service over internet

Commercial peer-to-peer live streaming services are getting popular nowadays, however, almost all systems target non-interactive applications. Few systems can address how to support interactive applications (such as online auction, person interview, video sharing & commenting, etc.) with peer-to-peer streaming technology. In an interactive channel, all or part of the participants can interact with the presenter or publisher at the source. Unlike existing P2P streaming service, there is different level of delay and synchronization requirements in interactive applications, meanwhile, the participant number in each channel is relatively small but the concurrent channel number is large. With such challenges, how much server bandwidth can be saved in P2P streaming is an interesting problem. We propose a very practical protocol iGridMedia and fully implement it. The basic tradeoff between consumed server bandwidth and the required delay is carefully studied. Simulation and real-world experiments show that our system consumes little server bandwidth under low delay even if the peer churn rate is very high and the peer bandwidth resource index is very limited.     

基于网络编码的对等网流媒体传输模型和算法

在对等网(peer-to-peer,简称P2P)流媒体系统中,节点(用户)的输出带宽(上行带宽)容量利用率的提高能够降低服务器的带宽开销.网络编码可以实现组播的最大吞吐率,因而具有提高系统中节点输出带宽容量利用率的潜力.将随机线性网络编码应用到P2P流媒体系统中,建立了基于随机线性网络编码的P2P流媒体传输过程模型,并据此建立传输算法的优化模型,比较研究了贪婪式算法、最少者优先算法和随机算法等.优化结果表明,随机算法可以平等均匀地获取数据包,能够最充分地利用节点的输出带宽容量,降低服务提供商的运营成本.通过对优化模型解的分析对实际系统中的传输算法给出了设计指导原则.     

应用层组播节点流媒体调度算法

针对终端转发机制存在的延迟问题和传输率波动问题,提出一种应用层组播节点流媒体调度算法,通过采用组播节点的缓冲延迟和视频编码预测信息,减弱流视频的突发性高带宽需求,从而实现实时流率平滑传输,以提供稳定的流视频服务。仿真实验结果表明,该算法是有效可行的。     

Server side,play buffer based quality control for adaptive media streaming

Existing media streaming protocols provide bandwidth adaptation features in order to deliver seamless video streams in an abrupt bandwidth shortage on the networks. For instance, popular HTTP streaming protocols such as HTTP Live Streaming (HLS) and MPEG-DASH are designed to select the most appropriate streaming quality based on client side bandwidth estimation. Unfortunately, controlling the quality at the client side means the effectiveness of the adaptive streaming is not controlled by service providers, and it harms the consistency in quality-of-service. In addition, recent studies show that selecting media quality based on bandwidth estimation may exhibit unstable behavior in certain network conditions. In this paper, we demonstrate that the drawbacks of existing protocols can be overcome with a server side, buffer based quality control scheme. Server side quality control solves the service quality problem by eliminating client assistance. Buffer based control scheme eliminates the side effects of bandwidth based stream selection. We achieve this without client assistance by designing a play buffer estimation algorithm. We prototyped the proposed scheme in our streaming service testbed which supports pre-transcoding and live-transcoding of the source media file. Our evaluation results show that the proposed quality control performs very well both in simulated and real environments.     

基于预测机制的云媒体网络自适应视频流选择算法

研究移动网络中,如何进一步提高云媒体服务带宽利用率的问题,提出一种基于预测机制的云媒体网络自适应视频流选择算法。该算法包含可用带宽预测模型和视频流决策模型。在带宽预测模型中,根据带宽评估结果把当前的网络状态划分为三个模式,每个模式对应不同的带宽预测方法,并且不同的模式能够相互转化。在视频流决策模型中,利用预测的可用带宽,结合可伸缩视频编码技术为用户自适应的调整视频质量。根据实验,在不同网络环境下,该方法能够有效的利用带宽,并提高媒体服务质量。     

移动网格中负载自平衡的集群式边缘流媒体服务器

迅速发展的3G网络和覆盖范围可达整个城域的Wi-Fi网络的逐渐普及,使移动网络环境中对流媒体服务的需求迅速增长.由于移动用户主要依赖Internet上的流媒体服务资源提供服务,所以位于无线网络和Internet交界处的边缘流媒体服务器可以充当桥梁和缓冲区,对于降低网络负载和提高服务质量有显著的效果,因而提出一种新的集群架构的边缘流媒体服务器(CESS)的设计,而且针对集群服务器中最重要的负载均衡问题做了分析和测试,提出了一种新的MCLBS缓存替换算法来使CESS能够达到自适应的负载均衡.最后,实验测试和结果分析证明,相对于传统的缓冲替换算法,MCLBS算法更适合集群式的服务器体系结构,在同样的条件下,缓存命中率明显提高,大大降低远程服务器的带宽.     

基于综合因素的P2P流媒体数据调度算法

为了提高P2P流媒体数据传输的效率,提出一种基于综合因素的P2P流媒体数据调度算法,该算法综合考虑数据块的提供者数量和紧急程度定义数据块优先级,并对与提供者之间的网络带宽进行有效估计,根据优先级和带宽调度P2P数据请求的次序和方向。仿真结果表明,该调度算法在节点吞吐量和系统连续性指标以及启动延迟方面比传统的P2P流媒体调度算法有明显提高。