一种新的基于网络丢包率的主动队列管理策略

提出了一种新的基于网络丢包率的动态自适应的主动队列管理的改进算法。该算法首先用早期网络的丢包率标记到达的数据包,并作为数据包的丢弃概率。这样使得到达数据包的丢弃概率逼近当前实际网络的丢包率,然后经过自适应调整丢弃概率使得缓冲队列长度保持在一定范围内。通过NS仿真实验表明了该算法可以很好的控制队列长度,降低延迟抖动。     

一种基于模糊逻辑的主动队列管理算法

主动队列管理 (ActiveQueueManagement,AQM)技术作为Internet拥塞控制的一种有效方法 ,对于提高In ternet的服务质量具有十分重要的作用 .本文根据TCP拥塞控制算法基于数据包丢失的窗口变化机制 ,设计了一种基于模糊逻辑的主动队列管理算法 .该算法依据路由器中队列长度的变化情况 ,根据一定的模糊自校正原则来调整数据包的丢弃概率 ,从而使路由器中的队列长度稳定在参考值附近 .仿真结果表明该算法不但十分有效 ,而且对不同的网络状况具有很好的适应能力 .     

一种基于模糊逻辑的主动队伍管理算法

主动队列管理（AciiveQueue Management,AQM)技术作为Internet拥塞控制的一种有效方法，对于提高Internet的服务质量具有十分重要的作用，本文根据TCP拥塞控制算法基于数据包丢失的窗口变化机制，设计了一种基于模糊逻辑的主动队伍管理算法，该算法依据路由器中队列长度的变化情况，根据一定的模糊自校正原则来调整数据包的丢弃概率，从而使路由器中的队列长度稳定在参考值附近，仿真结果表明该算法不但十分有效，而且对不同的网络状况具有很好的适应能力。     

一种基于负载的公平性主动队列管理算法

针对随机早期检测（RED,Random Early Detection）算法存在的公平性问题,提出了一种基于负载的公平性主动队列管理算法（LFED）。该算法通过引入网络负载和队列这两个概念,有效地判断当前网络拥塞程度,同时使用改进的丢包率公式,并且借鉴CHOKe算法的惩罚机制对非响应流进行有效惩罚,以保证不同数据流之间的公平。仿真实验表明,与RED和CHOKe相比,LFED可以得到稳定的瞬时队列长度和减少分组丢弃率,具有较好的公平性和稳定性。     

基于REDP算法的动态QoS时延保障

针对无线多跳自组织网络中随机网络编码,之前提出了一种网络编码随机优先级检测调度算法(Random early detection with priority,REDP),对两条不同优先级的数据流建立Markov数学模型,推导出了引入REDP算法下数据流的单节点的时延数学表达式,该算法使得数据流能够根据自身优先级和拥塞程度调整丢包概率,从而主动丢弃数据包提前避免拥塞。文中将单节点时延问题拓展到多节点的网络拓扑结构中,提出了动态时延分配算法,该算法能够根据各个节点队列的缓存情况,动态调整各个节点的时延分配,仿真结果表明,动态时延分配算法能够实现实时数据流严格的Qo S时延保障,更利于视频、语音等实时业务流的传输。     

一种适用于多用户OFDM系统下行链路的跨层设计调度算法

该文提出了一种适用于多用户正交频分复用(OFDM)系统下行链路的调度算法,基于跨层结构设计,并分别考虑了媒体接入控制(MAC)层的用户服务质量(QoS)要求、用户公平性、数据包队列状态信息以及物理层的信道状态信息(CSI)和功率约束等。多用户OFDM系统可以通过该调度算法充分利用MAC层和物理层的不同参数特性,在保证用户QoS的同时最大化频谱的利用效率。仿真结果表明,在小区内各个移动用户具有不同的接收SNR的情况下,该算法具有服务质量随信噪比波动平缓、丢包率低、实现复杂度低等特点。     

基于EPON承载IPTV业务的加权调度算法研究

针对IPTV业务数据量较大,容易产生数据包丢失和延时,设计了一种调度算法.该算法根据IPTV节目订制用户数,采用加权轮询调度机制(WRR),合理地分配IPTV下行数据流的带宽,从而有效地减少了组播数据包的丢失和延时,提高了IPTV的服务质量.通过OPNET仿真得到结果,WRR机制比没有加权重的轮询调度机制(RR)丢包率降低了86.7％,平均队列时延减少了23.4％.     

一种新的语音流队列管理调度机制

本文提出一种新的语音流队列管理调度机制,结合随机早期探测(RED)和主动丢包调度算法实现因特网语音流的队列管理和调度.采用仿真方法分析了新机制的性能特征,并与RED做了性能对比.当网络拥塞时,该算法可有效改善包转发的性能.语音质量测试表明新机制是可行的和有效的.     

一种面向无线视频流的包丢失处理算法

论文基于无线视频传输系统,讨论了在CDMA无线网络环境下终端丢包情况下的处理方法。该差错控制机制基于UDP协议包括失序差错检测、NACK差错通告、选择重传、缓存管理和请求队列管理几个部分,具有差错反应迅速、允许多次请求重传、状态信息和控制信息少的特点,通过缓存管理和请求队列管理,避免了无效的重传请求和重传数据包,进一步降低发送端和网络资源需求。实验证明,提出的方案能有效提高实时终端视频的重建质量。     

利用可变队列长度的AQM改进与研究

AQM（主动队列管理）通过在拥塞的早期阶段丢包来告知信源。AQM改进了响应流（如TCP）和非响应流（如UDP）之间的公平性,它可以带来较高的吞吐量及有效的链路。提出了可变队列长度适应RED算法,它基于每小时的可变队列长度,并且可以更有效地应对突发流。如果将当前可变队列长度与RED及ARED相比,这种方法可以提供更为平缓的平均度列长度。试验结果表明,可变队列长度适应RED比弃尾法提高了公平性,与ARED比较性能也大为改观。这样即使有突发流存在,也可以使数据包稳定地传输。     

Impact of playout buffer dynamics on the QoE of wireless adaptive HTTP progressive video

The quality of experience (QoE) of video streaming is degraded by playback interruptions, which can be mitigated by the playout buffers of end users. To analyze the impact of playout buffer dynamics on the QoE of wireless adaptive hypertext transfer protocol (HTTP) progressive video, we model the playout buffer as a G/D/1 queue with an arbitrary packet arrival rate and deterministic service time. Because all video packets within a block must be available in the playout buffer before that block is decoded, playback interruption can occur even when the playout buffer is non-empty. We analyze the queue length evolution of the playout buffer using diffusion approximation. Closed-form expressions for user-perceived video quality are derived in terms of the buffering delay, playback duration, and interruption probability for an infinite buffer size, the packet loss probability and re-buffering probability for a finite buffer size. Simulation results verify our theoretical analysis and reveal that the impact of playout buffer dynamics on QoE is content dependent, which can contribute to the design of QoE-driven wireless adaptive HTTP progressive video management.     

Improving Quality of Service and Assuring Fairness in WLAN Access Networks

As public deployment of wireless local area networks (WLANs) has increased and various applications with different service requirements have emerged, fairness and quality of service (QoS) are two imperative issues in allocating wireless channels. This study proposes a fair QoS agent (FQA) to simultaneously provide per-class QoS enhancement and per-station fair channel sharing in WLAN access networks. FQA implements two additional components above the 802.11 MAC: a dual service differentiator and a service level manager. The former is intended to improve QoS for different service classes by differentiating service with appropriate scheduling and queue management algorithms, while the latter is to assure fair channel sharing by estimating the fair share for each station and dynamically adjusting the service levels of packets. FQA assures (weighted) fairness among stations in terms of channel access time without decreasing channel utilization. Furthermore, it can provide quantitative service assurance in terms of queuing delay and packet loss rate. FQA neither resorts to any complex fair scheduling algorithm nor requires maintaining per-station queues. Since the FQA algorithm is an add-on scheme above the 802.11 MAC, it does not require any modification of the standard MAC protocol. Extensive ns-2 simulations confirm the effectiveness of the FQA algorithm with respect to the per class QoS enhancement and per-station fair channel sharing     

Resource allocation for downlink multiuser video transmission over wireless lossy networks

Demand for multimedia services, such as video streaming over wireless networks, has grown dramatically in recent years. The downlink transmission of multiple video sequences to multiple users over a shared resource-limited wireless channel, however, is a daunting task. Among the many challenges in this area are the time-varying channel conditions, limited available resources, such as bandwidth and power, and the different transmission requirements of different video content. This work takes into account the time-varying nature of the wireless channels, as well as the importance of individual video packets, to develop a cross-layer resource allocation and packet scheduling scheme for multiuser video streaming over lossy wireless packet access networks. Assuming that accurate channel feedback is not available at the scheduler, random channel losses combined with complex error concealment at the receiver make it impossible for the scheduler to determine the actual distortion of the sequence at the receiver. Therefore, the objective of the optimization is to minimize the expected distortion of the received sequence, where the expectation is calculated at the scheduler with respect to the packet loss probability in the channel. The expected distortion is used to order the packets in the transmission queue of each user, and then gradients of the expected distortion are used to efficiently allocate resources across users. Simulations show that the proposed scheme performs significantly better than a conventional content-independent scheme for video transmission.     

Managing QoS requirements for video streaming: from intra‐node to inter‐node

Streaming video over IP networks has become increasingly popular; however, compared to traditional data traffic, video streaming places different demands on quality of service (QoS) in a network, particularly in terms of delay, delay variation, and data loss. In response to the QoS demands of video applications, network techniques have been proposed to provide QoS within a network. Unfortunately, while efficient from a network perspective, most existing solutions have not provided end‐to‐end QoS that is satisfactory to users. In this paper, packet scheduling and end‐to‐end QoS distribution schemes are proposed to address this issue. The design and implementation of the two schemes are based on the active networking paradigm. In active networks, routers can perform user‐driven computation when forwarding packets, rather than just simple storing and forwarding packets, as in traditional networks. Both schemes thus take advantage of the capability of active networks enabling routers to adapt to the content of transmitted data and the QoS requirements of video users. In other words, packet scheduling at routers considers the correlation between video characteristics, available local resources and the resulting visual quality. The proposed QoS distribution scheme performs inter‐node adaptation, dynamically adjusting local loss constraints in response to network conditions in order to satisfy the end‐to‐end loss requirements. An active network‐based simulation shows that using QoS distribution and packet scheduling together increases the probability of meeting end‐to‐end QoS requirements of networked video. Copyright © 2005 John Wiley & Sons, Ltd.     

Scalable video transmission over multi-hop wireless networks with enhanced quality of experience using swarm intelligence

In this work, we take the advantages of the particle swarm optimization method which belongs to the family of swarm intelligence algorithms to find improved solutions for delivering digital video content with enhanced quality of experience to the end users over error-prone multi-hop wireless networks. In video transmission over such wireless networks, many network-based (packet loss, delay, etc.) and source-based (encoding quantization level, etc.) parameters can impair the perceived video quality. The main contributions of the proposed work are twofold. At first, an optimal bandwidth allocation framework is being developed based on the particle swarm optimization algorithm in which by incorporating an accurate video quality metric, the total weighted quality of experience of some competing video sources is being optimized. Secondly, these optimal rates have been used for differentiated quality of experience enforcement between multiple competing scalable video sources. The resulting optimal rates can be used as rate-feedbacks for on-line rate adaptation of a moderate scalable video encoder such as H.264/MPEG4 AVC. The aforementioned weight parameters are selected based on the importance of each video sequence's quality and can be associated with some previous service level agreement based prices. Some guidelines about the practical implementation of the proposed algorithm are given. Numerical analysis has been performed to validate the theoretical results and to verify the claims.     

传感器网络拥塞避免与控制的模糊AQM算法

传感器网络节点通信能力有限,当数据到达速率持续超过节点转发能力时网络会发生拥塞;传感器网络是任务型网络,对不同优先级的信息具有不同的服务质量要求.针对传感器网络信息传输的上述特性,提出了一种新的拥塞避免与控制算法FAQM（Fuzzy Active Queue Management）.该算法在综合考虑数据包的随机指数标记概率和优先级权值的基础上,建立了模糊逻辑推理系统,并以数据包丢弃因子作为参量来实现数据流的智能调控.NS2仿真实验结果表明：FAQM算法能减少高优先级数据包的丢弃率和节点间链路的时延,稳定节点队列长度,在有效避免与控制拥塞网络的同时提升网络整体QoS（Quality of Service）性能.     

Relative loss rate differentiation: performance of short‐lived TCP flows

The relative differentiated service model provides assurances for the relative quality ordering between service classes, rather than for the actual service level in each class. In this paper, we describe a relative loss rate differentiation scheme where packet drop probabilities are determined according to an active queue management (AQM) mechanism based on random early detection (RED) in a first‐in first‐out (FIFO) queue, are weighted in inverse proportion to the price that the network operator assigns to each service class. Basically, we describe a scheme where relative loss rate differentiation is incorporated directly into AQM. Most TCP flows today, particularly Web flows, can be characterized as short‐lived flows. Using simulations with short‐lived TCP flows, we show that the scheme is very effective in ensuring relative loss rate differentiation between service classes during times of network congestion. Copyright © 2004 John Wiley & Sons, Ltd.     

一种新的基于虚拟队列的无线多播网络编码调度策略

网络编码由于其传输效率高的特性,近年来在无线多播网络中得到广泛的应用。针对无线多播网络中丢包自动重传效率低的问题,该文提出一种新的基于虚拟队列中数据包到达时间的编码调度策略(CSAT)。在CSAT策略中,为了提高编码效率,采用虚拟队列来存放初始以及未被所有接收者接收到的数据包。考虑到队列的稳定性,CSAT策略按照一定的比率从主次队列选择发送;在次队列发送数据包时,结合了编码和非编码两种方式,根据数据包到达队列的先后,选取能够使较多数据包参与编码的方式发送。仿真结果表明,该文所提的CSAT编码调度策略在有效提高了数据包传输效率的同时,提高了网络的吞吐量并降低了平均等待时延。

     

Per-packet rate adaptation for wireless video

Streaming video over error-prone wireless channels is a challenge as the dynamic network conditions and slow adaptation to channel degradations may affect the quality of the streamed video. Unequal error protection (UEP) can potentially address this issue by considering the importance of each video packet and its impact on the quality of reconstructed video. This paper proposes a cross-layer UEP solution for wireless video streaming over IEEE 802.11 networks. Video packets are prioritized based on the relative importance of the video packet. UEP is achieved by adapting the link layer parameters on a per-packet basis, using inherent forward error correction and adaptive modulation capabilities of the 802.11n network. Experimental results revealed that the proposed solution achieves comparable performance to the state-of-the-art methods at a lower complexity.     

Rate-distortion based real-time wireless video streaming

This paper presents wireless video streaming techniques that exploit the characteristics of video content, transmission history, and physical layer channels to enable real-time efficient video streaming over wireless networks to a wireless client. The key contribution of the proposed video streaming techniques is the use of rate-distortion based, but simplified, low complexity packet scheduling as well as forward error correction (FEC) rate selection. To this end, we develop an optimization framework that jointly schedules the packets and selects the FEC rates. The rate-distortion optimized packet scheduling and FEC rate selection provides the optimum quality video on the receiver side albeit at a high computational cost. By some intelligent approximations, rate distortion optimized packet scheduling and FEC rate selection technique is transformed into two sub-optimal but low complexity video streaming techniques that can provide high video quality. We perform extensive simulations to understand the performance of our proposed techniques under different scenarios. Results show that, the proposed techniques improve video quality on the average by 4 dB. We conclude that significant benefits to end-user experience can be obtained by using such video streaming methods.