Performance analysis of multimedia based web traffic with QoS constraints

During the recent years, there has been a tremendous growth in the development and deployment of multimedia based networked applications such as video streaming, IP telephony, interactive games, among others. These applications, in contrast to elastic applications such as email and data sharing, are delay and delay jitter sensitive but can tolerate certain level of packet loss. A vital element of end-to-end delay and delay jitter is the random queueing delays in network switches and routers. Analysis of robust mechanisms for buffer management at network routers needs to be carried out in order to reduce end-to-end delay for traffic generated by multimedia applications. In this context, a threshold based buffer management scheme for accommodating multiple class multimedia traffic in network routers has been analysed. This technique effectively controls the allocation of buffer to various traffic classes according to their delay constraints. The forms of the joint state probabilities, as well as basic performance measures such as blocking probabilities are analytically established at equilibrium. Typical numerical experiments are included to illustrate the credibility of the proposed mechanism in the context of different quality of service (QoS) grades for various network traffic classes. This model, therefore, can be used as a powerful tool to provide a required grade of service to a particular class of multimedia based web traffic in any heterogeneous network.     

基于视频发送端速率控制的QoS技术研究

基于尽力而为的网络模式不能提供QoS保证,网络拥塞和分组丢失不可避免。在端到端视频单播结构下,论文提出了一个发送端速率控制框架SRCF,在此框架下首先利用RTCP报文中的字段提出了一种网络参数测量方法,然后设计了一个自适应速率算法SRCA,SRCA利用已得到的网络传输延迟和分组丢失率参数作为初始参数,来调整编码速率,达到充分利用带宽的目的,避免了视频质量由于调整参数带来的剧烈抖动。仿真结果表明,该算法在网络出现一定拥塞的条件下,能跟踪带宽的变化,网络和媒体QoS能保证视频质量较好。     

分布式星群网络中基于蚁群算法的通信量分类路由

给出了一种分布式星群网络(Distributed Satellite Cluster Network,DSCN)体系架构,阐明了DSCN拓扑变化的特点。在分析网络状态获取方式和路由计算方法的基础上,提出了一种适用于DSCN的基于蚁群算法的通信量分类路由(Ant Colony Optimization Based Traffic Classified Routing,ATCR)算法。ATCR算法将通信量分为时延敏感型通信量A、带宽敏感型通信量B以及提供尽力而为服务的通信量C,并对蚁群算法(Ant Colony Optimization,ACO)收敛慢的缺点进行了改进。仿真实验表明,ATCR算法提高了收敛速度,可以有效平衡网络流量。通信量A和C的端到端时延要小于未采用通信量分类的改进ACO算法。由于减少了重负载链路的数量及拥塞引起的丢包,ATCR算法在分组递交率上的表现优于改进的ACO算法。     

Rate control of multi class priority flows with end-to-end delay and rate constraints for QoS networks

To address end-to-end quality of service (QoS) requirements, we derive a novel distributed combined rate and end-to-end delay control in a network serving multi-class flows with priority packet scheduling. We show that the control is globally asymptotically stable without information time lags. The stable flows attain the end-to-end delay requirements and have no packet loss. We also show that by enhancing the network with bandwidth reservation and admission control, minimum rate is also guaranteed. The stability with very long time lags of a discrete time version control with non-greedy flows and random packet arrivals is studied numerically by an NS2 packet-based simulation of the Australian Academic and Research Network.     

Hop-count based probabilistic packet dropping: Congestion mitigation with loss rate differentiation

Network applications and users have very diverse service expectations and requirements, demanding for provisioning different levels of quality of service on the Internet. As the speed of network links has been rising at a pace that exceeds that of the growth in the buffer size, packet loss rate differentiation has been an active research topic. However, none of the existing packet dropping schemes for loss rate differentiation considered an important issue, that is, the retransmission overhead of dropped packets. In this paper, we design a hop-count based probabilistic packet dropper (HPPD) for congestion mitigation and loss rate differentiation. HPPD aims to meet a unique two-fold objective by two-dimensional loss rate differentiation: the primary one is the congestion mitigation that aims to reduce congestion in the first place by dropping intra-class packets differently based on their maturity levels to reduce retransmission cost; the other is inter-class proportional loss rate differentiation. The maturity level of a packet, the number of hops it has travelled, is inferred from its time-to-live value in the IP header. We propose a novel intra-class nth-root proportional dropping scheme. The scheme reduces retransmission cost by giving higher dropping probabilities to less mature packets while all packets have their forwarding chances. The n is a controllable parameter trading off dropping fairness for congestion mitigation. It provides great controllability to network operators. Simulation results show that HPPD can significantly mitigate the congestion by reducing the retransmission overhead of dropped packets and achieve the proportional loss rate differentiation at the same time.     

无线传感器网络中确保端到端传输质量的自适应调制缩放技术

在满足数据收集任务的时延要求及可靠性要求的前提下最小化能量消耗是传感嚣网络设计中的一个关键问题。选取不同的调制级数(即调制时的星座图大小,每符号位数)可以获得不同程度的能量节省,但引入不同程度的时延的同时也会造成不同程度的丢包。本文提出了一种优化模型在满足数据收集任务的时延要求及可靠性要求的前提下,最优选取各节点的调制级数以最小化网络能耗延长网络生命期。并在此基础之上提出了一个集中式控制算法以根据网络状况自适应地调节各节点的调制级数。实验表明本算法在所考查的各种网络环境下都能明显减少能耗,可以用该模型来确定时延及丢包率对能耗影响的敏感区。另外我们发现,如果将节点的服务率设置为与节点的包到达率近似成比例,在大多数情况下都能够获得好的节能效果。     

A Novel Framework for IP DiffServ over Optical Burst Switching Networks

This paper presents a novel framework for IP Differentiated Services (DiffServ) over optical bursts witching (OBS), namely, DS-OBS. The network architecture, functional model of edge nodes and core nodes,the control packet format, a novel burst assembly scheme at ingress nodes and scheduling algorithm of core nodes are presented. The basic idea is to apply DiffServ capable burst assembly at ingress nodes and perform different per hop behavior (PHB) electronic treatments for control packets of different QoS class services at core nodes. Simulation results show that the proposed schemes can provide the best differentiated service for expedited forwarding (EF), assured forwarding (AF) and best effort (BE) services in terms of end-to-end delay, throughput and IP packet loss probability.     

A real-time video multicast architecture for assured forwarding services

This paper presents our work on developing an architecture for multicasting real-time MPEG4 over IP networks that provide service differentiation. In particular, this work is targeted at assured forwarding (AF) style services. This work is an attempt to find a simple solution to the problem of multicast congestion control of real-time traffic by exploiting the service differentiation capabilities of AF networks. Our architecture assumes loss differentiation in the network and assumes the network's ability to provide explicit congestion notification messages to the sender. We do not consider policing/shaping at the edge routers. Rather, we consider a more general case where packet marking and flow control are provided at the senders. For this network model, we built an end-to-end architecture and developed a rate-adaptation algorithm that can operate in both unicast and multicast applications with a minor modification. The simulation results show how the rate-adaptation algorithm accommodates different receivers with different networking capabilities and provides receivers with different levels of quality by taking advantage of the queue management capabilities of the AF service. We test how the architecture scales to a large number of receivers, how multiple multicast sessions interact, and how it interacts with TCP.     

Quality-of-service mapping mechanism for packet video indifferentiated services network

This paper presents a futuristic framework for quality-of-service (QoS) mapping between practically categorized packet video and relative differentiated service (DiffServ or DS) network employing unified priority index and adaptive packet forwarding mechanism under a given pricing model (e.g., DiffServ level differentiated price/packet). Video categorization is based on the relative priority index (RPI), which represents the relative preference per each packet in terms of loss and delay. We propose an adaptive packet forwarding mechanism for a DiffServ network to provide persistent service differentiation. Effective QoS mapping is then performed by mapping video packets onto different DiffServ levels based on RPI. To verify the efficiency of proposed strategy, the end-to-end performance is evaluated through an error resilient packet video transmission using ITU-T H.263+ codec over a simulated DiffServ network. Results show that the proposed QoS mapping mechanism can exploit the relative DiffServ advantage and result in the persistent service differentiation among DiffServ levels and the enhanced end-to-end video quality with the same pricing constraint     

Design,implementation and experimental evaluation of an end-to-end vertical handover scheme on NCTUns simulator

This paper presents the simulation study of “Host based autonomous Mobile Address Translation” using NCTUns simulator. It is a network layer, end-to-end vertical handover solution, based upon modification of “Mobile IP with address Translation”. Vertical handover approaches generally require new network elements, a new layer in TCP/IP stack, or fixing a protocol at a particular layer. To enhance handover experience, recent approaches focus on reducing signalling, localizing the registration, creating hierarchies, using proxy, preparing handover in advance, predicting target network, or exploiting multicasting and path extension techniques. These approaches, however, demand change in the network infrastructure to support mobility and limit the scope of mobility. Despite end-to-end signalling, the Host based autonomous Mobile Address Translation scheme ensures minimum service disruption and distinctly allows global mobility of the mobile node without requiring any modification in the network. We have simulated the mobility of a multi-interface mobile node in a heterogeneous network environment composed of WiFi (IEEE802.11a, IEEE802.11b) and WiMAX (IEEE802.16e) access networks. Performance of the scheme is evaluated taking into account wide range of end-to-end delays between mobile node and the correspondent node, various speeds of the mobile node and different packet loss rates of the network. Based on our detailed simulation study, it has been observed that this scheme offers reduced service disruption time, packet loss and packet latency. The service disruption time is found to be significantly low (typically in the range of 10 ms) compared to that of Mobile IP (which is in the order of 100 ms); this makes this new scheme perfectly suitable for real time applications. Low service disruption time consequently reduces the packet loss by manyfold and the packet latency remains unaffected during and after handover due to translation of address at the source. The results suggest that this protocol is a viable vertical handover solution due to its simplicity, scalability, low overhead and ready deployability.     

Marking algorithms for service differentiation of TCP traffic

Class-based service architectures for quality-of-service (QoS) differentiation typically provide loss, throughput, and delay differentiation. However, proposals for class-based service differentiation generally do not account for the needs of TCP traffic, which are characterized by a coupling of packet losses and achievable throughput. Ignoring this coupling may result in poor service differentiation at the microflow level. This paper shows how Explicit Congestion Notification (ECN) can be used to achieve service differentiation for TCP traffic classes at the microflow level. We present a traffic-marking algorithm for routers, which, if used in conjunction with ECN, regulates the transmission rate of TCP sources in such a way that packet drops due to buffer overflows are avoided. We demonstrate how the algorithm can be integrated in a service architecture with absolute and proportional QoS guarantees. Simulation results illustrate the effectiveness of the presented algorithms at avoiding packet losses and regulating traffic for meeting service guarantees, and provide a comparison with other algorithms proposed in the literature.     

Joint call and packet QoS in cellular packet networks

This paper proposes a novel call admission control scheme capable of providing a combination of call and packet level quality of service requirements in cellular packet networks. Specifically, we propose a distributed call admission control scheme called PFG, which maximizes the wireless channel utilization subject to a predetermined bound on the call dropping and packet loss probabilities for variable-bit-rate traffic in a packet-switched wireless cellular network. We show that in wireless packet networks, the undesired event of dropping an ongoing call can be completely eliminated without sacrificing the bandwidth utilization. Extensive simulation results confirm that our scheme satisfies the hard constraint on call dropping and packet loss probabilities while maintaining a high bandwidth utilization.     

一种适用于网络处理器的队列管理算法

遵循比例区分服务模型,设计了一种适用于网络处理器的队列管理算法．算法包含两部分,分组入队列时实现丢失率控制的RR—PLR（round—robin based proportional loss rate）和分组出队列时实现时延控制的WRR—PAD（WRR based proportional average delay）．算法采用轮循的机制,避免了除法运算和排序操作,具有O（1）的复杂度,而且易于在网络处理器上实现．性能模拟以及实测的结果表明,该算法能有效实现平均分组丢失率和平均排队时延的比例控制,系统的总吞吐率达到了1．125Gbps（每个分组64B,即2．25Mpps）．     

End-to-end support for statistical quality of service in heterogeneous mobile ad hoc networks

In heterogeneous mobile ad hoc networks (MANETs), different types of mobile devices with diverse capabilities may coexist in the same network. The heterogeneity of MANETs makes end-to-end support for quality of service (QoS) guarantees more difficult than in other types of networks, not to mention the limited bandwidth and frequent topology changes of these networks. Since QoS routing is the first step toward achieving end-to-end QoS guarantees in heterogeneous MANETs, we propose a QoS routing protocol for heterogeneous MANETs. The proposed protocol, called virtual grid architecture protocol (VGAP), uses a cross-layer approach in order to provide end-to-end statistical QoS guarantees. VGAP operates on a fixed virtual rectilinear architecture (virtual grid), which is obtained using location information obtained from global positioning system (GPS). The virtual grid consists of a few, but possibly more powerful, mobile nodes known as ClusterHeads (CHs) that are elected periodically. CHs discover multiple QoS routes on the virtual grid using an extended version of the open shortest path first (OSPF) routing protocol and an extended version of WFQ scheduling policy that takes into account the wireless channel state. Moreover, VGAP utilizes a simple power control algorithm at the physical layer that provides efficient energy savings in this heterogeneous setting. Simulation experiments show that VGAP has a good performance in terms of packet delivery ratio, end-to-end packet delay, call blocking probability, and network scalability.     

基于改进Dijkstra算法的配用电通信网流量调度策略

针对配用电通信网中数据汇聚易产生拥塞的问题,提出了一种复合边权值流量调度路由算法。首先,依据跳数建立节点分层模型;然后,划分配用电业务优先级和节点拥塞等级;最后,以跳数、流量负载率和链路利用率为综合指标计算边权值,对需要流量调度的节点根据改进的Dijkstra算法进行路由选择,同时对重度拥塞节点按照配用电业务优先级进行调度。与最短路径（SPF）算法和贪婪背压算法（GBRA）相比,在数据生成率为80 kb/s时,所提算法紧急型业务丢包率分别减少了81.3%和67.7%,关键型业务丢包率分别减少了79%和63.8%。仿真结果表明,所提算法能有效缓解网络拥塞,提高网络有效吞吐量,降低网络端到端时延和高优先级业务的丢包率。     

Availability of end-to-end ideal QoS in IP packet networks

Network processor technology has advanced to the point where high-precision time-based store-and-forward logic is readily incorporated into packet switches and routers. With appropriate scheduling, packets from multiple flows can be serviced without contending for link resources. Accordingly, packet flows traversing a network of switching elements can have both path and time determinacy attributes which support ideal end-to-end QoS (zero jitter, zero loss, acceptable end-to-end latency) for real-time UDP packet flows and guaranteed goodput for TCP flows. One approach to packing a network with a relatively large number of such deterministic flows, i.e. achieving high availability of the ideal QoS service in a network, uses precise buffering of packets at each switch, which introduces latency. This paper describes analysis methods for quantifying how much buffering may be necessary to achieve high (99.999%) availability. For typical network topologies the analysis shows that buffering latency requirements are very small compared to transport delays, even when the network is highly utilized with heterogeneous (e.g. voice, video, circuit emulation, and data) traffic. Actual physical implementations have empirically validated the analysis results as well as the scalability of the end-to-end, time-based forwarding approach and the end-to-end availability of ideal QoS services in IP packet networks.     

基于联合优先级调度的协作动态频谱分配机制设计

在协作频谱接入过程中,共享频谱各网络中的频谱需求呈现高度动态性,现有机制难以兼顾分组传输时延和服务公平性。针对这一问题,提出一种基于联合优先级调度的协作动态频谱分配机制。将频谱分配问题建模为比例公平模型并联合网络的服务等级和用频请求的到达时间为各网络设定动态服务优先级,解得分配结果后,基于分配频谱的中心频率与需求频谱的中心频率最近的原则将频谱资源分配给各网络。仿真结果表明,与已有的基于网络固定优先级调度、基于网络动态优先级调度和基于用频请求动态优先级调度的协作频谱分配机制相比,本机制能够更好地保证网络获取频谱的公平性,并且在分组传输时延性能上有所改进。     

Congestion Adaptive Routing in Mobile Ad Hoc Networks

Mobility, channel error, and congestion are the main causes for packet loss in mobile ad hoc networks. Reducing packet loss typically involves congestion control operating on top of a mobility and failure adaptive routing protocol at the network layer. In the current designs, routing is not congestion-adaptive. Routing may let a congestion happen which is detected by congestion control, but dealing with congestion in this reactive manner results in longer delay and unnecessary packet loss and requires significant overhead if a new route is needed. This problem becomes more visible especially in large-scale transmission of heavy traffic such as multimedia data, where congestion is more probable and the negative impact of packet loss on the service quality is of more significance. We argue that routing should not only be aware of, but also be adaptive to, network congestion. Hence, we propose a routing protocol (CRP) with such properties. Our ns-2 simulation results confirm that CRP improves the packet loss rate and end-to-end delay while enjoying significantly smaller protocol overhead and higher energy efficiency as compared to AODV and DSR     

面向煤矿工作面的定位无线传感器网络传输性能优化

针对煤矿工作面定位无线传感器网络(PWSN)端到端时间较长、丢包率较大等问题,提出采用保障贪婪调度(GGS)算法来优化网络传输性能。GGS算法结合了粒子群优化(PSO)算法和贪婪算法,使用PSO算法对信道中的报文种群进行有序化处理,实现对种群的保障;使用贪婪算法对网络传输过程中的具体服务请求形成一种多层次、反复迭代的处理机制,以优化报文种群质量;利用PSO变异算法对种群进行检查和更新,以确保得到的是最优解。仿真结果表明,与现有文化基因算法(MA)、差分进化人工蜂群(DE-ABC)算法相比,GGS算法可在控制丢包率的前提下缩短传输时间,提升网络整体性能。     

基于包对推测丢包率的简化算法

在进行网络测量时，有时只能获得端到端的数据，然而得到网络内部的信息对网络性能的认识是非常重要的，因此就需要从网络端到端测量得到的数据推测网络内部链路的数据。本算法是基于单播网络的端到端测量的，利用包对进行统计，运用最大似然估计和EM算法进行计算，从而推算网络内部节点的丢包率。我们在文中给出了算法的逻辑分析和仿真结果。