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1.
This paper presents the use of adaptive bandwidth control (ABC) for a quantitative packet loss rate guarantee to aggregate traffic in packet switched networks. ABC starts with some initial amount of bandwidth allocated to a queue and adjusts it over time based on online measurements of system states to ensure that the allocated bandwidth is just enough to attain the specified loss requirement. Consequently, no a priori detailed traffic information is required, making ABC more suitable for efficient aggregate quality of service (QoS) provisioning. We propose an ABC algorithm called augmented Fuzzy (A-Fuzzy) control, whereby fuzzy logic control is used to keep an average queue length at an appropriate target value, and the measured packet loss rate is used to augment the standard control to achieve better performance. An extensive simulation study based on both theoretical traffic models and real traffic traces under a wide range of system configurations demonstrates that the A-Fuzzy control itself is highly robust, yields high bandwidth utilization, and is indeed a viable alternative and improvement to static bandwidth allocation (SBA) and existing adaptive bandwidth allocation schemes. Additionally, we develop a simple and efficient measurement-based admission control procedure which limits the amount of input traffic in order to maintain the performance of the A-Fuzzy control at an acceptable level.  相似文献   

2.
《Applied Soft Computing》2008,8(1):274-284
3G Wireless systems are to support multiple classes of traffic with widely different characteristics and quality of service (QoS) requirements. A major challenge in this system is to guarantee the promised QoS for the admitted users, while maximizing the resource allocation through dynamic resource sharing. In the case of multimedia call, each of the services has its own distinct QoS requirements concerning probability of blocking (PB), service access delay (SAD), and access delay variation (ADV). The 3G wireless system attempts to deliver the required QoS by allocating appropriate resources (e.g. bandwidth, buffers), and bandwidth allocation is a key in achieving this. Dynamic bandwidth allocation policies reported so far in the literature deal with audio source only. They do not consider QoS requirements. In this work, a fuzzy logic (FL)-based dynamic bandwidth allocation algorithm for multimedia services with multiple QoS (PB, SAD, ADV, and the arrival rate) requirements are presented and analyzed. Here, each service can declare a range of acceptable QoS levels (e.g. high, medium, and low). As QoS demand varies, the proposed algorithm allocates the best possible bandwidth to each of the services. This maximizes the utilization and fair distribution of resources. The proposed allocation method is validated in a variety of scenarios. The results show that the required QoS can be obtained by appropriately tuning the fuzzy logic controller (FLC).  相似文献   

3.
《Computer Networks》2007,51(4):1060-1071
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.  相似文献   

4.
With fast proliferation of QoS-enabled wireless packet networks, need for effective QoS control is increasing. In this paper, we focus on QoS provisioning in Mobile WiMAX access service network (ASN). We investigate a dynamic bandwidth provisioning method that can help to increase resource utilization. Our approach consists of two stages: traffic forecasting, followed by bandwidth provisioning. For the first stage, we use auto-regressive integrated moving average (ARIMA) model to forecast traffic based on online measurement. For the second stage, we use a bandwidth provisioning scheme that allocates bandwidths depending on the traffic forecasting. We modeled our problem as a Fractional Knapsack Problem for which we used a greedy algorithm in order to find an approximate solution. Through simulation studies with real-world data sets, we found that our approach could increase the bandwidth for the real-time traffic class and guarantee adequate service quality for the nonreal-time traffic class as well, while maximizing resource utilization.  相似文献   

5.
This paper proposes a new mechanism called the Priority Token Bank for admission control, scheduling and policing in integrated-services networks. In such networks, both arrival processes and performance objectives can vary greatly from one packet stream to another. There are two principal components to the Priority Token Bank: accepting or rejecting requests to admit entire packet streams, where acceptance means guaranteeing that the packet stream's performance objectives will be met, and scheduling the transmission of packets such that performance objectives are met, even under heavy loads. To the extent possible, the performance of traffic is also optimized beyond the requirements. The performance achieved with the Priority Token Bank is compared to that of other typical algorithms. It is shown that, when operating under the constraint that the performance objectives of applications such as packet voice, video and bulk data transfer must be met in an ATM network, the mean delay experienced by other traffic is much better with the Priority Token Bank. Furthermore, the admission control algorithm can guarantee requirements will be met, and admit more traffic than the common alternatives.  相似文献   

6.
《Computer Networks》2007,51(4):1183-1204
The Differentiated Service (DiffServ) network model has been defined as a scalable framework for providing Quality of Service to applications. In this model, traffic is classified into several service classes with different priorities inside queues of IP routers. The premium service class has the highest priority. Due to the high priority of premium traffic, the global network behaviour against this service class, including routing and scheduling of premium packets, may impose significant influences on traffic of other classes. These negative influences, which could degrade the performance of low-priority classes with respect to some important metrics such as the packet loss probability and the packet delay, are often called the inter-class effects. To reduce the inter-class effects, the premium-class routing algorithm must be carefully selected such that (1) it works correctly (i.e., without loop) under the hop-by-hop routing paradigm; and (2) the congestion resulted from the traffic of premium class over the network becomes minimum. In this paper, we first introduce a novel routing framework, named compatible routing, that guarantees loop-freedom in the context of hop-by-hop routing model. Then, upon this framework, we propose two multipath architectures for load balancing of high-priority traffic on DiffServ networks. Our extensive simulations clearly demonstrate that the proposed methods distribute the premium bandwidth requirements more efficiently over the whole network and perform better than the existing algorithms, especially in the case of complex and highly loaded networks.  相似文献   

7.
An efficient bandwidth allocation scheme in wireless networks should not only guarantee successful data transmission without collisions but also enhance the channel spatial reuse to maximize the system throughput. The design of high-performance wireless Local Area Network (LAN) technologies making use of TDMA/FDD MAC (Time Division Multiple Access/Frequency Division Duplex - Medium Access Control) is a very active area of research and development. Several protocols have been proposed in the literature as TDMA-based bandwidth allocation schemes. However, they do not have a convenient generic parameters or suitable frame repartition for dynamic adjustment. In this work, we undertake the design and performance evaluation of a QoS (Quality of Service)-aware scheme built on top of the underlying signaling and bandwidth allocation mechanisms provided by most wireless LANs standards. The main contribution of this study is the new guarantee-based dynamic adjustment algorithm used in MAC level to provide the required QoS for all traffic types in wireless medium especially Wireless ATM (Asynchronous Transfer Mode). Performance evaluation of this approach consists of improving the bandwidth utilization, supporting different QoS requirements and reducing call reject probability and packet latency.  相似文献   

8.
QoS provisioning is an important issue in the deployment of broadband wireless access networks with real-time and non-real-time traffic integration. An opportunistic MAC (OMAC) combines cross-layer design features with opportunistic scheduling scheme to achieve high system utilization while providing QoS support to various applications. A single scheduling algorithm cannot guarantee all the QoS requirements of traffics without the support of a suitable CAC and vice versa. In this paper, we propose a cross-layer MAC scheduling framework and a corresponding opportunistic scheduling algorithm in tandem with the CAC algorithm to support QoS in WiMAX point-to-multipoint (PMP) networks. Extensive experimental simulations have been carried out to evaluate the performance of our proposal. The simulation results show that our proposed solution can improve the performance of WiMAX networks in terms of packet delay, packet loss rate and throughput. The proposed CAC scheme can guarantee the admitted connections to meet their QoS requirements.  相似文献   

9.
Although MMORPGs are becoming increasingly popular as well as a highly profitable Internet business, there is still a fundamental design question: Which transport protocol should be used—TCP, UDP, or some other protocol? In this paper, we first evaluate whether TCP is suitable for MMORPGs, and then propose some novel transport strategies for this genre of games. Our analysis of a trace collected from a TCP-based MMORPG called ShenZhou Online indicates that TCP is unwieldy and inappropriate for MMORPGs. We find that the degraded network performance problems are due to the following characteristics of MMORPG traffic: 1) tiny packets, 2) a low packet rate, 3) application-limited traffic generation, and 4) bi-directional traffic. Since not all game packets require reliable transmission or in-order delivery, transmitting all packets with a strict delivery guarantee causes high delays and delay jitters. Therefore, our proposed transport strategies assign game packets with appropriate levels of transmission guarantee depending on the requirements of the packets’ contents. To compare the performance of our approach with that of existing transport protocols, we conduct network simulations with a real-life game trace from Angel’s Love. The results demonstrate that our strategies significantly reduce the end-to-end delay and delay jitter of packet delivery. Finally, we show that our strategies effectively raise satisfaction levels of the game players.  相似文献   

10.
区分服务和流量整形是保证实时应用和关键应用得到相应QoS的重要技术。提出基于二级令牌分配机制的流量整形结构,逐级设计令牌分配的具体算法。该算法使区分服务网络中优先级较高的数据流获得更多的出口带宽资源,同时根据输出需求动态调整同优先级数据流的带宽资源,兼顾优先和公平,提高企业网络出口有限带宽的使用效益。  相似文献   

11.
《Computer Networks》2003,41(2):247-267
In this paper we present a novel design technique for packet switched networks. The design is based on the construction of multiple virtual rings, which enjoy the one-bridge property: the path between any two nodes is either confined to a single ring or traverses exactly two rings (passing through a single bridge node). Our best designs are constructed by using finite generalized quadrangles of combinatorial design theory. We present novel routing and flow control protocols that capitalize on the one-bridge property of the multi-ring network. Our protocols ensure that (i) no loss due to congestion occurs inside a network, under arbitrary traffic patterns; (ii) all the packets reach their destinations within bounded time with low jitter; and (iii) the bandwidth is allocated fairly and no host is starved. We provide both a theoretical analysis and an extensive simulation-based performance evaluation of our protocols.  相似文献   

12.
分析了IEEE 802.11e协议HCCA信道接入机制下的简单带宽调度算法对多媒体业务的QoS支持情况,指出其不足并在其基础上进行了改进,提出了一种基于业务等级的带宽调度算法E-HCCA(Enhanced HCCA)。E-HCCA对不同优先等级业务的数据在带宽分配上采用不同的策略,在优先保证各个节点CBR业务的基础上,根据节点的VBR流量动态平均分配剩余带宽。相比较简单调度算法,E-HCCA算法更好地支持了多用户下的语音业务流和视频业务流,降低了分组时延,增加了系统吞吐率。  相似文献   

13.
Network-on-Chip (NoC) architecture has been widely used in many multi-core system designs. To improve the communication efficiency and the bandwidth utilization of NoC for various applications, we firstly propose a table-based algorithm for identifying the dominant flows at runtime. Then a two-layer NoC architecture with an application-driven bandwidth allocation scheme is presented, which is capable of identifying heavy-load dataflows and dynamically reconfiguring point-to-point (P2P) connections to optimize the heavy-load traffic. Experimental results reveal that our design (8 × 8 mesh NoC) achieves 28.5% performance improvement and 25.9% power consumption saving compared to the baseline NoC.  相似文献   

14.
在现代工业无线网络中,IEEE 802.15.4标准以其独特的低功耗、低成本特点被广泛应用。IEEE 802.15.4可以提供最低0.006%的占空比,最大限度降低功耗,同时提供的保障时隙GTS机制为节点提供了实时服务保障。然而,在为大规模节点提供保障时,IEEE 802.15.4提供的GTS机制缺乏灵活性,只能为有限节点提供实时保障服务。本文针对这一问题提出一种多节点共享保障时隙分配策略,允许多个数据流在满足延迟需求前提下,共享同一个GTS减少带宽浪费。分析表明,多节点共享的保障时隙分配策略与普通分配方法相比,可有效提高带宽利用率。  相似文献   

15.
概述了LOBS-HC(labeled optical burst switch with home circuits)技术及其网络的结构,及其能提供带宽保证并提高波长带宽利用率的特性。基于Visual C++设计并实现了LOBS-HC网络仿真平台,包括网络业务流的产生和汇聚、突发包的组装、HC(home circuit)逻辑链路的建立和撤销过程的模拟。通过吞吐率、网络延时、波长利用率等性能参数对LOBS-HC网络进行评测。仿真结果表明,LOBS-HC能在提供与OCS相同带宽保证的同时,获得更好的网络带宽利用率。  相似文献   

16.
《Computer Networks》2008,52(6):1291-1307
The possibility of adding multi protocol label switching (MPLS) support to transport networks is considered an important opportunity by telecom carriers that want to add packet services and applications to their networks. However, the question arises whether it is suitable to have MPLS nodes just at the edge of the network to collect packet traffic from users, or to introduce also MPLS facilities on a subset of the core nodes in order to exploit packet switching flexibility and multiplexing, thus inducing a better bandwidth allocation. In this paper, we propose a mathematical programming model for the design of two-layer networks where MPLS is considered on top of transport networks (SDH or WDM depending on required link speed). Our models take into account the tradeoff between the cost of adding MPLS support in the core nodes and the savings in the link bandwidth allocation due to the statistical multiplexing and the traffic grooming effects induced by MPLS nodes. The traffic matrix specifies for each point-to-point request a pair of values: a mean traffic value and an additional one. Using this traffic model, the effect of statistical multiplexing on a link allows to allocate a capacity equal to the sum of all the mean values of the traffic demands routed on the link and only the highest additional one. We propose a path-based Mixed Integer Programming (MIP) model for the problem of optimizing the number and location of MPLS nodes in the network and the link capacities. We apply Lagrangian relaxation to this model and use the subgradient method to obtain a lower bound of the network cost. As the number of path variables used to model the routing grows exponentially with the graph size, we use an initially limited number of variables and a column generation approach. We also introduce a heuristic approach to get a good feasible solution. Computational results are reported for small size and real-world instances.  相似文献   

17.
This work presents a study of RTP multiplexing schemes, which are compared with the normal use of RTP, in terms of experienced quality. Bandwidth saving, latency and packet loss for different options are studied, and some tests of Voice over IP (VoIP) traffic are carried out in order to compare the quality obtained using different implementations of the router buffer. Voice quality is calculated using ITU R-factor, which is a widely accepted quality estimator. The tests show the bandwidth savings of multiplexing, and also the importance of packet size for certain buffers, as latency and packet loss may be affected. The customer’s experience improvement is measured, showing that the use of multiplexing can be interesting in some scenarios, like an enterprise with different offices connected via the Internet. The system is also tested using different numbers of samples per packet, and the distribution of the flows into different tunnels is found to be an important factor in order to achieve an optimal perceived quality for each kind of buffer. Grouping all the flows into a single tunnel will not always be the best solution, as the increase of the number of flows does not improve bandwidth efficiency indefinitely. If the buffer penalizes big packets, it will be better to group the flows into a number of tunnels. The router processing capacity has to be taken into account too, as the limit of packets per second it can manage must not be exceeded. The obtained results show that multiplexing is a good way to improve customer’s experience of VoIP in scenarios where many RTP flows share the same path.  相似文献   

18.
A load-balanced two-stage switch is scalable and can provide close to 100% throughput. Its major problem is that packets can be mis-sequenced when they arrive at outputs. In a recent work, the packet mis-sequencing problem is elegantly solved by a feedback-based switch architecture. In this paper, we extend the feedback-based switch from two-stage to three-stage for further cutting down average packet delay while still ensuring in-order packet delivery and close to 100% throughput. The basic idea is to use the third stage switch to map heavy flows to experience less middle-stage delays. To identity heavy flows, an adaptive traffic estimation algorithm is proposed. To ensure max-min fairness in bandwidth allocation under any inadmissible traffic pattern, an efficient fair scheduler is devised.  相似文献   

19.
链路带宽测量方法改进   总被引:5,自引:1,他引:5  
链路带宽是网络性能分析,容量优化规划的基本指标,链路带宽测量常用的方法是VPS(variable packet size),但VPS具有误差累计和背景流量影响的缺陷,对VPS方法进行改进,提出和实现一个任意链路带宽测量方法PTVS(packet train with variable size),消除逐跳测量造成的误差累计和背景流量影响,测量实验表明,PTVS具有精确,高效,迅速的特点,PTVS可测量的其他性能指标还包括RTT,单向延迟,丢包率,端到端瓶颈带宽以及链路利用率。  相似文献   

20.
针对VoIP(Voice over IP)业务在无线Mesh网上进行传输时存在服务质量(QoS)需求难以保证、带宽利用率低的问题,介绍了VoIP的QoS影响因素,分析了端到端时延、时延抖动和丢包率等几个重要参数,并对VoIP在无线Mesh网中的传输性能进行了论述。提出了基于无线Mesh网络的QoS保证机制,可以为端到端的数据传输公平的分配带宽,并能在保证QoS下实现大规模的实时任务的多跳转发。仿真试验表明能有效降低端到端时延,有着更好的QoS性能。  相似文献   

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