Providing deterministic quality-of-service guarantees on WDM optical networks

A major challenge in the design of future generation high-speed networks is the provision of guaranteed quality-of-service (QoS) for a wide variety of multimedia applications. In this paper we investigate the problem of providing QoS guarantees to real-time variable length messages (e.g., IP packets) in wavelength division multiplexing (WDM) optical networks. In particular, we propose a systematic mechanism comprised of admission control, traffic regulation, and message scheduling that provide guaranteed performance service for real-time application streams made up of variable-length messages. We formulate an analytical model based on the theory of max-plus algebra to evaluate the deterministic bounded message delay in a WDM network environment using our proposed QoS guarantee mechanism to determine the "schedulability conditions" of multimedia application streams, We also conduct a series of discrete-event and trace-driven simulations to verify the accuracy of the analytical model. The simulation results demonstrate that the analytic delay bound we obtained for our WDM optical network is valid and accurate.     

Fluid analysis of delay and packet discard performance for QoSsupport in wireless networks

Providing quality-of-service (QoS) guarantees over wireless links requires thorough understanding and quantification of the interactions among the traffic source, the wireless channel, and the underlying link-layer error control mechanisms. We account for such interactions in an analytical model that we use to investigate the delay distribution and the packet discard rate (PDR) over a wireless link. Our analysis accommodates the inherent autocorrelations in both the traffic source as well as the channel error characteristics. An on-off fluid process is used to model the arrival of packets at the transmitter. These packets are temporarily stored in a first-in-first-out (FIFO) buffer before being transmitted over a channel with a time-varying and autocorrelated service rate. Using fluid analysis, we first derive the distribution for the queueing delay at the transmitter. As part of this analysis, we solve a fundamental fluid problem, namely, the probability distribution for the workload generated by a two-state fluid source over a fixed time interval. We then use the delay analysis to derive the PDR at the receiver. A closed-form expression for the effective bandwidth subject to a delay constraint is provided as a function of the source, channel, and error scheme parameters. This expression enables fast assessment of the bandwidth requirement of real-time traffic over QoS-based wireless networks. Numerical results and simulations are used to verify the adequacy of the analysis and to study the interactions among various system parameters     

Stochastic performance analysis of non-feedforward networks

Many Internet applications are both delay and loss sensitive, and need network performance guarantees that include bandwidth, delay/delay jitter, and packet loss rate. It is very important to quantify and exploit the capabilities of guaranteed service provisioning of communication networks. In this paper, we study the queueing behaviors of non-feedforward networks (a non-feedforward network is a network in which at least one set of acyclic traffic routes forms a cycle; a feedforward network is a network in which any set of acyclic traffic routes does not form a cycle) with FIFO scheduling discipline and Regulated, Markov On-Off, and Fractional Brownian traffic sources. We develop a new methodology to analyze the probabilistic bounds on the delays experienced by traffic. By leveraging the large deviations and fixed-point techniques, we turn probability problems into deterministic optimization problems and translate a probabilistic delay bound into a fixed point of a non-linear real function. Our contribution in this paper is the derivation of a probabilistic bound on the delays experienced by traffic in non-feedforward networks, based on an assumption, i.e., the tail probability of the difference between the beginning time of a busy interval of a server and the earliest arriving time at the corresponding network ingress of the traffic arrivals that arrive at this server during this busy interval can be bounded by the maximum of the violation probabilities of the accumulative upper stream delay bound suffered by this server‘s traffic arrivals. Consequently, our new results not only consummate the theory of stochastic analysis of network performance, but also facilitate the design of protocols and algorithms for non-feedforward networks to provide performance guarantees to various applications with diverse performance requirements.     

Design and analysis of RT-Ring: a protocol for supporting real-time communications

Distributed applications with quality of service (QoS) requirements are more and more used in several areas (e.g., automated factory networks, embedded systems, conferencing systems). These applications produce a type of traffic with hard timing requirements, i.e., transmissions must be completed within specified deadlines. To handle these transmissions, the communication system must use real-time protocols to provide a communication service that is able to satisfy the QoS requirements of the distributed applications. In this paper, the authors propose a new real-time protocol, called RT-Ring, able to support transmissions of both real-time and generic traffic over a ring network. RT-Ring provides both network guarantees and high network resource utilization, while ensuring the compatibility with the emerging differentiated service architectures. Network guarantees are fully proved and high network utilization is highlighted by a comparative study with the FDDI protocol. This comparison shows that RT-Ring network capacities are greater than the corresponding FDDI capacities. In fact, by assuming the FDDI frames with a length equal to the RT-Ring slot size and by using the same traffic load the authors show that the capacities of FDDI are equal to the lower bound capacities of RT-Ring.     

A LP-based Admission Control Using Artificial Neural Networks for Integrated Services in Mobile Networks

In mobile networks the traffic fluctuation is unpredictable dueto mobility and varying resource requirements of multimedia applications.Henceit is essential to maintain the traffic within the network capacity to providethe service guarantees to running applications. Thispaper proposes an Admission Control (AC) scheme in a single mobile cellularenvironment supporting real-time and non-real-time application traffic. In thecase of a real-time and non-real-time multimedia applications, eachapplication has its own distinct range of acceptable Quality of Service (QoS)requirements(e.g., packet loss, delay, jitter, etc.). The network provides the service bymaintaining the application specified QoS range. We propose a LinearProgrammingResource Reduction (LP-RR) principle for admission control by maintainingQoSguarantees to existing applications and to increase the percentage ofadmissionto real-time and non-real-time applications. Artificial Neural Networks (ANNs)are used to solve linear programming problem, which facilitates an on-lineadmissioncontrol decision in the practical systems.The simulation results demonstrate that the proposed AC schemeperforms well in terms of admitted applications and maintains lower percentageof rejection to hand-off and new applications of different traffic classes.The suggested principle also shown that it is appropriate for the fairresourceallocation with improved resource utilization.     

An efficient ATM network switch scheduling

The problem of allocating network resources to application sessions backlogged at an individual switch has a great impact on the end-to-end delay and throughput guarantees offered by the network. There exists a class of algorithms based on weighted fair queueing (WFQ) for scheduling packets which are work-conserving and they guarantee fairness to the backlogged sessions. These algorithms also apply to ATM networks with a packet equal to a single cell or an ATM block (of fixed size). Bursts are groups of varying numbers of cells. We generalize WFQ to schedule bursts. Our motivation is to derive an adaptive algorithm which generalizes the (fixed size) packet level to a varying size packet level. The new algorithm enhances the performance of the switch service for many important applications. The proposed scheme maintains the work-conserving property, and also provides throughput and fairness guarantees. The worst-case delay bound is also given. We use simulation to study the performance characteristics of our algorithm. Our results demonstrate the efficiency of the new algorithm.     

A measurement-based admission control algorithm for integratedservice packet networks

Many designs for integrated services networks offer a bounded delay packet delivery service to support real-time applications. To provide a bounded delay service, networks must use admission control to regulate their load. Previous work on admission control mainly focused on algorithms that compute the worst case theoretical queueing delay to guarantee an absolute delay bound for all packets. In this paper, we describe a measurement-based admission control algorithm (ACA) for predictive service, which allows occasional delay violations. We have tested our algorithm through simulations on a wide variety of network topologies and driven with various source models, including some that exhibit long-range dependence, both in themselves and in their aggregation. Our simulation results suggest that measurement-based approach combined with the relaxed service commitment of predictive service enables us to achieve a high level of network utilization while still reliably meeting the delay bound     

移动Ad Hoc网络中一种分布式QoS保证的多址接入协议

基于随机竞争和冲突解决的思想,本文为多跳移动Ad Hoc网络提出了一种分布式服务质量(QoS)保证的多址接入(QMA)协议.该协议中,节点在发送业务分组前利用预报突发进行竞争接入,节点根据业务分组时延情况和最早失效优先原则确定预报突发的长度,所发预报突发能持续到最后的节点优先获得接入.同时,具有实时业务的节点可以按照其优先级在更早的竞争微时隙中开始发送预报突发,而有非实时业务的节点只能在前面竞争微时隙空闲的情况下,才能在后面的微时隙开始发送预报突发,因此发送实时业务的节点可以比发送非实时业务的节点更优先接入信道,从而在移动Ad Hoc网络中实现了对多媒体业务的QoS保证.最后利用OPNET仿真评估了QMA协议的多址性能,并与IEEE 802.11e协议的性能做了比较,结果表明QMA协议可以提供较高的吞吐量和较低的实时业务时延.     

Approximating Fluid Schedules in Crossbar Packet-Switches and Banyan Networks

We consider a problem motivated by the desire to provide flexible, rate-based, quality of service guarantees for packets sent over input queued switches and switch networks. Our focus is solving a type of online traffic scheduling problem, whose input at each time step is a set of desired traffic rates through the switch network. These traffic rates in general cannot be exactly achieved since they assume arbitrarily small fractions of packets can be transmitted at each time step. The goal of the traffic scheduling problem is to closely approximate the given sequence of traffic rates by a sequence of transmissions in which only whole packets are sent. We prove worst-case bounds on the additional buffer use, which we call backlog, that results from using such an approximation. We first consider the NtimesN, input queued, crossbar switch. Our main result is an online packet-scheduling algorithm using no speedup that guarantees backlog at most (N+1)² /4 packets at each input port and each output port. Upper bounds on worst-case backlog have been proved for the case of constant fluid schedules, such as the N²-2N+2 bound of Chang, Chen, and Huang (INFOCOM, 2000). Our main result for the crossbar switch is the first, to our knowledge, to bound backlog in terms of switch size N for arbitrary, time-varying fluid schedules, without using speedup. Our main result for Banyan networks is an exact characterization of the speedup required to maintain bounded backlog, in terms of polytopes derived from the network topology     

Towards Universal Power Efficient Scheduling in Gaussian Channels

In this paper, we propose a framework for designing power efficient schedulers for transmitting bursty traffic sources over Gaussian wireless channels that provides deterministic and statistical guarantees on absolute delays experienced by the source packets. The proposed schedulers compute the transmission rate and power using temporal water-filling techniques without any knowledge of the arrival traffic statistics. The schedulers reduce the average transmission power substantially (55% in some scenarios) for small increases in delay. The framework allows us to design schedulers that artfully tradeoff the performance with the complexity of computing the schedulers. We also introduce an iterative process to compute a lower bound on the transmit power of any scheduler that provides absolute delay guarantees. The utility of having accurate traffic predictors is demonstrated; specifically, we show that a perfect one step predictor achieves near optimal performances for small delay bounds. The proposed schedulers and iterative method of computing the lower bound are also shown to provide statistical guarantees on packet delays.     

一种基于紧急程度的自时钟开始时间公平排队分组调度算法

为了克服目前GPS (Generalized Processor Sharing)类调度算法中实时应用分组的排队时延较大且不稳定的局限性,该文提出一种新的分组排队调度算法,该调度算法在计算分组服务标签时添加了一个紧急程度函数,调整了到达分组间的竞争关系,从而可以按照实时性应用的要求来调整到达分组的转发先优级,由此显著降低了实时性应用分组的排队时延和抖动幅度。分析和仿真实验表明,与GPS类其它调度算法相比,该调度算法对于实时应用的分组能提供较低的、更稳定的排队时延保证,同时还继承了GPS类算法的公平性和排队时延有界等特性,而且对系统虚拟时间的跟踪计算更为简捷高效。     

Performance evaluation of the RQMA-CDMA scheme for multimedia traffic with QoS guarantees

The rapid growth of cellular mobile technology in recent years, coupled with the explosive growth of the Internet, has significantly increased the demand for wireless data services. Traffic on mobile devices is expected to be a mix of real-time traffic such as video, voice, and data, with users requiring diverse quality of service (QoS) guarantees for different types of traffic (video, voice and data). One of the primary challenges of providing QoS is how to prioritize and allocate network resources among contending applications. In order to achieve these goals, a scheduling scheme that can provide equitable and effective packet routing is required. This paper proposes a scheduling scheme called remote queuing multiple access-code division multiple access (RQMA-CDMA), whose purpose is to equitably assign bandwidth resources with QoS guarantees to different mobile devices. RQMA-CDMA is a rate scheduling scheme that can be used to assign bandwidth resources in conjunction with GPS (generalized processor sharing). Additionally, we analyze an admission control that is based on signal to interference plus noise ratio (SINR) for multimedia traffic. Finally, the performance of RQMA-CDMA is evaluated and compared to schemes based on CDMA-GPS according to dropped packets, delay, and throughput.     

Fair resource allocation with guaranteed statistical QoS for multimedia traffic in wideband CDMA cellular network

A dynamic fair resource allocation scheme is proposed to efficiently support real-time and non-real-time multimedia traffic with guaranteed statistical quality of service (QoS) in the uplink of a wideband code-division multiple access (CDMA) cellular network. The scheme uses the generalized processor sharing (GPS) fair service discipline to allocate uplink channel-resources, taking into account the characteristics of channel fading and intercell interference. In specific, the resource allocated to each traffic flow is proportional to an assigned weighting factor. For real-time traffic, the assigned weighting factor is a constant in order to guarantee the traffic statistical delay bound requirement; for non-real-time traffic, the assigned weighting factor can be adjusted dynamically according to fading, channel states and the traffic statistical fairness bound requirement. Compared with the conventional static-weight scheme, the proposed dynamic-weight scheme achieves capacity gain. A flexible trade-off between the GPS fairness and efficient resource utilization can also be achieved. Analysis and simulation results demonstrate that the proposed scheme enhances radio resource utilization and guarantees statistical QoS under different fairness bound requirements.     

A scheduling algorithm for QoS support in IEEE802.11 networks

This article presents a scheduling algorithm for the IEEE 802.11e hybrid coordination function under definition by the IEEE 802.11e task group. HCF can be used to provide IP quality of service guarantees in IEEE802.11e infrastructure WLANs. The enhanced distributed coordination function is mainly used for data transmission without QoS guarantees, but can also be used to decrease the transmission delay of QoS-sensitive traffic. Scheduling of queued packets follows a delay-earliest-due-date algorithm. The proposed algorithm is compatible with the link adaptation mechanisms implemented in commercial WLANs, as it limits the amount of time during which the stations control the wireless medium. The performance of the algorithm is evaluated through computer simulation and compared with the reference scheduler presented by the IEEE 802.11e task group.     

Credit-based processor sharing for decoupled delay and bandwidthallocation

We propose a credit-based processor sharing (CPS) approach for decoupled allocation of delay and bandwidth. For bandwidth guarantees, a CPS system serves traffic flows in proportion to their service weights. For delay guarantees, on the other hand, a CPS system allows real-time flows to temporarily borrow bandwidth from nonreal-time flows using service credits. This borrowing mechanism boosts the delay performance of real-time flows without increasing their long-term bandwidth requirements. By systematically regulating service credits of real-time flows, nonreal-time flows are protected for their aggregate long-term bandwidth share     

Providing deterministic packet delays and packet losses in multimedia wireless networks

‘Anytime, anywhere’ communication, information access and processing are much cherished in modern societies because of their ability to bring flexibility, freedom and increased efficiency to individuals and organizations. Wireless communications, by providing ubiquitous and tetherless network connectivity to mobile users, are therefore bound to play a major role in the advancement of our society. Although initial proposals and implementations of wireless communications are generally focused on near‐term voice and electronic messaging applications, it is recognized that future wireless communications will have to evolve towards supporting a wider range of applications, including voice, video, data, images and connections to wired networks. This implies that future wireless networks must provide quality‐of‐service (QoS) guarantees to various multimedia applications in a wireless environment. Typical traffic in multimedia applications can be classified as either Constant‐Bit‐Rate (CBR) traffic or Variable‐Bit‐Rate (VBR) traffic. In particular, scheduling the transmission of VBR multimedia traffic streams in a wireless environment is very challenging and is still an open problem. In general, there are two ways to guarantee the QoS of VBR multimedia streams, either deterministically or statistically. In particular, most connection admission control (CAC) algorithms and medium access control (MAC) protocols that have been proposed for multimedia wireless networks only provide statistical, or soft, QoS guarantees. In this paper, we consider deterministic QoS guarantees in multimedia wireless networks. We propose a method for constructing a packet‐dropping mechanism that is based on a mathematical framework that determines how many packets can be dropped while the required QoS can still be preserved. This is achieved by employing: (1) An accurate traffic characterization of the VBR multimedia traffic streams; (2) A traffic regulator that can provide bounded packet loss and (3) A traffic scheduler that can provide bounded packet delay. The combination of traffic characterization, regulation and scheduling can provide bounded loss and delay deterministically. This is a distinction from traditional deterministic QoS schemes in which a 0% packet loss are always assumed with deterministically bounding the delay. We performed a set of performance evaluation experiments. The results will demonstrate that our proposed QoS guarantee schemes can significantly support more connections than a system, which does not allow any loss, at the same required QoS. Moreover, from our evaluation experiments, we found that the proposed algorithms are able to out‐perform scheduling algorithms adopted in state‐of‐the‐art wireless MAC protocols, for example Mobile Access Scheme Based on Contention and Reservation for ATM (MASCARA) when the worst‐case traffic is being considered. Copyright © 2002 John Wiley & Sons, Ltd.     

Exact admission control for networks with a bounded delay service

To support the requirements for the transmission of continuous media, such as audio and video, multiservice packet-switching networks must provide service guarantees to connections, including guarantees on throughput, network delays, and network delay variations. For the most demanding applications, the network must offer a service which provides deterministically bounded delay guarantees, referred to as “bounded delay service.” The admission control functions in a network with a bounded delay service require `schedulability conditions' that detect violations of delay guarantees in a network switch. Exact schedulability conditions are presented for three packet scheduling methods: earliest-deadline-first (EDF), static-priority (SP), and a novel scheduling method, referred to as rotating-priority-queues (RPQ). By characterizing the worst-case traffic with general subadditive functions, the presented schedulability conditions can be applied to a large class of traffic models. Examples, which include actual MPEG video traces, are presented to demonstrate the trade-offs involved in selecting a packet scheduling method for a bounded delay service     

Performance Evaluation of the IEEE 802.16 MAC for QoS Support

The IEEE 802.16 is a standard for broadband wireless communication in metropolitan area networks (MAN). To meet the QoS requirements of multimedia applications, the IEEE 802.16 standard provides four different scheduling services: unsolicited grant service (UGS), real-time polling service (rtPS), non-real-time polling service (nrtPS), and Best Effort (BE). The paper is aimed at verifying, via simulation, the effectiveness of rtPS, nrtPS, and BE (but UGS) in managing traffic generated by data and multimedia sources. Performance is assessed for an IEEE 802.16 wireless system working in point-to-multipoint (PMP) mode, with frequency division duplex (FDD), and with full-duplex subscriber stations (SSs). Our results show that the performance of the system, in terms of throughput and delay, depends on several factors. These include the frame duration, the mechanisms for requesting uplink bandwidth, and the offered load partitioning, i.e., the way traffic is distributed among SSs, connections within each SS, and traffic sources within each connection. The results also highlight that the rtPS scheduling service is a very robust scheduling service for meeting the delay requirements of multimedia applications     

基于测量的软实时业务接纳控制方法

软实时业务对业务传送的实时性并不要求绝对的延时上界保证,因此通常不需要基于最坏的情况进行接纳控制。文章针对软实时业务的业务特性提出了一种新型的基于测量的接纳控制方法,该方法首先通过对网络流量进行实时测量来估计流量的统计特性,然后结合软实时业务的实时性指标进行接纳控制。仿真结果表明,提出的接纳控制方法不仅有效地提高了带宽利用率,而且在带宽利用率与业务实时性指标之间建立了显式关联,因此可以更加灵活有效地为软实时业务提供服务质量保证。