Enhancing QoS provisioning and granularity in next generation internet

Next Generation IP technology has the potential to prevail, both in the access and in the core networks, as we are moving towards a multi-service, multimedia and high-speed networking environment. Many new applications, including the multimedia applications, have been developed and deployed, and demand Quality of Service (QoS) support from the Internet, in addition to the current best effort service. Therefore, QoS provisioning techniques in the Internet to guarantee some specific QoS parameters are more a requirement than a desire. Due to the large amount of data flows and bandwidth demand, as well as the various QoS requirements, scalability and fine granularity in QoS provisioning are required. In this dissertation, the end-to-end QoS provisioning mechanisms are mainly studied, in order to provide scalable services with fine granularity to the users, so that both users and network service providers can achieve more benefits from the QoS provisioned in the network. To provide the end-to-end QoS guarantee, single-node QoS provisioning schemes have to be deployed at each router, and therefore, in this dissertation, such schemes are studied prior to the study of the end-to-end QoS provisioning mechanisms. Specifically, the effective sharing of the output bandwidth among the large amount of data flows is studied, so that fairness in the bandwidth allocation among the flows can be achieved in a scalable fashion. A dual-rate grouping architecture is proposed in this dissertation, in which the granularity in rate allocation can be enhanced, while the scalability of the one-rate grouping architecture is still maintained. It is demonstrated that the dual-rate grouping architecture approximates the ideal per-flow based PFQ architecture better than the one-rate grouping architecture, and provides better immunity capability. On the end-to-end QoS provisioning, a new Endpoint Admission Control scheme for Diffserv networks, referred to as Explicit Endpoint Admission Control (EEAC), is proposed, in which the admission control decision is made by the end hosts based on the end-to-end performance of the network. A novel concept, namely the service vector, is introduced, by which an end host can choose different services at different routers along its data path. Thus, the proposed service provisioning paradigm decouples the end-to-end QoS provisioning from the service provisioning at each router, and the end-to-end QoS granularity in the Diffserv networks can be enhanced, while the implementation complexity of the Diffserv model is maintained. Furthermore, several aspects of the implementation of the EEAC and service vector paradigm, referred to as EEAC-SV, in the Diffserv architecture are also investigated. The performance analysis and simulation results demonstrate that the proposed EEAC-SV scheme, not only increases the benefit to the service users, but also enhances the benefit to the network service provider in terms of network resource utilization. The study also indicates that the proposed EEAC-SV scheme can provide a compatible and friendly networking environment to the conventional TCP flows, and the scheme can be deployed in the current Internet in an incremental and gradual fashion

Performance Management in ATM Networks

ATM is representative of the connection-oriented resource provisioning classof protocols. The ATM network is expected to provide end-to-end QoS guaranteesto connections in the form of bounds on delays, errors and/or losses. Performancemanagement involves measurement of QoS parameters, and application of controlmeasures (if required) to improve the QoS provided to connections, or to improvethe resource utilization at switches. QoS provisioning is very important for realtimeconnections in which losses are irrecoverable and delays cause interruptionsin service. QoS of connections on a node is a direct function of the queueing andscheduling on the switch. Most scheduling architectures provide static allocationof resources (scheduling priority, maximum buffer) at connection setup time. Endto-end bounds are obtainable for some schedulers, however these are precluded forheterogeneously composed networks. The resource allocation does not adapt to theQoS provided on connections in real time. In addition, mechanisms to measurethe QoS of a connection in real-time are scarce.In this thesis, a novel framework for performance management is proposed. Itprovides QoS guarantees to real time connections. It comprises of in-service QoSmonitoring mechanisms, a hierarchical scheduling algorithm based on dynamicpriorities that are adaptive to measurements, and methods to tune the schedulers atindividual nodes based on the end-to-end measurements. Also, a novel scheduler isintroduced for scheduling maximum delay sensitive traffic. The worst case analysisfor the leaky bucket constrained traffic arrivals is presented for this scheduler. Thisscheduler is also implemented on a switch and its practical aspects are analyzed.In order to understand the implementability of complex scheduling mechanisms,a comprehensive survey of the state-of-the-art technology used in the industry isperformed. The thesis also introduces a method of measuring the one-way delayand jitter in a connection using in-service monitoring by special cells

ESTADO DEL ARTE DE SCHEDULING ALGORITHMS (ALGORITMOS DE PROGRAMACIÓN DE ENVÍO)

El presente artículo presenta un estado del arte de los llamados algoritmos de programación de envío, o en inglés “scheduling algorithms” los cuales determinan el método en que un dispositivo de red envía paquetes al medio de transmisión. Estos algoritmos son importantes para la consecución de calidad de servicio en redes convergentes y la correcta comprensión de ellos es determinante para entender los fenómenos de tráfico y cumplir los requerimientos dados en los diferentes tipos de redes (Ethernet, TDM, WiMAX, Wireless, etc)

Network Calculus

Network Calculus is a collection of results based on Min-Plus algebra, which applies to deterministic queuing systems found in communication networks. It can be used for example to understand - the computations for delays used in the IETF guaranteed service - why re-shaping delays can be ignored in shapers or spacer-controllers - a common model for schedulers - deterministic effective bandwidth and much more

Dynamic bandwidth scheduling and burst construction algorithm for downlink in (4G) mobile WiMAX networks

Advanced wireless systems, also called fourth generation (4G) wireless systems, such as Mobile Worldwide interoperability for Microwave Access (WiMAX), are developed to provide broadband wireless access in true sense. Therefore, it becomes mandatory for such kind of systems to provide Quality of Service (QoS) support for wide range of applications. In such types of systems, wireless base stations are responsible for distributing proper amount of bandwidth among different mobile users, thus satisfying a user’s QoS requirements. The task of distributing proper amount of bandwidth rests upon a scheduling algorithm, typically executed at the base station. 2G and 3G wireless systems are able to provide only voice, low data rate, and delay insensitive services, such as Web browsing. This is due to the lack of development in digital modulation and multiple access schemes, which are two major aspects of physical layer of these systems. Digital modulation is used to combat with location-dependent channel errors which get introduced in the data transmitted by base station on a wireless channel to a mobile station. Hence, different locations of every mobile station in a cell coverage area require different modulation and coding schemes for error-free transmission. Link adaptation is a technique that makes the use of variable modulation and coding schemes possible, according to varying location of mobile stations. This technique is used by 4G systems to achieve error free transmissions. 2G and 3G systems are not capable of achieving error-free transmissions in many cases due to significantly fewer or no choice of modulation and coding schemes for different locations of mobile stations. In such cases, most of the time, wireless channel is either error-prone or error-free for mobile station. Scheduling algorithms developed for 2G and 3G systems focussed on providing long term average rate requirements of users, which are satisfied at the expense of zero transmission for mobile users experiencing bad or error prone channel. This approach was adopted to achieve efficient use of wireless channel capacity. This was the best approach adopted by majority of scheduling algorithms because delay sensitive applications were not supported in such systems and hence bounded delay was not a matter of concern. Hence, the majority of the algorithms focussed on providing long term average rate requirements while maximizing cell throughput. This helped in making efficient use of wireless channel capacity at the expense of zero transmission for mobile users experiencing bad channel and compromising delay performance. These approaches, however, will not be suitable for 4G systems as such systems support wide range of applications ranging from delay-insensitive to highly delay-sensitive. Hence in this thesis, a dynamic bandwidth scheduling algorithm called Leaky Bucket Token Bank (LBTB) is proposed. This algorithm exploits some advanced features of 4G systems, like link adaptation and multiple access scheme, to achieve long term average rate requirements for delay-insensitive applications and bounded delay for delay-sensitive applications. Advanced features of 4G systems also bring more challenges. One such challenge is Orthogonal Frequency Division Multiple Access (OFDMA), a multiple access scheme deployed in 4G systems. In OFDMA, scheduled data for different mobile stations is packed into bursts and mapped to a two dimensional structure of time and frequency, called OFDMA frame. It has been observed that the way bursts are mapped to OFDMA frame affects the wakeup time of mobile stations receiving data and therefore causes power consumption. Wakeup time is the time duration in OFDMA frame for which the mobile station becomes active. Since OFDMA frame is a limited and precious radio resource, the efficient use of such radio resource is necessary. Efficient use requires that the wastage of such radio resource be minimized. Hence in this thesis, a burst construction algorithm called Burst Construction for Fairness in Power (BCFP) is also proposed. The algorithm attempts to achieve fairness in power consumption of different mobile stations by affecting their wakeup time. It also attempts to minimize wastage of radio resources. For comparing the performance of joint proposed algorithms (LBTB+BCFP), the proposed burst construction algorithm (BCFP) is joined to the two other existing scheduling algorithms namely: Token Bank Fair Queuing (TBFQ) and Adaptive Token Bank Fair Queuing (ATBFQ). TBFQ is an algorithm developed for 3G wireless networks whereas, ATBFQ is an extension to the TBFQ and is developed for 4G wireless networks. Therefore, the performance of the proposed algorithms jointly together (LBTB+BCFP) is compared with the joint TBFQ and proposed burst construction algorithm (TBFQ+BCFP), as well as joint ATBFQ and proposed burst construction algorithm (ATBFQ+BCFP). We compare the performance in terms of average queuing delay, average cell throughput, packet loss, fairness among different mobile users, fairness in average wakeup times (average power consumption), and fraction of radio resources wasted. The performance of proposed burst construction algorithm (BCFP) is also compared with Round Robin algorithm in terms of fairness in average power consumption as well as fraction of radio resources wasted, for varying number of users

Application of learning algorithms to traffic management in integrated services networks.

SIGLEAvailable from British Library Document Supply Centre-DSC:DXN027131 / BLDSC - British Library Document Supply CentreGBUnited Kingdo

Design and implementation of multiple address parallel transmission architecture for storage area network

Master'sMASTER OF ENGINEERIN

Bits of Internet traffic control

In this work, we consider four problems in the context of Internet traffic control. The first problem is to understand when and why a sender that implements an equation-based rate control would be TCP-friendly, or not—a sender is said to be TCP-friendly if, under the same operating conditions, its long-term average send rate does not exceed that of a TCP sender. It is an established axiom that some senders in the Internet would need to be TCP-friendly. An equation-based rate control sender plugs-in some on-line estimates of the loss-event rate and an expected round-trip time in a TCP throughput formula, and then at some points in time sets its send rate to such computed values. Conventional wisdom held that if a sender adjusts its send rate as just described, then it would be TCP-friendly. We show exact analysis that tells us when we should expect an equation-based rate control to be TCP-friendly, and in some cases excessively so. We show experimental evidence and identify the causes that, in a realistic scenario, make an equation-based rate control grossly non-TCP-friendly. Our second problem is to understand the throughput achieved by another family of send rate controls—we termed these "increase-decrease controls," with additive-increase/multiplicative-decrease as a special case. One issue that we consider is the allocation of long-term average send rates among senders that adjust their send rates by an additive-increase/multiplicative-decrease control, in a network of links with arbitrary fixed routes, and arbitrary round-trip times. We show what the resulting send rate allocation is. This result advances the state-of-the-art in understanding the fairness of the rate allocation in presence of arbitrary round-trip times. We also consider the design of an increase-decrease control to achieve a given target loss-throughput function. We show that if we design some increase-decrease controls under a commonly used reference loss process—a sequence of constant inter-loss event times—then we know that these controls would overshoot their target loss-throughput function, for some more general loss processes. A reason to study the design problem is to construct an increase-decrease control that would be friendly to some other control, TCP, for instance. The third problem that we consider is how to obtain probabilistic bounds on performance for nodes that conform to the per-hop-behavior of Expedited Forwarding, a service of differentiated services Internet. Under the assumption that the arrival process to a node consists of flows that are individually regulated (as it is commonplace with Expedited Forwarding) and the flows are stochastically independent, we obtained probabilistic bounds on backlog, delay, and loss. We apply our single-node performance bounds to a network of nodes. Having good probabilistic bounds on the performance of nodes that conform to the per-hop-behavior of Expedited Forwarding, would enable a dimensioning of those networks more effectively, than by using some deterministic worst-case performance bounds. Our last problem is on the latency of an input-queued switch that implements a decomposition-based scheduler. With decomposition-based schedulers, we are given a rate demand matrix to be offered by a switch in the long-term between the switch input/output port pairs. A given rate demand matrix is, by some standard techniques, decomposed into a set of permutation matrices that define the connectivity of the input/output port pairs. The problem is how to construct a schedule of the permutation matrices such that the schedule offers a small latency for each input/output port pair of the switch. We obtain bounds on the latency for some schedulers that are in many situations smaller than a best-known bound. It is important to be able to design switches with bounds on their latencies in order to provide guarantees on delay-jitter

Video traffic : characterization, modelling and transmission

EThOS - Electronic Theses Online ServiceGBUnited Kingdo