Applications with timing requirements, such as multimedia and live multi-user interaction, are becoming more prevalent in wide area networks. The desire to provide more predictable performance for such applications in packet switched wide area networks is evident in the channel management provided by Asynchronous Transfer Mode (ATM) networks and in the extensions to the Internet protocols proposed by the Internet Engineering Task Force (IETF) working groups on integrated and differentiated service. The ability to provide guarantees on the performance of traffic flows, such as packet delay and loss characteristics, relies on an accurate model of the traffic arrival and service at each node in the network. This thesis surveys the work in bounding packet delay based on various proposed queuing disciplines and proposes a method for more accurately defining the traffic arrival and worst case backlog experienced by packets. The methods are applied to the first in first out (FIFO) queuing discipline to define equations for determining the worst case backlog and queuing delay in multihop networks. Simulation results show a significant improvement in the accuracy of the delay bounds over existing bounds published in the literature. An improvement of two orders of magnitude can be realised for a ten hop path and the improvement increases exponentially with the length of the path for variable rate network traffic. The equations derived in the thesis also take into consideration the effect of jitter on delay, thereby removing the requirement for rate controllers or traffic shaping within the network. In addition to providing more accurate delay bounds, the problem of providing fault tolerance to channels with guaranteed quality of service (QoS) is also explored. This thesis introduces a method for interleaving resource requirements of backup channels to reduce the overall resource reservations that are required to provide guaranteed fault recovery with the same QoS as the original failed channel. An algorithm for selecting recovery paths that can meet a channel's QoS requirements during recovery is also introduced. / Thesis (Ph.D.)--Computer Science, 2003.

Στην εργασία παρουσιάζεται η μελέτη, η πιστοποίηση και η εφαρμογή των μηχανισμών εκείνων που οδηγούν στην παροχή end-to-end QoS σε ενοποιημένα L2 και MPLS δίκτυα. / In a real IP network such as the Internet, the basic type of service offered is the best effort one. In the best effort service all packets are treated equally and there are no guarantees, variations or attempt to enforce justice. However, the network seeks to promote as much traffic as possible with “reasonable” quality. Network congestion is a frequent phenomenon that is introduced when a network device stores packets at the output queue as it receives more packets from those that it can transmit. During congestion packets suffer from delay and once the output queue becomes full, these packets are dropped. However, there are applications that require certain guarantees (especially regarding delay and packet drops) such as real-time data transmission applications (e.g. IP telephony, voice over IP) and videoconference. Quality guarantees for these applications can be ensured if they can cross empty or nearempty network queues. This can only be achieved through mechanisms that can ensure the capacity and availability of the network queues. A means to provide quality guarantees to certain types of traffic is the special management of certain packets compared to the other. At this point the term Quality of Service (QoS) is introduced. A definition for QoS is: "the ability of a network element to provide a level of guarantee to a subset of traffic that ensures that the requirements of the service can be achieved with a defined (high) probability". In reality, the mechanisms of QoS do not provide larger network capacity or something similar, but they rather provide better network management so that it can be used more effectively and it can meet and address the requirements of the applications. In recent years, efforts have focused on providing quality of service at the network layer (Layer 3) so that it can be also applied on the Internet. Using architectures such as IntServ and DiffServ it is now possible to provide quality service at the network layer. However, the requirement for end-to-end QoS along with the expansion of networks towards switching equipment, creates the need for the application of QoS in the next lower layer, that is the data link layer (Layer 2). Hence, it should be borne in mind that the interoperability between the network and data link layers will lead to the provision of a single, transparent level of QoS. Based on the above, it becomes clear that in order to achieve end-to-end QoS, apart from the need for extension of the QoS to the data link layer there is also the need and requirement of interoperability with existing implementations in the network layer. In this direction, this dissertation focuses on studying the application of QoS to the data link layer. Furthermore, given the provision of quality of service to the IP layer, this dissertation considers the integration of QoS provision at Layer 2 and Layer 3. Therefore, the objective of this dissertation is twofold: a) QoS provision over Layer2-Ethernet networks and b) QoS provision over Layer 2 VPNs. For the implementation of Layer 2 QoS over Ethernet networks the IEEE 802.1p standard has been proposed. This standard has 3 bits length and is part of Tag Control Information field. During this dissertation performance tests were carried out on switches sorting traffic under CoS, which results in 8 different classes of traffic. Furthermore, queue configuration techniques on switches have been studied along with the cases of per port/per 802 .1q priorities and traffic classification. For the implementation of L2 QoS over VPNs there are techniques that are strongly related to the VPN type. This dissertation presents cases that L2 MPLS VPNs are used for the provisioning of either point-to-point (EoMPLS) or point-tomultipoint (VPLS) VPNs. In addition, research has been carried out for the extension of QoS provision over L2 MPLS VPNs to end-points that is purely L2 domain. The analysis at L2 domain was realized with the IEEE 802.1 p standard. Furthermore, the ability to provide QoS over multipoint L2 VPNs has been studied. Initially, the focus was on L3 devices (routers) and it was then extended to L2 using IEEE 802.1 p. Thus, the overall implementation was based on the combined use of 802.1p, DSCP and MPLS EXP. In addition, this dissertation presents methods, techniques and configurations of switches and routers that allow for the expansion of QoS from the network layer at a lower layer, thereby providing a consistent QoS level both at Layer 3 and Layer 2. Finally, the automated delivery/provision of these services in a real production network, GRNET, is presented. More specifically, the modeling of L2 QoS-enabled switches is described along with the automated configuration production for providing integrated QoS and issues related to the discovery, mapping and monitoring of QoS in switches using the SNMP protocol. The effectiveness of Layer 2 QoS mechanisms was tested and reinforced with experiments, which were conducted small scale at first in the lab and in the department of the University and then moved on to large scale at the production network of GRNET. The experiments showed that regardless of the expansion of a network towards Layer 2 devices, it is feasible to provide a unified QoS framework. All the above resulted in the provisioning of end-to-end QoS services at GRNET’s network.

This thesis deals with the quality of service (QoS) mechanism in Evolved Packet System (EPS). The basic architecture and characteristics of the EPS are introduced. The concept of QoS is defined. Furthermore, QoS related entities, their functions and corresponding procedures are described. The methodology for measuring and evaluating of end-to-end QoS as seen from a user’s point of view is designed. This methodology is used to measure QoS of experimental cellular network at UTKO FEEC BUT. Finally, configuration change for optimizing the QoS support of experimental cellular network is designed.

This diploma thesis deals with OMNeT++, a simulation tool for network traffic simulations. It describes its installation, basic functions and its use together with an extension INET framework. Further it shows construction of a simple simulation, its parts and analysis of the traffic output. Briefly it explains Quality of Service, which is further practically implemented in visual simulations. The paper further contains two laboratory exercises with model protocols.

No / Quality of Service (QoS) is of extreme importance in accommodating the increasingly diverse range of services and types of traffic in present day communication networks and delay is one of the most important QoS metrics. This paper presents a new approach for constraining queueing delay in a buffer to a specified level as the arrival rate changes with time. A discrete-time control algorithm is presented that operates on a buffer (queue) which incorporates a moveable threshold. An algorithm is developed that controls the delay by dynamically adjusting the threshold which, in turn, controls the arrival rate. The feasibility of the system is examined using both theoretical analysis and simulation.

In the recent decade, peer-to-peer (P2P) technology has greatly enhanced the scalability of multimedia streaming on the Internet by enabling efficient cooperation among end-users. However, existing streaming applications are plagued by the problems of long playback latency and long churn-induced delays. First of all, many streaming applications, such as IPTV and video conferencing, have rigorous constraints on end-to-end delays. Moreover, churn-induced delays, including delays from channel switching and streaming recovery, in current P2P streaming applications are typically in the scale of 10-60 seconds, which is far below the favorable user experience as in cable TV systems. These two issues in terms of playback latency and churn-induced delays have hindered the extensive commercial deployment of P2P systems. Motivated by this, in this dissertation, we focus on reducing delays in P2P live streaming systems. Specifically, we propose solutions for reducing delays in P2P live streaming systems in four problem spaces: (1) minimizing the maximum end-to-end delay in P2P streaming; (2) minimizing the average end-to-end delay in P2P streaming; (3) minimizing the average delay in multi-channel P2P streaming; and (4) reducing churn-induced delays. We devise a streaming scheme to minimize the maximum end-to-end streaming delay under a mesh-based overlay network paradigm. We call this problem, the MDPS problem. We formulate the MDPS problem and prove its NP-completeness. We then present a polynomial-time approximation algorithm, called Fastream-I, for this problem, and show that the performance of Fastream-I is bounded by a ratio of O(SQRT(log n)), where n is the number of peers in the system. We also develop a distributed version of Fastream-I that can adapt to network dynamics. Our simulation study reveals the effectiveness of Fastream-I, and shows a reasonable message overhead. While Fastream-I yields the minimum maximum end-to-end streaming delay (within a factor of O(SQRT(log n)), in many P2P settings, users may desire the minimum average end-to-end P2P streaming delay. Towards this, we devise a streaming scheme which optimizes the bandwidth allocation to achieve the minimum average end-to-end P2P streaming delay. We call this problem, the MADPS problem. We first develop a generic analytical framework for the MADPS problem. We then present Fastream-II as a solution to the MADPS problem. The core part of Fastream-II is a fast approximation algorithm, called APX-Fastream-II, based on primal-dual schema. We prove that the performance of APX-Fastream-II is bounded by a ratio of 1+w, where w is an adjustable input parameter. Furthermore, we show that the flexibility of w provides a trade-off between the approximation factor and the running time of Fastream-II. The third problem space of the dissertation is minimizing the average delay in multi-channel P2P streaming systems. Toward this, we present an algorithm, called Fastream-III. To reduce the influence from frequent channel-switching behavior, we build Fastream-III for the view-upload decoupling (VUD) model, where the uploaded content from a serving node is independent of the channel it views. We devise an approximation algorithm based on primal-dual schema for the critical component of Fastream-III, called APX-Fastream-III. In contrast to APX-Fastream-II, APX-Fastream-III addresses the extra complexity in the multichannel scenario and maintains the approximation bound by a ratio of 1+w. Besides playback lag, delays occurring in P2P streaming may arise from two other factors: node churn and channel switching. Since both stem from the re-connecting request in churn, we call them churn-induced delays. Optimizing churn-induced delays is the dissertation's fourth problem space. Toward this, we propose NAP, a novel agent-based P2P scheme, that provides preventive connections to all channels. Each channel in NAP selects powerful peers as agents to represent the peers in the channel to minimize control and message overheads. Agents distill the bootstrapping peers with superior bandwidth and lifetime expectation to quickly serve the viewer in the initial period of streaming. We build a queueing theory model to analyze NAP. Based on this model, we numerically compare NAP's performance with past efforts. The results of the numerical analysis reveal the effectiveness of NAP. / Ph. D.

Le Cloud Computing est de plus en plus reconnu comme une nouvelle façon d'utiliser, à la demande, les services de calcul, de stockage et de réseau d'une manière transparente et efficace. Dans cette thèse, nous abordons le problème d'ordonnancement de workflows sur les infrastructures distribuées hétérogènes du Cloud Computing. Les approches d'ordonnancement de workflows existantes dans le Cloud se concentrent principalement sur l'optimisation biobjectif du makespan et du coût. Dans cette thèse, nous proposons des algorithmes d'ordonnancement de workflows basés sur des métaheuristiques. Nos algorithmes sont capables de gérer plus de deux métriques de QoS (Quality of Service), notamment, le makespan, le coût, la fiabilité, la disponibilité et l'énergie dans le cas de ressources physiques. En outre, ils traitent plusieurs contraintes selon les exigences spécifiées dans le SLA (Service Level Agreement). Nos algorithmes ont été évalués par simulation en utilisant (1) comme applications: des workflows synthétiques et des workflows scientifiques issues du monde réel ayant des structures différentes; (2) et comme ressources Cloud: les caractéristiques des services de Amazon EC2. Les résultats obtenus montrent l'efficacité de nos algorithmes pour le traitement de plusieurs QoS. Nos algorithmes génèrent une ou plusieurs solutions dont certaines surpassent la solution de l'heuristique HEFT sur toutes les QoS considérées, y compris le makespan pour lequel HEFT est censé donner de bons résultats. / Cloud Computing is increasingly recognized as a new way to use on-demand, computing, storage and network services in a transparent and efficient way. In this thesis, we address the problem of workflows scheduling on distributed heterogeneous infrastructure of Cloud Computing. The existing workflows scheduling approaches mainly focus on the bi-objective optimization of the makespan and the cost. In this thesis, we propose news workflows scheduling algorithms based on metaheuristics. Our algorithms are able to handle more than two QoS (Quality of Service) metrics, namely, makespan, cost, reliability, availability and energy in the case of physical resources. In addition, they address several constraints according to the specified requirements in the SLA (Service Level Agreement). Our algorithms have been evaluated by simulations. We used (1) synthetic workflows and real world scientific workflows having different structures, for our applications; and (2) the features of Amazon EC2 services for our Cloud. The obtained results show the effectiveness of our algorithms when dealing multiple QoS metrics. Our algorithms produce one or more solutions which some of them outperform the solution produced by HEFT heuristic over all the QoS considered, including the makespan for which HEFT is supposed to give good results.

International Telemetering Conference Proceedings / October 18-21, 2004 / Town & Country Resort, San Diego, California / In the past few years, an evolution has been occurring in test range network topologies. With the proliferation of IP-based networks at the desktop, range officers are seeking ways to extend IP-based networks to the test range, to derive the cost and operational benefits offered with IP technology. This transition is not without its own set of problems. The operational transition from the traditional, ATM-based ranges to IP-based ranges must be addressed. In many cases, it is desired to maintain the ATM range, and add IP capabilities as time and budget permits. The net result is that frequently a mixed protocol network emerges. Terawave Communications has been developing telemetry transport solutions for both ATM and IP-based networks, along with technology to enable convergence of additional services such as video, voice, and data across test ranges. Terawave has developed a solution for various network topologies from ATM-only and IP-only to mixed protocol implementations, which supports end-to-end interworking of telemetry, video, and additional services over mixed protocol networks. In this paper, Terawave will detail the implementation decisions made in the course of application development, and share a framework for enabling seamless intra- and inter- range communication of telemetry and mixed services.

With the rising popularity of Voice over IP (VoIP) services, understanding the effects of a global network on Quality of Service is critical for the providers of VoIP applications. This thesis builds on a model that analyzes the round trip time, packet delay jitter, and packet loss between endpoints on an Autonomous System (AS) level, extending it by mapping AS pairs onto an Internet topology. This model is used to produce a mean opinion score estimate. The mapping is introduced to reduce the size of the problem in order to improve computation times and improve accuracy of estimates. The results of testing show that estimating mean opinion score from this model is not desirable. It also shows that the path mapping does not affect accuracy, but does improve computation times as the input data grows in volume.

Iraq may be considered as the largest wireless market in the Gulf region. A key driving factor in the market of wireless communication, it has seen enormous growth in the mobile phone market over the last five years leading to almost 24 million subscribers in 2011. Moreover, there are several technologies and services working in Iraq; three GSM Operators, three CDMA national operators and three CDMA provinces operators. The recent growth in the mobile phone market is based on the Global System for Mobile (GSM) communications and Code Division Multiple Access (CDMA) standards creating the next-generation wireless technologies in the Iraqi Wireless Communication market. One of the essential issues of this research is to investigate the performance of the decreased Quality Of Service (QoS) caused by interferences in the services on GSM/CDMA operators in Iraq. Many issues should be studied and taken into consideration, such as; does the Multi-Coalition Forces cause the interferences, jamming, higher rate of calls drop and false ringing; or are they caused by bad design and planning? Do we need to optimise our network due to the large number of users? All these factors are investigated and the measurements of most service providers and government agencies will be gathered. A detailed analysis was included from the providers with measurements of performance and the reasons for the deterioration of wireless services. The novel contributions of this thesis is the extensive radio measurement campaign over the three mobile an CDMA operator networks and the analysis and recommendations that were drawn to suggest the best approach to improve the QoS of Wireless communication technologies. Awareness of actual reasons behind the deterioration of services will be raised to the Iraqi Government, CMC and the wireless service providers.