800 resultados para cloud computing datacenter performance QoS
Resumo:
We present observations of total cloud cover and cloud type classification results from a sky camera network comprising four stations in Switzerland. In a comprehensive intercomparison study, records of total cloud cover from the sky camera, long-wave radiation observations, Meteosat, ceilometer, and visual observations were compared. Total cloud cover from the sky camera was in 65–85% of cases within ±1 okta with respect to the other methods. The sky camera overestimates cloudiness with respect to the other automatic techniques on average by up to 1.1 ± 2.8 oktas but underestimates it by 0.8 ± 1.9 oktas compared to the human observer. However, the bias depends on the cloudiness and therefore needs to be considered when records from various observational techniques are being homogenized. Cloud type classification was conducted using the k-Nearest Neighbor classifier in combination with a set of color and textural features. In addition, a radiative feature was introduced which improved the discrimination by up to 10%. The performance of the algorithm mainly depends on the atmospheric conditions, site-specific characteristics, the randomness of the selected images, and possible visual misclassifications: The mean success rate was 80–90% when the image only contained a single cloud class but dropped to 50–70% if the test images were completely randomly selected and multiple cloud classes occurred in the images.
Resumo:
The spatial and temporal patterns of fog and low clouds along the South-Western African coast are characterized based on an evaluation of Meteosat SEVIRI satellite data. A technique for the detection of fog/low clouds in the region is introduced, and validated using 1 year of CALIOP cloud lidar products, showing reliable performance. The frequency of fog and low cloud in the study area is analyzed by systematic application of the technique to all available Meteosat SEVIRI scenes from 2004 to 2009, for 7:00 UTC and 14:00 UTC. The highest frequencies are encountered in the area around Walvis Bay, with a peak in the summer months. Fog and low clouds clear by 14:00 UTC almost everywhere over land.
Resumo:
In this paper we generalize the Continuous Adversarial Queuing Theory (CAQT) model (Blesa et al. in MFCS, Lecture Notes in Computer Science, vol. 3618, pp. 144–155, 2005) by considering the possibility that the router clocks in the network are not synchronized. We name the new model Non Synchronized CAQT (NSCAQT). Clearly, this new extension to the model only affects those scheduling policies that use some form of timing. In a first approach we consider the case in which although not synchronized, all clocks run at the same speed, maintaining constant differences. In this case we show that all universally stable policies in CAQT that use the injection time and the remaining path to schedule packets remain universally stable. These policies include, for instance, Shortest in System (SIS) and Longest in System (LIS). Then, we study the case in which clock differences can vary over time, but the maximum difference is bounded. In this model we show the universal stability of two families of policies related to SIS and LIS respectively (the priority of a packet in these policies depends on the arrival time and a function of the path traversed). The bounds we obtain in this case depend on the maximum difference between clocks. This is a necessary requirement, since we also show that LIS is not universally stable in systems without bounded clock difference. We then present a new policy that we call Longest in Queues (LIQ), which gives priority to the packet that has been waiting the longest in edge queues. This policy is universally stable and, if clocks maintain constant differences, the bounds we prove do not depend on them. To finish, we provide with simulation results that compare the behavior of some of these policies in a network with stochastic injection of packets.
Resumo:
In this paper we generalize the Continuous Adversarial Queuing Theory (CAQT) model (Blesa et al. in MFCS, Lecture Notes in Computer Science, vol. 3618, pp. 144–155, 2005) by considering the possibility that the router clocks in the network are not synchronized. We name the new model Non Synchronized CAQT (NSCAQT). Clearly, this new extension to the model only affects those scheduling policies that use some form of timing. In a first approach we consider the case in which although not synchronized, all clocks run at the same speed, maintaining constant differences. In this case we show that all universally stable policies in CAQT that use the injection time and the remaining path to schedule packets remain universally stable. These policies include, for instance, Shortest in System (SIS) and Longest in System (LIS). Then, we study the case in which clock differences can vary over time, but the maximum difference is bounded. In this model we show the universal stability of two families of policies related to SIS and LIS respectively (the priority of a packet in these policies depends on the arrival time and a function of the path traversed). The bounds we obtain in this case depend on the maximum difference between clocks. This is a necessary requirement, since we also show that LIS is not universally stable in systems without bounded clock difference. We then present a new policy that we call Longest in Queues (LIQ), which gives priority to the packet that has been waiting the longest in edge queues. This policy is universally stable and, if clocks maintain constant differences, the bounds we prove do not depend on them. To finish, we provide with simulation results that compare the behavior of some of these policies in a network with stochastic injection of packets.
Resumo:
We present a technique to estimate accurate speedups for parallel logic programs with relative independence from characteristics of a given implementation or underlying parallel hardware. The proposed technique is based on gathering accurate data describing one execution at run-time, which is fed to a simulator. Alternative schedulings are then simulated and estimates computed for the corresponding speedups. A tool implementing the aforementioned techniques is presented, and its predictions are compared to the performance of real systems, showing good correlation.
Resumo:
Performance studies of actual parallel systems usually tend to concéntrate on the effectiveness of a given implementation. This is often done in the absolute, without quantitave reference to the potential parallelism contained in the programs from the point of view of the execution paradigm. We feel that studying the parallelism inherent to the programs is interesting, as it gives information about the best possible behavior of any implementation and thus allows contrasting the results obtained. We propose a method for obtaining ideal speedups for programs through a combination of sequential or parallel execution and simulation, and the algorithms that allow implementing the method. Our approach is novel and, we argüe, more accurate than previously proposed methods, in that a crucial part of the data - the execution times of tasks - is obtained from actual executions, while speedup is computed by simulation. This allows obtaining speedup (and other) data under controlled and ideal assumptions regarding issues such as number of processor, scheduling algorithm and overheads, etc. The results obtained can be used for example to evalúate the ideal parallelism that a program contains for a given model of execution and to compare such "perfect" parallelism to that obtained by a given implementation of that model. We also present a tool, IDRA, which implements the proposed method, and results obtained with IDRA for benchmark programs, which are then compared with those obtained in actual executions on real parallel systems.
Resumo:
The 4CaaSt project aims at developing a PaaS framework that enables flexible definition, marketing, deployment and management of Cloud-based services and applications. The major innovations proposed by 4CaaSt are the blueprint and its lifecycle management, a one stop shop for Cloud services and a PaaS level resource management featuring elasticity. 4CaaSt also provides a portfolio of ready to use Cloud native services and Cloud-aware immigrant technologies.
Resumo:
The popularity of MapReduce programming model has increased interest in the research community for its improvement. Among the other directions, the point of fault tolerance, concretely the failure detection issue seems to be a crucial one, but that until now has not reached its satisfying level. Motivated by this, I decided to devote my main research during this period into having a prototype system architecture of MapReduce framework with a new failure detection service, containing both analytical (theoretical) and implementation part. I am confident that this work should lead the way for further contributions in detecting failures to any NoSQL App frameworks, and cloud storage systems in general.
Resumo:
Ubiquitous sensor network deployments, such as the ones found in Smart cities and Ambient intelligence applications, require constantly increasing high computational demands in order to process data and offer services to users. The nature of these applications imply the usage of data centers. Research has paid much attention to the energy consumption of the sensor nodes in WSNs infrastructures. However, supercomputing facilities are the ones presenting a higher economic and environmental impact due to their very high power consumption. The latter problem, however, has been disregarded in the field of smart environment services. This paper proposes an energy-minimization workload assignment technique, based on heterogeneity and application-awareness, that redistributes low-demand computational tasks from high-performance facilities to idle nodes with low and medium resources in the WSN infrastructure. These non-optimal allocation policies reduce the energy consumed by the whole infrastructure and the total execution time.
Resumo:
Data grid services have been used to deal with the increasing needs of applications in terms of data volume and throughput. The large scale, heterogeneity and dynamism of grid environments often make management and tuning of these data services very complex. Furthermore, current high-performance I/O approaches are characterized by their high complexity and specific features that usually require specialized administrator skills. Autonomic computing can help manage this complexity. The present paper describes an autonomic subsystem intended to provide self-management features aimed at efficiently reducing the I/O problem in a grid environment, thereby enhancing the quality of service (QoS) of data access and storage services in the grid. Our proposal takes into account that data produced in an I/O system is not usually immediately required. Therefore, performance improvements are related not only to current but also to any future I/O access, as the actual data access usually occurs later on. Nevertheless, the exact time of the next I/O operations is unknown. Thus, our approach proposes a long-term prediction designed to forecast the future workload of grid components. This enables the autonomic subsystem to determine the optimal data placement to improve both current and future I/O operations.
Resumo:
Telecommunications networks have been always expanding and thanks to it, new services have appeared. The old mechanisms for carrying packets have become obsolete due to the new service requirements, which have begun working in real time. Real time traffic requires strict service guarantees. When this traffic is sent through the network, enough resources must be given in order to avoid delays and information losses. When browsing through the Internet and requesting web pages, data must be sent from a server to the user. If during the transmission there is any packet drop, the packet is sent again. For the end user, it does not matter if the webpage loads in one or two seconds more. But if the user is maintaining a conversation with a VoIP program, such as Skype, one or two seconds of delay in the conversation may be catastrophic, and none of them can understand the other. In order to provide support for this new services, the networks have to evolve. For this purpose MPLS and QoS were developed. MPLS is a packet carrying mechanism used in high performance telecommunication networks which directs and carries data using pre-established paths. Now, packets are forwarded on the basis of labels, making this process faster than routing the packets with the IP addresses. MPLS also supports Traffic Engineering (TE). This refers to the process of selecting the best paths for data traffic in order to balance the traffic load between the different links. In a network with multiple paths, routing algorithms calculate the shortest one, and most of the times all traffic is directed through it, causing overload and packet drops, without distributing the packets in the other paths that the network offers and do not have any traffic. But this is not enough in order to provide the real time traffic the guarantees it needs. In fact, those mechanisms improve the network, but they do not make changes in how the traffic is treated. That is why Quality of Service (QoS) was developed. Quality of service is the ability to provide different priority to different applications, users, or data flows, or to guarantee a certain level of performance to a data flow. Traffic is distributed into different classes and each of them is treated differently, according to its Service Level Agreement (SLA). Traffic with the highest priority will have the preference over lower classes, but this does not mean it will monopolize all the resources. In order to achieve this goal, a set policies are defined to control and alter how the traffic flows. Possibilities are endless, and it depends in how the network must be structured. By using those mechanisms it is possible to provide the necessary guarantees to the real-time traffic, distributing it between categories inside the network and offering the best service for both real time data and non real time data. Las Redes de Telecomunicaciones siempre han estado en expansión y han propiciado la aparición de nuevos servicios. Los viejos mecanismos para transportar paquetes se han quedado obsoletos debido a las exigencias de los nuevos servicios, que han comenzado a operar en tiempo real. El tráfico en tiempo real requiere de unas estrictas garantías de servicio. Cuando este tráfico se envía a través de la red, necesita disponer de suficientes recursos para evitar retrasos y pérdidas de información. Cuando se navega por la red y se solicitan páginas web, los datos viajan desde un servidor hasta el usuario. Si durante la transmisión se pierde algún paquete, éste se vuelve a mandar de nuevo. Para el usuario final, no importa si la página tarda uno o dos segundos más en cargar. Ahora bien, si el usuario está manteniendo una conversación usando algún programa de VoIP (como por ejemplo Skype) uno o dos segundos de retardo en la conversación podrían ser catastróficos, y ninguno de los interlocutores sería capaz de entender al otro. Para poder dar soporte a estos nuevos servicios, las redes deben evolucionar. Para este propósito se han concebido MPLS y QoS MPLS es un mecanismo de transporte de paquetes que se usa en redes de telecomunicaciones de alto rendimiento que dirige y transporta los datos de acuerdo a caminos preestablecidos. Ahora los paquetes se encaminan en función de unas etiquetas, lo cual hace que sea mucho más rápido que encaminar los paquetes usando las direcciones IP. MPLS también soporta Ingeniería de Tráfico (TE). Consiste en seleccionar los mejores caminos para el tráfico de datos con el objetivo de balancear la carga entre los diferentes enlaces. En una red con múltiples caminos, los algoritmos de enrutamiento actuales calculan el camino más corto, y muchas veces el tráfico se dirige sólo por éste, saturando el canal, mientras que otras rutas se quedan completamente desocupadas. Ahora bien, esto no es suficiente para ofrecer al tráfico en tiempo real las garantías que necesita. De hecho, estos mecanismos mejoran la red, pero no realizan cambios a la hora de tratar el tráfico. Por esto es por lo que se ha desarrollado el concepto de Calidad de Servicio (QoS). La calidad de servicio es la capacidad para ofrecer diferentes prioridades a las diferentes aplicaciones, usuarios o flujos de datos, y para garantizar un cierto nivel de rendimiento en un flujo de datos. El tráfico se distribuye en diferentes clases y cada una de ellas se trata de forma diferente, de acuerdo a las especificaciones que se indiquen en su Contrato de Tráfico (SLA). EL tráfico con mayor prioridad tendrá preferencia sobre el resto, pero esto no significa que acapare la totalidad de los recursos. Para poder alcanzar estos objetivos se definen una serie de políticas para controlar y alterar el comportamiento del tráfico. Las posibilidades son inmensas dependiendo de cómo se quiera estructurar la red. Usando estos mecanismos se pueden proporcionar las garantías necesarias al tráfico en tiempo real, distribuyéndolo en categorías dentro de la red y ofreciendo el mejor servicio posible tanto a los datos en tiempo real como a los que no lo son.
Resumo:
En la última década ha aumentado en gran medida el interés por las redes móviles Ad Hoc. La naturaleza dinámica y sin infraestructura de estas redes, exige un nuevo conjunto de algoritmos y estrategias para proporcionar un servicio de comunicación fiable extremo a extremo. En el contexto de las redes móviles Ad Hoc, el encaminamiento surge como una de las áreas más interesantes para transmitir información desde una fuente hasta un destino, con la calidad de servicio de extremo a extremo. Debido a las restricciones inherentes a las redes móviles, los modelos de encaminamiento tradicionales sobre los que se fundamentan las redes fijas, no son aplicables a las redes móviles Ad Hoc. Como resultado, el encaminamiento en redes móviles Ad Hoc ha gozado de una gran atención durante los últimos años. Esto ha llevado al acrecentamiento de numerosos protocolos de encaminamiento, tratando de cubrir con cada uno de ellos las necesidades de los diferentes tipos de escenarios. En consecuencia, se hace imprescindible estudiar el comportamiento de estos protocolos bajo configuraciones de red variadas, con el fin de ofrecer un mejor encaminamiento respecto a los existentes. El presente trabajo de investigación muestra precisamente una solución de encaminamiento en las redes móviles Ad Hoc. Dicha solución se basa en el mejoramiento de un algoritmo de agrupamiento y la creación de un modelo de encaminamiento; es decir, un modelo que involucra la optimización de un protocolo de enrutamiento apoyado de un mecanismo de agrupación. El algoritmo mejorado, denominado GMWCA (Group Management Weighted Clustering Algorithm) y basado en el WCA (Weighted Clustering Algorithm), permite calcular el mejor número y tamaño de grupos en la red. Con esta mejora se evitan constantes reagrupaciones y que los jefes de clústeres tengan más tiempo de vida intra-clúster y por ende una estabilidad en la comunicación inter-clúster. En la tesis se detallan las ventajas de nuestro algoritmo en relación a otras propuestas bajo WCA. El protocolo de enrutamiento Ad Hoc propuesto, denominado QoS Group Cluster Based Routing Protocol (QoSG-CBRP), utiliza como estrategia el empleo de clúster y jerarquías apoyada en el algoritmo de agrupamiento. Cada clúster tiene un jefe de clúster (JC), quien administra la información de enrutamiento y la envía al destino cuando esta fuera de su área de cobertura. Para evitar que haya constantes reagrupamientos y llamados al algoritmo de agrupamiento se consideró agregarle un jefe de cluster de soporte (JCS), el que asume las funciones del JC, siempre y cuando este haya roto el enlace con los otros nodos comunes del clúster por razones de alejamiento o por desgaste de batería. Matemáticamente y a nivel de algoritmo se han demostrado las mejoras del modelo propuesto, el cual ha involucrado el mejoramiento a nivel de algoritmo de clustering y del protocolo de enrutamiento. El protocolo QoSG-CBRP, se ha implementado en la herramienta de simulación Network Simulator 2 (NS2), con la finalidad de ser comparado con el protocolo de enrutamiento jerárquico Cluster Based Routing Protocol (CBRP) y con un protocolo de enrutamiento Ad Hoc reactivo denominado Ad Hoc On Demand Distance Vector Routing (AODV). Estos protocolos fueron elegidos por ser los que mejor comportamiento presentaron dentro de sus categorías. Además de ofrecer un panorama general de los actuales protocolos de encaminamiento en redes Ad Hoc, este proyecto presenta un procedimiento integral para el análisis de capacidades de la propuesta del nuevo protocolo con respecto a otros, sobre redes que tienen un alto número de nodos. Estas prestaciones se miden en base al concepto de eficiencia de encaminamiento bajo parámetros de calidad de servicio (QoS), permitiendo establecer el camino más corto posible entre un nodo origen y un nodo destino. Con ese fin se han realizado simulaciones con diversos escenarios para responder a los objetivos de la tesis. La conclusiones derivadas del análisis de los resultados permiten evaluar cualitativamente las capacidades que presenta el protocolo dentro del modelo propuesto, al mismo tiempo que avizora un atractivo panorama en líneas futuras de investigación. ABSTRACT In the past decade, the interest in mobile Ad Hoc networks has greatly increased. The dynamic nature of these networks without infrastructure requires a new set of algorithms and strategies to provide a reliable end-to-end communication service. In the context of mobile Ad Hoc networks, routing emerges as one of the most interesting areas for transmitting information from a source to a destination, with the quality of service from end-to-end. Due to the constraints of mobile networks, traditional routing models that are based on fixed networks are not applicable to Ad Hoc mobile networks. As a result, the routing in mobile Ad Hoc networks has experienced great attention in recent years. This has led to the enhancement of many routing protocols, trying to cover with each one of them, the needs of different types of scenarios. Consequently, it is essential to study the behavior of these protocols under various network configurations, in order to provide a better routing scheme. Precisely, the present research shows a routing solution in mobile Ad Hoc networks. This solution is based on the improvement of a clustering algorithm, and the creation of a routing model, ie a model that involves optimizing a routing protocol with the support of a grouping mechanism. The improved algorithm called GMWCA (Group Management Weighted Clustering Algorithm) and based on the WCA (Weighted Clustering Algorithm), allows to calculate the best number and size of groups in the network. With this enhancement, constant regroupings are prevented and cluster heads are living longer intra-cluster lives and therefore stability in inter-cluster communication. The thesis details the advantages of our algorithm in relation to other proposals under WCA. The Ad Hoc routing protocol proposed, called QoS Group Cluster Based Routing Protocol (QoSG-CBRP), uses a cluster-employment strategy and hierarchies supported by the clustering algorithm. Each cluster has a cluster head (JC), who manages the routing information and sends it to the destination when is out of your coverage area. To avoid constant rearrangements and clustering algorithm calls, adding a support cluster head (JCS) was considered. The JCS assumes the role of the JC as long as JC has broken the link with the other nodes in the cluster for common restraining reasons or battery wear. Mathematically and at an algorithm level, the improvements of the proposed model have been showed, this has involved the improvement level clustering algorithm and the routing protocol. QoSG-CBRP protocol has been implemented in the simulation tool Network Simulator 2 (NS2), in order to be compared with the hierarchical routing protocol Cluster Based Routing Protocol (CBRP) and with the reactive routing protocol Ad Hoc On Demand Distance Vector Routing (AODV). These protocols were chosen because they showed the best individual performance in their categories. In addition to providing an overview of existing routing protocols in Ad Hoc networks, this project presents a comprehensive procedure for capacity analysis of the proposed new protocol with respect to others on networks that have a high number of nodes. These benefits are measured based on the concept of routing efficiency under the quality of service (QoS) parameters, thus allowing for the shortest possible path between a source node and a destination node. To meet the objectives of the thesis, simulations have been performed with different scenarios. The conclusions derived from the analysis of the results to assess qualitatively the protocol capabilities presented in the proposed model, while an attractive scenario for future research appears.
Resumo:
Providing QoS in the context of Ad Hoc networks includes a very wide field of application from the perspective of every level of the architecture in the network.In order for simulation studies to be useful, it is very important that the simulation results match as closely as possible with the test bed results. In this Paper, we study the throughput performance (parameter QoS) in Mobile Ad Hoc Networks (MANETs) and compares emulated test bed results with simulation results from NS2 (Network Simulator). The performance of the Mobile Ad Hoc Networks is very sensitive to the number of users and the offered load. When the number of users/offered load is high then the collisions increase resulting in larger wastage of the medium and lowering overall throughput. The aim of this research is to compare the throughput of Mobile Ad Hoc Networks using three different scenarios: 97, 100 and 120 users (nodes) using simulator NS2. By analyzing the graphs in MANETs, it is concluded When the number of users o nodes is increased beyond the certain limit, throughput decreases.
Resumo:
Today P2P faces two important challenges: design of mechanisms to encourage users’ collaboration in multimedia live streaming services; design of reliable algorithms with QoS provision, to encourage multimedia providers employ the P2P topology in commercial streaming services. We believe that these two challenges are tightly-related and there is much to be done with respect. This paper proposes a novel monetary incentive for P2P multimedia streaming. The incentive model classifies the users in groups according to the perceived video quality. We apply the model to a streaming system’s billing model in order to evaluate its feasibility and visualize its quantitative effect on the users’ motivation and the provider’s profit. We conclude that monetary incentive can boost up users’ cooperation, loyalty and enhance the overall system integrity and performance. Moreover the model defines the constraints for the provider’s cost and profit when the system is leveraged on the cloud. Considering those constraints, a multimedia content provider can adapt the billing model of his streaming service and achieve desirable discount-profit trade-off. This will moreover contribute to better promotion of the service, across the users on the Internet.
Resumo:
The inherent complexity of modern cloud infrastructures has created the need for innovative monitoring approaches, as state-of-the-art solutions used for other large-scale environments do not address specific cloud features. Although cloud monitoring is nowadays an active research field, a comprehensive study covering all its aspects has not been presented yet. This paper provides a deep insight into cloud monitoring. It proposes a unified cloud monitoring taxonomy, based on which it defines a layered cloud monitoring architecture. To illustrate it, we have implemented GMonE, a general-purpose cloud monitoring tool which covers all aspects of cloud monitoring by specifically addressing the needs of modern cloud infrastructures. Furthermore, we have evaluated the performance, scalability and overhead of GMonE with Yahoo Cloud Serving Benchmark (YCSB), by using the OpenNebula cloud middleware on the Grid’5000 experimental testbed. The results of this evaluation demonstrate the benefits of our approach, surpassing the monitoring performance and capabilities of cloud monitoring alternatives such as those present in state-of-the-art systems such as Amazon EC2 and OpenNebula.