988 resultados para IP Network


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Mestrado em Engenharia Electrotécnica e de Computadores

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As the wireless cellular market reaches competitive levels never seen before, network operators need to focus on maintaining Quality of Service (QoS) a main priority if they wish to attract new subscribers while keeping existing customers satisfied. Speech Quality as perceived by the end user is one major example of a characteristic in constant need of maintenance and improvement. It is in this topic that this Master Thesis project fits in. Making use of an intrusive method of speech quality evaluation, as a means to further study and characterize the performance of speech codecs in second-generation (2G) and third-generation (3G) technologies. Trying to find further correlation between codecs with similar bit rates, along with the exploration of certain transmission parameters which may aid in the assessment of speech quality. Due to some limitations concerning the audio analyzer equipment that was to be employed, a different system for recording the test samples was sought out. Although the new designed system is not standard, after extensive testing and optimization of the system's parameters, final results were found reliable and satisfactory. Tests include a set of high and low bit rate codecs for both 2G and 3G, where values were compared and analysed, leading to the outcome that 3G speech codecs perform better, under the approximately same conditions, when compared with 2G. Reinforcing the idea that 3G is, with no doubt, the best choice if the costumer looks for the best possible listening speech quality. Regarding the transmission parameters chosen for the experiment, the Receiver Quality (RxQual) and Received Energy per Chip to the Power Density Ratio (Ec/N0), these were subject to speech quality correlation tests. Final results of RxQual were compared to those of prior studies from different researchers and, are considered to be of important relevance. Leading to the confirmation of RxQual as a reliable indicator of speech quality. As for Ec/N0, it is not possible to state it as a speech quality indicator however, it shows clear thresholds for which the MOS values decrease significantly. The studied transmission parameters show that they can be used not only for network management purposes but, at the same time, give an expected idea to the communications engineer (or technician) of the end-to-end speech quality consequences. With the conclusion of the work new ideas for future studies come to mind. Considering that the fourth-generation (4G) cellular technologies are now beginning to take an important place in the global market, as the first all-IP network structure, it seems of great relevance that 4G speech quality should be subject of evaluation. Comparing it to 3G, not only in narrowband but also adding wideband scenarios with the most recent standard objective method of speech quality assessment, POLQA. Also, new data found on Ec/N0 tests, justifies further research studies with the intention of validating the assumptions made in this work.

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El projecte consisteix en l' aplicació practica i real de les tecnologies sense fils actuals per l' ampliació dels serveis d' accés a la xarxa TCP / IP en una empresa

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Today, information technology is strategically important to the goals and aspirations of the business enterprises, government and high-level education institutions – university. Universities are facing new challenges with the emerging global economy characterized by the importance of providing faster communication services and improving the productivity and effectiveness of individuals. New challenges such as provides an information network that supports the demands and diversification of university issues. A new network architecture, which is a set of design principles for build a network, is one of the pillar bases. It is the cornerstone that enables the university’s faculty, researchers, students, administrators, and staff to discover, learn, reach out, and serve society. This thesis focuses on the network architecture definitions and fundamental components. Three most important characteristics of high-quality architecture are that: it’s open network architecture; it’s service-oriented characteristics and is an IP network based on packets. There are four important components in the architecture, which are: Services and Network Management, Network Control, Core Switching and Edge Access. The theoretical contribution of this study is a reference model Architecture of University Campus Network that can be followed or adapted to build a robust yet flexible network that respond next generation requirements. The results found are relevant to provide an important complete reference guide to the process of building campus network which nowadays play a very important role. Respectively, the research gives university networks a structured modular model that is reliable, robust and can easily grow.

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The Thesis focused on hardware based Load balancing solution of web traffic through a load balancer F5 content switch. In this project, the implemented scenario for distributing HTTPtraffic load is based on different CPU usages (processing speed) of multiple member servers.Two widely used load balancing algorithms Round Robin (RR) and Ratio model (weighted Round Robin) are implemented through F5 load balancer. For evaluating the performance of F5 content switch, some experimental tests has been taken on implemented scenarios using RR and Ratio model load balancing algorithms. The performance is examined in terms of throughput (bits/sec) and Response time of member servers in a load balancing pool. From these experiments we have observed that Ratio Model load balancing algorithm is most suitable in the environment of load balancing servers with different CPU usages as it allows assigning the weight according to CPU usage both in static and dynamic load balancing of servers.

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T'his dissertation proposes alternative models to allow the interconnectioin of the data communication networks of COSERN Companhia Energética do Rio Grande do Norte. These networks comprise the oorporative data network, based on TCP/IP architecture, and the automation system linking remote electric energy distribution substations to the main Operatin Centre, based on digital radio links and using the IEC 60870-5-101 protoco1s. The envisaged interconnection aims to provide automation data originated from substations with a contingent route to the Operation Center, in moments of failure or maintenance of the digital radio links. Among the presented models, the one chosen for development consists of a computational prototype based on a standard personal computer, working under LINUX operational system and running na application, developesd in C language, wich functions as a Gateway between the protocols of the TCP/IP stack and the IEC 60870-5-101 suite. So, it is described this model analysis, implementation and tests of functionality and performance. During the test phase it was basically verified the delay introduced by the TCP/IP network when transporting automation data, in order to guarantee that it was cionsistent with the time periods present on the automation network. Besides , additional modules are suggested to the prototype, in order to handle other issues such as security and prioriz\ation of the automation system data, whenever they are travesing the TCP/IP network. Finally, a study hás been done aiming to integrate, in more complete way, the two considered networks. It uses IP platform as a solution of convergence to the communication subsystem of na unified network, as the most recente market tendencies for supervisory and other automation systems indicate

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There is a wide range of telecommunications services that transmit voice, video and data through complex transmission networks and in some cases, the service has not an acceptable quality level for the end user. In this sense the study of methods for assessing video quality and voice have a very important role. This paper presents a classification scheme, based on different criteria, of the methods and metrics that are being studied in recent years. This paper presents how the video quality is affected by degradation in the transmission channel in two kinds of services: Digital TV (ISDB-TB) due the fading in the air interface and video streaming service on an IP network due packet loss. For Digital TV tests was set up a scenario where the digital TV transmitter is connected to an RF channel emulator, where are inserted different fading models and at the end, the videos are saved in a mobile device. The tests of streaming video were performed in an isolated scenario of IP network, which are scheduled several network conditions, resulting in different qualities of video reception. The video quality assessment is performed using objective assessment methods: PSNR, SSIM and VQM. The results show how the losses in the transmission channel affects the quality of end-user experience on both services studied.

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Service providers make use of cost-effective wireless solutions to identify, localize, and possibly track users using their carried MDs to support added services, such as geo-advertisement, security, and management. Indoor and outdoor hotspot areas play a significant role for such services. However, GPS does not work in many of these areas. To solve this problem, service providers leverage available indoor radio technologies, such as WiFi, GSM, and LTE, to identify and localize users. We focus our research on passive services provided by third parties, which are responsible for (i) data acquisition and (ii) processing, and network-based services, where (i) and (ii) are done inside the serving network. For better understanding of parameters that affect indoor localization, we investigate several factors that affect indoor signal propagation for both Bluetooth and WiFi technologies. For GSM-based passive services, we developed first a data acquisition module: a GSM receiver that can overhear GSM uplink messages transmitted by MDs while being invisible. A set of optimizations were made for the receiver components to support wideband capturing of the GSM spectrum while operating in real-time. Processing the wide-spectrum of the GSM is possible using a proposed distributed processing approach over an IP network. Then, to overcome the lack of information about tracked devices’ radio settings, we developed two novel localization algorithms that rely on proximity-based solutions to estimate in real environments devices’ locations. Given the challenging indoor environment on radio signals, such as NLOS reception and multipath propagation, we developed an original algorithm to detect and remove contaminated radio signals before being fed to the localization algorithm. To improve the localization algorithm, we extended our work with a hybrid based approach that uses both WiFi and GSM interfaces to localize users. For network-based services, we used a software implementation of a LTE base station to develop our algorithms, which characterize the indoor environment before applying the localization algorithm. Experiments were conducted without any special hardware, any prior knowledge of the indoor layout or any offline calibration of the system.

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Service providers make use of cost-effective wireless solutions to identify, localize, and possibly track users using their carried MDs to support added services, such as geo-advertisement, security, and management. Indoor and outdoor hotspot areas play a significant role for such services. However, GPS does not work in many of these areas. To solve this problem, service providers leverage available indoor radio technologies, such as WiFi, GSM, and LTE, to identify and localize users. We focus our research on passive services provided by third parties, which are responsible for (i) data acquisition and (ii) processing, and network-based services, where (i) and (ii) are done inside the serving network. For better understanding of parameters that affect indoor localization, we investigate several factors that affect indoor signal propagation for both Bluetooth and WiFi technologies. For GSM-based passive services, we developed first a data acquisition module: a GSM receiver that can overhear GSM uplink messages transmitted by MDs while being invisible. A set of optimizations were made for the receiver components to support wideband capturing of the GSM spectrum while operating in real-time. Processing the wide-spectrum of the GSM is possible using a proposed distributed processing approach over an IP network. Then, to overcome the lack of information about tracked devices’ radio settings, we developed two novel localization algorithms that rely on proximity-based solutions to estimate in real environments devices’ locations. Given the challenging indoor environment on radio signals, such as NLOS reception and multipath propagation, we developed an original algorithm to detect and remove contaminated radio signals before being fed to the localization algorithm. To improve the localization algorithm, we extended our work with a hybrid based approach that uses both WiFi and GSM interfaces to localize users. For network-based services, we used a software implementation of a LTE base station to develop our algorithms, which characterize the indoor environment before applying the localization algorithm. Experiments were conducted without any special hardware, any prior knowledge of the indoor layout or any offline calibration of the system.

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A real-time surveillance system for IP network cameras is presented. Motion, part-body, and whole-body detectors are efficiently combined to generate robust and fast detections, which feed multiple compressive trackers. The generated trajectories are then improved using a reidentification strategy for long term operation.

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El desarrollo de las redes IP ha marcado un creciente interés por unificar todas las comunicaciones sobre una misma infraestructura aprovechando así el cableado existente. Con esta idea nació la tecnología conocida como VoIP (Voice Over Internet Protocol) que consiste en la trasmisión de la voz sobre paquetes IP. El objetivo principal de este proyecto es el diseño e implementación de una infraestructura de voz sobre IP que utilice una red de datos existente. La primera parte del proyecto estará formada por un estudio detallado de los factores que influyen en esta tecnología: codecs, protocolos y otros factores a tener en cuenta. Tras esta parte, aprovechando la experiencia adquirida durante casi tres años en una empresa integradora de servicios, se realizará un caso de estudio en el que, bajo las premisas impuestas por un supuesto cliente, se diseñará una solución que cumpla todos los requisitos y aporte un valor añadido sobre el sistema de telefonía que posee el cliente. El diseño de la mejor solución se hará utilizando productos del fabricante Cisco Systems y, además del coste económico, se valorarán los esfuerzos personales que costará montar dicha solución, incluyendo un valor añadido como es el dotar de buzón de voz y mensajería a todos los usuarios. La última parte del caso de estudio consistirá en la implementación de la infraestructura anteriormente diseñada adquiriendo conocimientos sobre virtualización de servidores utilizando productos de la compañía VMWare (especialista en virtualización), instalación y parametrización de aplicativos de Cisco y, finalmente, la interconexión con la red pública a través de gateways para poder hacer llamadas al exterior. El proyecto finalizará con la presentación de unas conclusiones finales y la exposición de unas líneas futuras de trabajo. ABSTRACT. The IP network development has marked a growing interest in unifying all communications over the same infrastructure taking advantage of the existing wiring. From this idea, a technology was born known as VoIP (Voice Over Internet Protocol) which consists of the transmission of voice packets over the IP network. The main goal of this project is the design and implementation of a VoIP infrastructure for transmitting voice packets over the existing wired network infrastructure on the client. The first part of the project will consist of a detailed study of the factors influencing this technology: codecs, protocols, and other factors to consider. After this part, drawing on the experience gained during nearly three years in an integrated services company, a case study will be made under the premises imposed for a supposed client. A solution that serves all the requirements will be designed and provide an added value on the customer’s telephone system. The design of the best solution will be done using Cisco Systems products and besides the economic cost of the whole solution, the personal effort cost will be valued. The added value named before will be provided with two important applications such as voice mail and instant messaging for all users. The last part of the case study will consist in the implementation of an infrastructure designed to acquire knowledge about virtualization, using VMWare company products (specialist in virtualization), installation and configuration of applications from Cisco Systems and finally the interconnection with the public network through gateway routers in order to make external calls. The project will end with the presentation of final conclusions and exposing future working lines.

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Este proyecto consiste en el diseño completo, de una red de distribución de TDT, a nivel local, mediante difusión SFN, Single Frequency Network. Este tipo de difusión, tiene la capacidad de difundir los servicios de televisión en una única frecuencia, cubriendo un área, ya sea local o estatal, aprovechando en las zonas de interferencia los rebotes de la señal y así evitar el uso de una frecuencia distinta por cada centro de emisión, todos los que componen un área de cobertura. Para el diseño de la red, se ha optado por diseñar una red IP, mediante distribución multicast, ya que esta es la tecnología imperante a día de hoy, quedando obsoleta ya, la distribución analógica, ya que consume muchos más recursos y por consiguiente mucho más costosa de implementar. El documento se divide en cuatro capítulos. En el primer capítulo se realizará una introducción teórica a las redes de distribución SFN, centrándose en el cálculo de los retardos, punto fundamental en el diseño de este tipo de redes. Se continuará unas nociones básicas de redes IP y el protocolo multicast, en el que se basa el trasporte de la señal. El capítulo dos, se centra en el diseño de la red, desde los centros de producción, donde se generan los programas a emitir, hasta los diferentes centros de difusión que cubrirán todo el área de cobertura requerida, pasando por el centro de multiplexación, donde se sitúa la cabecera que compondrá el múltiplex a difundir. Se describirán los equipos y el diseño de los distintos centros que conforman la red, centros de producción, multiplexación y difusión. A demás se realizará el cálculo de retardo de la señal, necesario en este tipo de redes. Se continuará con el capítulo tres, donde se describirá la configuración de la red, tanto a nivel de equipamiento, como el diseño y asignación IP de toda la red, separando la red de servicio de la red de gestión para una mayor confiabilidad y eficiencia de la red. Se finalizará con la descripción de la gestión de la red, que mediante diferentes herramientas, proporcionan un monitoreado en tiempo real de todo el sistema, dando la posibilidad de adelantarsey previniendo posibles incidencias que, puedan causar alguna deficiencia en el servicio que se entrega al usuario final. ABSTRACT. This project involves the complete design of a network´s TDT distribution, locally, by broadcast SFN (Single Frequency Network). This type of broadcast, has the ability to broadcast television´s services on a single frequency, covering an area, whether local or state, drawing on the interference zones, signal´s rebounds, to avoid the use of a different frequency each broadcast center, all those who make a coverage area. For the design of the network, has been chosen to design an IP network using multicast distribution, since this is the prevailing technology today, as the analogue distribution, consumes more resources and therefore, much more costly to implement. The document is divided into four chapters. In the first chapter you can find a theoretical introduction to SFN distribution networks, focusing on the calculation of delays, fundamental point, in the design of these networks. A basic understanding of IP networks and the multicast protocol, in which the transport of the signal is based, will continue. Chapter two focuses on the design of the network, from production centers, where the programs are created to broadcast, to different distribution centers covering the entire area of coverage required, through the multiplexing center, where the head is located, which comprise the multiplex. Also, the equipment and design of the various centers in the network, production centers, multiplexing center and distribution centers, are described. Furthermore, the calculation of signal delay, necessary in such networks, is performed. We will continue with the chapter three, where the network configuration will be described, both in termsofequipment, such as design IP mapping of the entire network, separating the service network and management network, for increased the reliability and efficiency of the network. It will be completed with the description of the management of the network, using different tools provide real-time monitoring of the entire system, making it possible, to anticipate and prevent any incidents that might cause a deficiency in the service being delivered to final user.

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This paper describes work conducted as a joint collaboration between the Virtual Design Team (VDT) research group at Stanford University (USA) , the Systems Engineering Group (SEG) at De Montfort University (UK) and Elipsis Ltd . We describe a new docking methodology in which we combine the use of two radically different types of organizational simulation tool. The VDT simulation tool operates on a standalone computer, and employs computational agents during simulated execution of a pre-defined process model (Kunz, 1998). The other software tool, DREAMS , operates over a standard TCP/IP network, and employs human agents (real people) during a simulated execution of a pre-defined process model (Clegg, 2000).

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Erasure control coding has been exploited in communication networks with an aim to improve the end-to-end performance of data delivery across the network. To address the concerns over the strengths and constraints of erasure coding schemes in this application, we examine the performance limits of two erasure control coding strategies, forward erasure recovery and adaptive erasure recovery. Our investigation shows that the throughput of a network using an (n, k) forward erasure control code is capped by r =k/n when the packet loss rate p ≤ (te/n) and by k(l-p)/(n-te) when p > (t e/n), where te is the erasure control capability of the code. It also shows that the lower bound of the residual loss rate of such a network is (np-te)/(n-te) for (te/n) < p ≤ 1. Especially, if the code used is maximum distance separable, the Shannon capacity of the erasure channel, i.e. 1-p, can be achieved and the residual loss rate is lower bounded by (p+r-1)/r, for (1-r) < p ≤ 1. To address the requirements in real-time applications, we also investigate the service completion time of different schemes. It is revealed that the latency of the forward erasure recovery scheme is fractionally higher than that of the scheme without erasure control coding or retransmission mechanisms (using UDP), but much lower than that of the adaptive erasure scheme when the packet loss rate is high. Results on comparisons between the two erasure control schemes exhibit their advantages as well as disadvantages in the role of delivering end-to-end services. To show the impact of the bounds derived on the end-to-end performance of a TCP/IP network, a case study is provided to demonstrate how erasure control coding could be used to maximize the performance of practical systems. © 2010 IEEE.

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Advertising investment and audience figures indicate that television continues to lead as a mass advertising medium. However, its effectiveness is questioned due to problems such as zapping, saturation and audience fragmentation. This has favoured the development of non-conventional advertising formats. This study provides empirical evidence for the theoretical development. This investigation analyzes the recall generated by four non-conventional advertising formats in a real environment: short programme (branded content), television sponsorship, internal and external telepromotion versus the more conventional spot. The methodology employed has integrated secondary data with primary data from computer assisted telephone interviewing (CATI) were performed ad-hoc on a sample of 2000 individuals, aged 16 to 65, representative of the total television audience. Our findings show that non-conventional advertising formats are more effective at a cognitive level, as they generate higher levels of both unaided and aided recall, in all analyzed formats when compared to the spot.