Perceptionization of FM/FD/1 queuing model under various fuzzy numbers

We present a FM/FD/1 queuing model with unbounded limit under different fuzzy numbers. The arrival (landing) rate and service (administration) rate are thought to be fuzzy numbers such as triangular, trapezoidal and pentagonal fuzzy numbers. Because random event can only be observed in an uncertain manner, the fuzzy result of an uncertainty mapping is a fuzzy random variable. Consequently, it is conceivable to characterize the specific connection between randomness and fuzziness. The execution proportions of this lining miniature are fuzzified after that examined by utilizing α-cut estimations and DSW algorithm (Dong, Shah and Wong). Relating to different fuzzy numbers, the numerical precedents are delineated to test the attainability of this model (miniature). A comparative illustration corresponding to each fuzzy number is accomplished for various estimations of α

Quality aspects of Internet telephony

Internet telephony has had a tremendous impact on how people communicate. Many now maintain contact using some form of Internet telephony. Therefore the motivation for this work has been to address the quality aspects of real-world Internet telephony for both fixed and wireless telecommunication. The focus has been on the quality aspects of voice communication, since poor quality leads often to user dissatisfaction. The scope of the work has been broad in order to address the main factors within IP-based voice communication. The first four chapters of this dissertation constitute the background material. The first chapter outlines where Internet telephony is deployed today. It also motivates the topics and techniques used in this research. The second chapter provides the background on Internet telephony including signalling, speech coding and voice Internetworking. The third chapter focuses solely on quality measures for packetised voice systems and finally the fourth chapter is devoted to the history of voice research. The appendix of this dissertation constitutes the research contributions. It includes an examination of the access network, focusing on how calls are multiplexed in wired and wireless systems. Subsequently in the wireless case, we consider how to handover calls from 802.11 networks to the cellular infrastructure. We then consider the Internet backbone where most of our work is devoted to measurements specifically for Internet telephony. The applications of these measurements have been estimating telephony arrival processes, measuring call quality, and quantifying the trend in Internet telephony quality over several years. We also consider the end systems, since they are responsible for reconstructing a voice stream given loss and delay constraints. Finally we estimate voice quality using the ITU proposal PESQ and the packet loss process. The main contribution of this work is a systematic examination of Internet telephony. We describe several methods to enable adaptable solutions for maintaining consistent voice quality. We have also found that relatively small technical changes can lead to substantial user quality improvements. A second contribution of this work is a suite of software tools designed to ascertain voice quality in IP networks. Some of these tools are in use within commercial systems today

Performance analysis and network path characterization for scalable internet streaming

Delivering high-quality of video to end users over the best-effort Internet is a challenging task since quality of streaming video is highly subject to network conditions. A fundamental issue in this area is how real-time applications cope with network dynamics and adapt their operational behavior to offer a favorable streaming environment to end users. As an effort towards providing such streaming environment, the first half of this work focuses on analyzing the performance of video streaming in best-effort networks and developing a new streaming framework that effectively utilizes unequal importance of video packets in rate control and achieves a near-optimal performance for a given network packet loss rate. In addition, we study error concealment methods such as FEC (Forward-Error Correction) that is often used to protect multimedia data over lossy network channels. We investigate the impact of FEC on the quality of video and develop models that can provide insights into understanding how inclusion of FEC affects streaming performance and its optimality and resilience characteristics under dynamically changing network conditions. In the second part of this thesis, we focus on measuring bandwidth of network paths, which plays an important role in characterizing Internet paths and can benefit many applications including multimedia streaming. We conduct a stochastic analysis of an end-to-end path and develop novel bandwidth sampling techniques that can produce asymptotically accurate capacity and available bandwidth of the path under non-trivial cross-traffic conditions. In addition, we conduct comparative performance study of existing bandwidth estimation tools in non-simulated networks where various timing irregularities affect delay measurements. We find that when high-precision packet timing is not available due to hardware interrupt moderation, the majority of existing algorithms are not robust to measure end-to-end paths with high accuracy. We overcome this problem by using signal de-noising techniques in bandwidth measurement. We also develop a new measurement tool called PRC-MT based on theoretical models that simultaneously measures the capacity and available bandwidth of the tight link with asymptotic accuracy

Analysis and design of efficient techniques for video transmission in IEEE 802.11 wireless ad hoc networks

[EN] Wireless mobile ad ho networks, also known as MANETs, are omposed by independent mobile stations that ommuni ate without requiring any sort of infrastru ture for support. These networks are hara terized by variable bandwidth values and frequent path breaks, whi h are due to hannel noise, interferen e between stations and mobility. Su h fa tors require significant adaptation apabilites at different levels of the proto ol suites employed, enabling stations to qui kly respond to fast- hanging network onditions. Resear h on the most adequate proto- ols for the physi al, MAC and routing layers is still on-going, though some basi onsensus has already been rea hed and several testbeds have been setup around the world. To deploy real-time multimedia servi es, namely voi e and video, on top of su h an unreliable network environment is a very hallenging task. In this thesis we propose to a hieve that goal starting from urrently available Wi-Fi te hnology, and gradually finding the most adequate enhan ements to ea h proto ol layer of interest; we then ombine these enhan ements until we a hieve a omplete QoS framework for ad ho networks. By using urrently available te hnology we assure that the proposal of this thesis has an inherent high-level of appli ability on real life environments. Sin e our working field fo uses on video transmission over wireless ad ho networks, we will show how it is possible to support several QoS- onstrained video streams in MANET environments hara terized by moderate to high mobility levels, and by a significant amount of best efort traffic[ES] Las redes inalámbricas ad hoc, también conocidas como redes MANET, están compuestas por un conjunto de estaciones móviles independientes capaces de omunicarse entre sí sin necesidad de ningún tipo de infraestructura común de comunicaciones. Estas redes se caracterizan por tener un ancho de banda variable y pérdidas frecuentes de ruta que se pueden atribuir al ruido del anal inalámbrico, a la interferencia entre las estaciones móviles o bien a la movilidad de las estaciones. Dichos factores requieren una gran capacidad de adaptación en las diferentes capas de la arquitectura de protocolos, permitiendo a una estación responder rápidamente a posibles cambios bruscos en las condiciones de la red. A pesar de que aún se están realizando trabajos de investigación en bus a de los protocolos más adecuados para las capas físicas, a eso al medio (MAC) y encaminamiento, ha sido posible llegar a un nivel básico de consenso, lo cual ha permitido el despliegue de plataformas y entornos aplicados que utilizan tecnología de red MANET. Ofrecer servicios multimedia, como voz y vídeo, en redes con tan poca habilidad es un desafío importante. En esta tesis nos proponemos alcanzar este objetivo partiendo de la tecnología Wi-Fi actualmente disponible, encontrando de forma paulatina las mejoras más importantes en las diferentes capas de la arquitectura de red, para llegar, finalmente, a una solución integrada capaz de ofrecer calidad de servicio (QoS) en las redes MANET. Al utilizar la tecnología que disponemos actualmente nos aseguramos que las propuestas de esta tesis tengan un alto grado de aplicabilidad en entornos reales. Ya que la línea de trabajo de la tesis está aplicada a la transmisión de vídeo en redes MANET, demostraremos que es posible ofrecer calidad de servicio a varios flujos de vídeo en una red MANET caracterizada por altos grados de movilidad en sus nodos y un nivel significativo de tráfico o de tipo best effortTavares De Araujo Cesariny Calafate, CM. (2006). Analysis and design of efficient techniques for video transmission in IEEE 802.11 wireless ad hoc networks [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/135282TESI

Best effort measurement based congestion control

Abstract available: p.

Scheduling algorithms for next generation cellular networks

Next generation wireless and mobile communication systems are rapidly evolving to satisfy the demands of users. Due to spectrum scarcity and time-varying nature of wireless networks, supporting user demand and achieving high performance necessitate the design of efficient scheduling and resource allocation algorithms. Opportunistic scheduling is a key mechanism for such a design, which exploits the time-varying nature of the wireless environment for improving the performance of wireless systems. In this thesis, our aim is to investigate various categories of practical scheduling problems and to design efficient policies with provably optimal or near-optimal performance. An advantage of opportunistic scheduling is that it can effectively be incorporated with new communication technologies to further increase the network performance. We investigate two key technologies in this context. First, motivated by the current under-utilization of wireless spectrum, we characterize optimal scheduling policies for wireless cognitive radio networks by assuming that users always have data to transmit. We consider cooperative schemes in which secondary users share the time slot with primary users in return for cooperation, and our aim is to improve the primary systems performance over the non-cooperative case. By employing Lyapunov Optimization technique, we develop optimal scheduling algorithms which maximize the total expected utility and satisfy the minimum data rate requirements of the primary users. Next, we study scheduling problem with multi-packet transmission. The motivation behind multi-packet transmission comes from the fact that the base station can send more than one packets simultaneously to more than one users. By considering unsaturated queueing systems we aim to stabilize user queues. To this end, we develop a dynamic control algorithm which is able to schedule more than one users in a time slot by employing hierarchical modulation which enables multi-packet transmission. Through Lyapunov Optimization technique, we show that our algorithm is throughput-optimal. We also study the resulting rate region of developed policy and show that it is larger than that of single user scheduling. Despite the advantage of opportunistic scheduling, this mechanism requires that the base station is aware of network conditions such as channel state and queue length information of users. In the second part of this thesis, we turn our attention to the design of scheduling algorithms when complete network information is not available at the scheduler. In this regard, we study three sets of problems where the common objective is to stabilize user queues. Specifically, we first study a cellular downlink network by assuming that channels are identically distributed across time slots and acquiring channel state information of a user consumes a certain fraction of resource which is otherwise used for transmission of data. We develop a joint scheduling and channel probing algorithm which collects channel state information from only those users with su±ciently good channel quality. We also quantify the minimum number of users that must exist to achieve larger rate region than Max-Weight algorithm with complete channel state information. Next, we consider a more practical channel models where channels can be time-correlated (possibly non-stationary) and only a fixed number of channels can be probed. We develop learning based scheduling algorithm which tracks and predicts instantaneous transmission rates of users and makes a joint scheduling and probing decision based on the predicted rates rather than their exact values. We also characterize the achievable rate region of these policies as compared to Max-Weight policy with exact channel state information. Finally, we study a cellular uplink system and develop a fully distributed scheduling algorithm which can perform over general fading channels and does not require explicit control messages passing among the users. When continuous backoff time is allowed, we show that the proposed distributed algorithm can achieve the same performance as that of centralized Max-Weight algorithm in terms of both throughput and delay. When backoff time can take only discrete values, we show that our algorithm can perform well at the expense of low number of mini-slots for collision resolution