Dynamic Adaptive Streaming over HTTP (DASH) is a convenient approach to transfer videos in an adaptive and dynamic way to the user. Therefore, this system makes best use of the bandwidth available. In this thesis, we investigate Dynamic Adaptive Streaming over HTTP (DASH) based on data collected from the lab experiments and user’s experiments. The objectives include investigation of how three different DASH players behave at different network conditions and up to which limit the players are tolerating the disturbances. We summarized the outcome of lab experiments on DASH at different adverse conditions and checked the lab results with user quality of experience at different adverse conditions to see up to which extent the users could tolerate the disturbances in different DASH players.

Transmission of video content over wireless access networks (in particular, Wireless Local Area Networks (WLAN) and Third Generation Universal Mobile Telecommunication System (3G UMTS)) is growing exponentially and gaining popularity, and is predicted to expose new revenue streams for mobile network operators. However, the success of these video applications over wireless access networks very much depend on meeting the user’s Quality of Service (QoS) requirements. Thus, it is highly desirable to be able to predict and, if appropriate, to control video quality to meet user’s QoS requirements. Video quality is affected by distortions caused by the encoder and the wireless access network. The impact of these distortions is content dependent, but this feature has not been widely used in existing video quality prediction models. The main aim of the project is the development of novel and efficient models for video quality prediction in a non-intrusive way for low bitrate and resolution videos and to demonstrate their application in QoS-driven adaptation schemes for mobile video streaming applications. This led to five main contributions of the thesis as follows:(1) A thorough understanding of the relationships between video quality, wireless access network (UMTS and WLAN) parameters (e.g. packet/block loss, mean burst length and link bandwidth), encoder parameters (e.g. sender bitrate, frame rate) and content type is provided. An understanding of the relationships and interactions between them and their impact on video quality is important as it provides a basis for the development of non-intrusive video quality prediction models.(2) A new content classification method was proposed based on statistical tools as content type was found to be the most important parameter. (3) Efficient regression-based and artificial neural network-based learning models were developed for video quality prediction over WLAN and UMTS access networks. The models are light weight (can be implemented in real time monitoring), provide a measure for user perceived quality, without time consuming subjective tests. The models have potential applications in several other areas, including QoS control and optimization in network planning and content provisioning for network/service providers.(4) The applications of the proposed regression-based models were investigated in (i) optimization of content provisioning and network resource utilization and (ii) A new fuzzy sender bitrate adaptation scheme was presented at the sender side over WLAN and UMTS access networks. (5) Finally, Internet-based subjective tests that captured distortions caused by the encoder and the wireless access network for different types of contents were designed. The database of subjective results has been made available to research community as there is a lack of subjective video quality assessment databases.

Network Utility Maximization (NUM) provides a key conceptual framework to study reward allocation amongst a collection of users/entities in disciplines as diverse as economics, law and engineering. However when the available resources and/or users' utilities vary over time, reward allocations will tend to vary, which in turn may have a detrimental impact on the users' overall satisfaction or quality of experience. In this thesis, we introduce a generalization of the NUM framework which incorporates the detrimental impact of temporal variability in a user's allocated rewards and explicitly incorporates Mean-Variability-Fairness tradeoffs, i.e., tradeoffs amongst the mean and variability in users' reward allocations, as well as fairness across users. We propose a simple online algorithm to realize these tradeoffs, which, under stationary ergodic assumptions, is shown to be asymptotically optimal, i.e., achieves a long term performance equal to that of an offline algorithm with knowledge of the future variability in the system. This substantially extends work on NUM to an interesting class of relevant problems where users/entities are sensitive to temporal variability in their service or allocated rewards. We extend the theoretical framework and tools developed for realizing Mean-Variability-Fairness tradeoffs to develop a simple online algorithm to solve the problem of optimizing video delivery in networks. The tremendous increase in mobile video traffic projected for the future along with insufficiency of available wireless network capacity makes this one of the most important networking problems today. Specifically, we consider a network supporting video clients streaming stored video, and focus on the problem of jointly optimizing network resource allocation and video clients' video quality adaptation. Our objective is to fairly maximize video clients' video Quality of Experience (QoE) realizing Mean-Variability-Fairness tradeoffs, incorporating client preferences on rebuffering time and the cost of video delivery. We present a simple asymptotically optimal online algorithm NOVA (Network Optimization for Video Adaptation) to solve the problem. Our algorithm uses minimal communication, 'distributes' the tasks of network resource allocation to a centralized network controller, and video clients' video quality adaptation to the respective video clients. Further, the quality adaptation is also optimal for standalone video clients, and is an asynchronous algorithm well suited for use in the Dynamic Adaptive Streaming over HTTP (DASH) framework. We also extend NOVA for use with more general video QoE models, and study NOVA accounting for practical considerations like time varying number of video clients, sharing with other types of traffic, performance under legacy resource allocation policies, videos with variable sized segments etc. / text

Made available in DSpace on 2015-04-20T12:33:26Z (GMT). No. of bitstreams: 1 Leandro.pdf: 2831865 bytes, checksum: 5804d85c95f338cf4054c799f4dfd45d (MD5) Previous issue date: 2011-10-07 / Voice calls based on Voice over Internet Protocol (VoIP) technology are liable to several impairments from both application and network layer, such as codec compression, end-to-end delay, and packet loss. For years, this problem has been challenging researchers and practitioners, who have been designing and improving QoS control mechanisms for VoIP applications. Such mechanisms aim to make optimum use of network and terminal resources so as to minimize the effects of network impairments on voice quality. Among the several proposed QoS control mechanisms for VoIP, some of them seek to adapt the voice flow or other VoIP-related parameters in accordance with significant changes in the network, end users preferences, or service providers requirements. VoIP systems are particularly likely to require a dynamic adaptation solution for dealing with the complex trade-off between speech quality and impairments, because of the decentralized control nature of IP networks and the stochastic nature of data packet delivery. Although the existing adaptive solutions for QoS control of VoIP show some performance improvement and exhibit some sort of feedback, they do not provide explicit focus on the control loop. This document shows the current progress of our thesis, which addresses the adjustment of internal parameters of VoIP terminals (at application layer) that affect the voice flow, with the aim of improving speech quality in response to changes in network conditions. It is not in the scope of the thesis to propose adaptive solutions that focus exclusively on signaling, billing, security issues, or operate at the network layer. Therefore, this thesis addresses the problem of how adjust encoding parameters in response to variations in delay and packet loss, in order to optimize speech quality. The objective is to optimize user-perceptible attributes of speech, under the perspective of self-adaptive software systems. The emphasis is not to develop new audio codecs, but to build a control loop in the core of sender and receiver terminals to adapt voice flow settings according to network conditions. The main contributions of this thesis are the following: determination of user s perception during codec switching; parametrization of codec precedence for supporting codec switching decision; explicit design of a monitoring analysis planning execution control loop as the core of the adaptation process; and efficiency analysis of feedback message exchanging. / Chamadas de voz baseadas na tecnologia VoIP (Voice over Internet Protocol) estão suscetíveis a degradações diversas, provenientes tanto da camada de aplicação, como da camada de rede, tais como compressão do codec, atraso fim a fim e perda de pacotes. Durante anos, esse problema tem desafiado pesquisadores e profissionais, que têm concebido e melhorado mecanismos de controle de QoS para aplicações VoIP. Tais mecanismos visam otimizar a utilização dos recursos da rede e do terminal VoIP de modo a minimizar os efeitos deletérios da rede subjacente sobre a qualidade de voz. Entre as várias propostas de mecanismos de controle de QoS para VoIP, alguns deles procuram adaptar o fluxo de voz ou outros parâmetros VoIP de acordo com mudanças significativas na rede, preferências de usuário, ou requisitos dos provedores de serviços VoIP. Sistemas VoIP particularmente exigem soluções de adaptação dinâmica para lidar com a complexa relação de compromisso entre qualidade de voz e fatores de degradação, por causa da natureza descentralizada e estocástica das redes IP na entrega de pacotes de voz. Embora as soluções adaptativas existentes para controle de QoS em VoIP mostrem alguma melhora de desempenho e apresentem algum tipo de feedback, elas não fornecem foco explícito na ciclo de controle (control loop). Este documento mostra o progresso atual da nossa tese, que aborda o ajuste de parâmetros internos de terminais VoIP (camada de aplicação) que afetam o fluxo de voz, com o objetivo de melhorar a qualidade da fala em resposta a mudanças nas condições da rede. Não faz parte do escopo da tese abordar soluções adaptativas que se concentram exclusivamente em sinalização, bilhetagem, problemas de segurança, ou que operam no nível da camada de rede. Portanto, esta tese aborda o problema da concepção e avaliação de estratégias adaptativas que explorem as relações de compromisso entre qualidade da fala e os seguintes fatores de degradação: compressão do codec, atraso fim a fim e perda de pacotes. A finalidade é otimizar atributos da fala perceptíveis aos usuário, sob a perspectiva de sistemas de software autoadaptativo. A ênfase não reside em desenvolver novos codecs de áudio, mas sim em desenvolver um ciclo de controle como entidade central de um terminal VoIP, que possa adaptar as configurações do fluxo de voz de acordo com as condições da rede. As principais contribuições desta tese são as seguintes: determinação da percepção do usuário durante a comutação de codec; parametrização de precedência de codecs para suporte de decisão de comutação de codec; enfoque no ciclo de controle baseado nas atividades de monitoramento análise planejamento execução como núcleo do processo de adaptação; e análise de eficiência de troca de mensagens de feedback.