QoS Evaluation of BandwidthSchedulers in IPTV Networks OfferedSRD Fluid Video Traffic

Internet protocol TV (IPTV) is predicted to be the key technology winner in the future. Efforts to accelerate the deployment of IPTV centralized model which is combined of VHO, encoders, controller, access network and Home network. Regardless of whether the network is delivering live TV, VOD, or Time-shift TV, all content and network traffic resulting from subscriber requests must traverse the entire network from the super-headend all the way to each subscriber's Set-Top Box (STB).IPTV services require very stringent QoS guarantees When IPTV traffic shares the network resources with other traffic like data and voice, how to ensure their QoS and efficiently utilize the network resources is a key and challenging issue. For QoS measured in the network-centric terms of delay jitter, packet losses and bounds on delay. The main focus of this thesis is on the optimized bandwidth allocation and smooth datatransmission. The proposed traffic model for smooth delivering video service IPTV network with its QoS performance evaluation. According to Maglaris et al [5] First, analyze the coding bit rate of a single video source. Various statistical quantities are derived from bit rate data collected with a conditional replenishment inter frame coding scheme. Two correlated Markov process models (one in discrete time and one incontinuous time) are shown to fit the experimental data and are used to model the input rates of several independent sources into a statistical multiplexer. Preventive control mechanism which is to be include CAC, traffic policing used for traffic control.QoS has been evaluated of common bandwidth scheduler( FIFO) by use fluid models with Markovian queuing method and analysis the result by using simulator andanalytically, Which is measured the performance of the packet loss, overflow and mean waiting time among the network users.

QoS evaluation of bandwidth schedulersin IPTV networks offered SRD fluidvideo traffic: a simulation study

IPTV is now offered by several operators in Europe, US and Asia using broadcast video over private IP networks that are isolated from Internet. IPTV services rely ontransmission of live (real-time) video and/or stored video. Video on Demand (VoD)and Time-shifted TV are implemented by IP unicast and Broadcast TV (BTV) and Near video on demand are implemented by IP multicast. IPTV services require QoS guarantees and can tolerate no more than 10-6 packet loss probability, 200 ms delay, and 50 ms jitter. Low delay is essential for satisfactory trick mode performance(pause, resume,fast forward) for VoD, and fast channel change time for BTV. Internet Traffic Engineering (TE) is defined in RFC 3272 and involves both capacity management and traffic management. Capacity management includes capacityplanning, routing control, and resource management. Traffic management includes (1)nodal traffic control functions such as traffic conditioning, queue management, scheduling, and (2) other functions that regulate traffic flow through the network orthat arbitrate access to network resources. An IPTV network architecture includes multiple networks (core network, metronetwork, access network and home network) that connects devices (super head-end, video hub office, video serving office, home gateway, set-top box). Each IP router in the core and metro networks implements some queueing and packet scheduling mechanism at the output link controller. Popular schedulers in IP networks include Priority Queueing (PQ), Class-Based Weighted Fair Queueing (CBWFQ), and Low Latency Queueing (LLQ) which combines PQ and CBWFQ.The thesis analyzes several Packet Scheduling algorithms that can optimize the tradeoff between system capacity and end user performance for the traffic classes. Before in the simulator FIFO,PQ,GPS queueing methods were implemented inside. This thesis aims to implement the LLQ scheduler inside the simulator and to evaluate the performance of these packet schedulers. The simulator is provided by ErnstNordström and Simulator was built in Visual C++ 2008 environmentand tested and analyzed in MatLab 7.0 under windows VISTA.

Speech Assessment for the Classification of Hypokinetic Dysthria in Parkinson Disease

The aim of this thesis is to investigate computerized voice assessment methods to classify between the normal and Dysarthric speech signals. In this proposed system, computerized assessment methods equipped with signal processing and artificial intelligence techniques have been introduced. The sentences used for the measurement of inter-stress intervals (ISI) were read by each subject. These sentences were computed for comparisons between normal and impaired voice. Band pass filter has been used for the preprocessing of speech samples. Speech segmentation is performed using signal energy and spectral centroid to separate voiced and unvoiced areas in speech signal. Acoustic features are extracted from the LPC model and speech segments from each audio signal to find the anomalies. The speech features which have been assessed for classification are Energy Entropy, Zero crossing rate (ZCR), Spectral-Centroid, Mean Fundamental-Frequency (Meanf0), Jitter (RAP), Jitter (PPQ), and Shimmer (APQ). Naïve Bayes (NB) has been used for speech classification. For speech test-1 and test-2, 72% and 80% accuracies of classification between healthy and impaired speech samples have been achieved respectively using the NB. For speech test-3, 64% correct classification is achieved using the NB. The results direct the possibility of speech impairment classification in PD patients based on the clinical rating scale.

QoS evaluation of Bandwidth Schedulers in IPTV Networks Offered SRD Fluid Video Traffic

Internet protocol TV (IPTV) is predicted to be the key technology winner in the future. Efforts to accelerate the deployment of IPTV centralized model which is combined of VHO, encoders, controller, access network and Home network. Regardless of whether the network is delivering live TV, VOD, or Time-shift TV, all content and network traffic resulting from subscriber requests must traverse the entire network from the super-headend all the way to each subscriber's Set-Top Box (STB). IPTV services require very stringent QoS guarantees When IPTV traffic shares the network resources with other traffic like data and voice, how to ensure their QoS and efficiently utilize the network resources is a key and challenging issue. For QoS measured in the network-centric terms of delay jitter, packet losses and bounds on delay. The main focus of this thesis is on the optimized bandwidth allocation and smooth data transmission. The proposed traffic model for smooth delivering video service IPTV network with its QoS performance evaluation. According to Maglaris et al [5] first, analyze the coding bit rate of a single video source. Various statistical quantities are derived from bit rate data collected with a conditional replenishment inter frame coding scheme. Two correlated Markov process models (one in discrete time and one in continuous time) are shown to fit the experimental data and are used to model the input rates of several independent sources into a statistical multiplexer. Preventive control mechanism which is to be including CAC, traffic policing used for traffic control. QoS has been evaluated of common bandwidth scheduler( FIFO) by use fluid models with Markovian queuing method and analysis the result by using simulator and analytically, Which is measured the performance of the packet loss, overflow and mean waiting time among the network users.

Um modelo de videoconferência para computador pessoal orientado ao perfil de aplicação

A informática vem adquirindo papéis cada vez mais importantes na vida cotidiana. Um dos papéis mais significativos, hoje, é o suporte a comunicações; atualmente, é muito difícil pensar em comunicações – mesmo interpessoais – sem fazer associação às áreas de informática e redes. Dentre as aplicações que utilizam informática e redes como suporte, a tecnologia de videoconferência tem recebido papel de destaque. Os avanços na tecnologia de redes e conectividade, aliados à padronização e à crescente oferta de produtos de videoconferência, têm aumentado a aplicabilidade e a popularidade destes produtos, sobretudo utilizados sobre arquitetura de redes TCP/IP. Trata-se de uma tecnologia atraente em termos de resultado, por agregar, além do áudio – recurso comum há muito tempo como suporte à comunicação – os recursos de vídeo e aplicações integradas (como quadro-branco compartilhado, Chat, troca de arquivos e outros). Contudo, essas aplicações são bastante exigentes, tanto em termos de banda quanto de qualidade de serviço (QoS) da rede. O primeiro item se justifica pelo volume de dados gerados pelas aplicações de videoconferência; o segundo, pela significativa influência que os problemas de qualidade da infraestrutura de rede (como elevada latência, jitter e descartes) podem exercer sobre tais aplicações. A busca para as soluções destes problemas não é tarefa simples, pois muitas vezes envolve investimentos que desencorajam a adoção da tecnologia de videoconferência – principalmente para uso pessoal ou por empresas pequenas. Este trabalho propõe uma solução aos problemas mencionados, visando proporcionar uma melhor aceitação e maior disseminação da tecnologia de videoconferência, valendo-se de recursos com pouca demanda de investimento. A estratégia abordada é a adaptação de tráfego, com um enfoque diferenciado: o de levar em conta, para cada aplicação, o comportamento que o processo de adaptação apresentasse. A partir dessa orientação, é proposto um modelo de adaptação de tráfego orientado ao perfil da aplicação, voltado ao interesse do usuário, e que disponibilize uma forma ao mesmo tempo simples e eficiente para que o usuário realize a adequação do mecanismo de adaptação do sistema às suas necessidades e expectativas. A partir desta proposta, foi implementado um protótipo de aplicação, com o objetivo de verificar a funcionalidade do modelo em termos práticos. As observações dos resultados dos testes, bem como as conclusões geradas, serviram como validação da proposta.

Análise do atraso fim-a-fim observado por diferentes classes de tráfego em redes Ethernet micro-segmentadas

A tecnologia de rede dominante em ambientes locais é ethernet. Sua ampla utilização e o aumento crescente nos requisitos impostos pelas aplicações são as principais razões para a evolução constante presenciada desde o surgimento deste tipo de rede simples e barata. Na busca por maior desempenho, configurações baseadas na disputa por um meio de transmissão compartilhado foram substituídas por topologias organizadas ao redor de switches e micro-segmentadas, ou seja, com canais de transmissão individuais para cada estação. Neste mesmo sentido, destacam-se os dispositivos de qualidade de serviço padronizados a partir de 1998, que permitem a diferenciação de tráfego prioritário. O uso desta tecnologia em ambientes de automação sempre foi refreado devido à imprevisibilidade do protocolo CSMA/CD. Entretanto, o uso de switched ethernet em conjunto com a priorização de tráfego representa um cenário bastante promissor para estas aplicações, pois ataca as duas fontes principais de indeterminismo: colisões e enfileiramento. Este trabalho procura avaliar esta estrutura em ambientes de controle distribuído, através de uma análise temporal determinística. Como resultado, propõe-se um modelo analítico de cálculo capaz de prever os valores máximos, médios e mínimos do atraso fim-a-fim observado nas comunicações de diferentes classes de tráfego. Durante a análise, outras métricas relevantes são investigadas como a variação no atraso (jitter) e a interferência entre classes de prioridades distintas.

TCP HolyWood

We introduce a new end-to-end, sender side Transport Control Protocol called TCP HolyWood or in short TCP-HW. In a simulated wired environment, TCP HolyWood outperforms in average throughput, three of the more important TCP protocols ever made, we are talking about TCP Reno, TCP Westwood, and TCP Vegas; and in average jitter to TCP Reno and TCP Vegas too. In addition, according to Jain’s index, our proposal is as fair as TCP Reno, the Standard.

Uma extensão do protocolo CAN para aplicações críticas em sistemas distribuídos

Sistemas computacionais de tempo-real são tipicamente construídos a partir de primitivas de sincronização que fornecem uma noção do tempo no objetivo de coordenar a execução múltiplos fluxos de instruções em um processador. Quando o processamento é centralizado, a base de tempo destas primitivas é extraída do oscilador local da plataforma, permitindo que as ações do sistema sejam devidamente ordenadas, respeitando restrições de tempo e causalidade. No entanto, em sistemas distribuídos o problema não pode ser resolvido desta forma em decorrência de imperfeições nos dispositivos físicos. Diferenças mínimas na freqüência de osciladores fazem com que as bases de tempo dos componentes divirjam cada vez mais ao longo do tempo, dificultando ou até mesmo impossibilitando um ordenamento consistente de eventos. Por esta razão, sincronização de relógios é um serviço de fundamental importância, sobretudo em aplicações críticas, onde os níveis de confiabilidade exigidos são mais elevados. O presente trabalho consiste na proposta e implementação de uma plataforma de comunicação otimizada para sistemas de controle distribuídos, caracterizados por uma alta regularidade no comportamento da comunicação. O objetivo é propor uma solução em baixo nível com suporte para o projeto de sistemas distribuídos no domínio de aplicações críticas. A plataforma proposta, à qual foi atribuído o nome CASCA, sigla para “Communication Architecture for Safety- Critical Applications”, é de fato uma extensão time-triggered do protocolo CAN. Acima da camada de enlace do protocolo original foram projetados mecanismos sincronização de relógios e criação inicial da base de tempo, implementados na forma de uma combinação de hardware e software. Principais características da plataforma são jitter mínimo, uma base de tempo global essencialmente distribuída e particionamento temporal. Diferentes alternativas de projeto foram consideradas, observando com maior atenção a viabilidade de prototipação em dispositivos FPGA para fins de validação e aplicação imediata em plataformas reconfiguráveis. Como forma de validação da plataforma, um sistema elementar formado por três nodos foi sintetizado com sucesso em bancada obtendo-se como resultado uma base de tempo essencialmente distribuída com precisão menor do que um micro-segundo.

Estudo da qualidade de serviço de uma aplicação VoIP em ambientes wireless com handoff

This work deals with experimental studies about VoIP conections into WiFi 802.11b networks with handoff. Indoor and outdoor network experiments are realised to take measurements for the QoS parameters delay, throughput, jitter and packt loss. The performance parameters are obtained through the use of software tools Ekiga, Iperf and Wimanager that assure, respectvely, VoIP conection simulation, trafic network generator and metric parameters acquisition for, throughput, jitter and packt loss. The avarage delay is obtained from the measured throughput and the concept of packt virtual transmition time. The experimental data are validated based on de QoS level for each metric parameter accepted as adequated by the specialized literature

Dysphonia and laryngeal sequelae in paracoccidioidomycosis patients: a morphological and phoniatric study

The larynx is the third most commonly involved organ in paracoccidioidomycosis (PCM). While a few studies have evaluated laryngeal sequelae, there have not been any investigations of voice abnormalities in PCM patients. To evaluate persistent dysphonia and laryngeal lesions, we studied 15 normal subjects and 30 post-treatment PCM patients, i.e., 15 with only pulmonary and 15 with both laryngeal and pulmonary involvement. Perceptual and acoustic voice analysis were performed with all patients, while endoscopic studies were also conducted with the 15 laryngeal patients. Voice analysis showed instability by perceptual analysis (P < 0.01) in both groups, but more severe dysphonia was noted in the laryngeal group (P < 0.01). The dysponia, seen in 66.7% of these patients (dysphonia index < 7.0), was characterized by roughness and breathness. The Dr. Speech (Tiger Electronics) analysis program did not accept five voices from the laryngeal group due to the severe dysphonia. Jitter was elevated in five laryngeal lesion patients. Endoscopy showed that 80% of patients with laryngeal lesion had two or more laryngeal structures involved. Vocal fold alterations were seen in all laryngeal lesion patients, which included involvement of the arythenoids, epiglottis, and vestibular folds. This first functional study of laryngeal sequelae in PCM revealed frequent and severe dysphonia that may have important social consequences for patients.

A low-voltage low-sensitivity sinusoidal VCO for DPLL realizations

A quasi-sinusoidal linearly tunable OTA-C VCO built with triode-region transconductors is presented. Oscillation upon power-on is ensured by RHP poles associated with gate-drain capacitances of OTA input devices. Since the OTA nonlinearity stabilizes the amplitude, the oscillation frequency f0 is first-order independent of VDD, making the VCO adequate to mixed-mode designs. A range of simulations attests the theoretical analysis. As part of a DPLL, the VCO was prototyped on a 0.8μm CMOS process, occupying an area of 0.15mm2. Nominal f0 is 1MHz, with K VCo=8.4KHz/mV. Measured sensitivity to VDD is below 2.17, while phase noise is -86dBc at 100-KHz offset. The feasibility of the VCO for higher frequencies is verified by a redesign based on a 0.35μm CMOS process and VDD=3.3V, with a linear frequency-span of l3.2MHz - 61.5MHz.

Autoregressive decomposition and pole tracking applied to vocal fold nodule signals

This letter describes a novel algorithm that is based on autoregressive decomposition and pole tracking used to recognize two patterns of speech data: normal voice and disphonic voice caused by nodules. The presented method relates the poles and the peaks of the signal spectrum which represent the periodic components of the voice. The results show that the perturbation contained in the signal is clearly depicted by pole's positions. Their variability is related to jitter and shimmer. The pole dispersion for pathological voices is about 20% higher than for normal voices, therefore, the proposed approach is a more trustworthy measure than the classical ones. © 2007.

Using serial bluetooth converters as a sensor link in networked control systems

Networked control systems (NCS) are distributed control system in which sensors, actuators and controllers are physically separated and connected through communication networks. NCS represent the evolution of networked control architectures providing greater modularity and control decentralization, ease maintenance and diagnosis and lower cost of implementation. A recent trend in this research topic is the development of NCS using wireless networks which enable interoperability between existing wired and wireless systems. This paper presents the feasibility analysis of using a serial RS-232 to Bluetooth converter as a wireless sensor link in NCS. In order to support this investigation, relevant performance metrics for wireless control applications such as jitter, time delay and messages lost are highlighted and calculated to evaluate the converter capabilities. In addition the control performance of an implemented motor control system using the converter is analyzed. Experimental results led to the conclusion that serial RS-232 Bluetooth converters can be used to implement wireless networked control systems (WNCS) providing transmission rates and closed control loop times which are acceptable for NCS applications. © 2011 IEEE.

Evaluating serial ZigBee devices for application in wireless networked control systems

Networked control systems (NCS) are distributed control system where the sensors, actuators and controllers are physically separated and connected through communication networks. NCS represent the evolution of networked control architectures providing greater modularity and control decentralization, ease maintenance and diagnosis and lower cost of implementation. A recent trend in this research topic is the development of NCS using wireless networks (WNCS) enabling interoperability between existing wired and wireless systems. This paper evaluates a serial RS-232 ZigBee device as a wireless sensor link in NCS. In order to support this investigation, relevant performance metrics for wireless control applications such as jitter, time delay and messages lost are highlighted and calculated to evaluate the device capabilities. In addition the control performance of an implemented motor control system using the device is analyzed. Experimental results led to the conclusion that serial RS-232 ZigBee devices can be used to implement WNCS and the use of this device delay information in the PID controller discretization can improve the control performance of the system. © 2012 IEEE.

Análise perceptivo-auditiva e acústica da voz em crianças de 4 a 12 anos com obstrução nasal crônica

Coordenação de Aperfeiçoamento de Pessoal de Nível Superior (CAPES)