This thesis presents a stereo coding architecture for the ITU-T G.719 fullband mono codec. G.719 is suitable for teleconferencing applications with a competitive audio quality for speech and audio signals that are encoded at 32, 48 and 64 kbps. The proposed stereo architecture comprises parametric stereo coding where the spatial properties of the stereo channels are modeled with the use of parameters, which are encoded and transmitted to the decoder together with an encoded downmix of the stereo channels. The stereo architecture has been implemented in MATLAB with an external mono coding using a floating point ANSI-C implementation of the ITU-T G.719 codec. Two parametric stereo models have been implemented in a framework operating in the complex-valued Modified Discrete Fourier Transform (MDFT) domain. The first model is based on the inter-channel cues that represent level differences, time differences and coherences between the stereo channels. The cues approximate the corresponding interaural cues that characterize our localization of sound in space. The second model is based on the Karhunen-Loève Transform (KLT) with the associated rotation angles, the inter-channel time differences and the residual scaling parameters. An improved MDFT domain extraction of the inter-channel time difference between the stereo channels has been used for both stereo models. The extracted stereo parameters have been non-uniformly quantized based on the spatial accuracy and the frequency dependency of the human auditory system. The data rate of the stereo parameters has been estimated for each model to around 4 kbps. As a result G.719 has been used as a core codec at 44 and 60 kbps in order to subjectively evaluate the performance of the fullband stereo codec at 48 and 64 kbps. In the comparison with G.719 dual mono coding, i.e. independent mono coding of the stereo channels, the evaluation showed a higher performance of the proposed stereo models for complex clean and reverberant speech signals. However, no consistent gain of the parametric stereo coding was revealed for noisy speech, mixed content and music signals. In addition, the first stereo model showed consistently a slightly higher performance than the second model in the subjective evaluation but with no significant difference. The results revealed a high potential for parametric stereo coding using the ITU-T G.719 codec. In comparison to the existing stereo codecs 3GPP AMR-WB+ and 3GPP eAAC+ the average performance was better at the equal bitrate of 48 kbps.

Made available in DSpace on 2018-08-02T00:00:53Z (GMT). No. of bitstreams: 1 tese_3858_DissertacaoMestradoPedroPauloFavatoBarcelos.pdf: 2238551 bytes, checksum: 7a7dd07c841d1887b77a0bdbed3ff9ac (MD5) Previous issue date: 2011-04-07 / A recomendação ITU-T G.805 (ITU-T, 2000) é uma importante recomendação para redes de transporte, pois descreve uma arquitetura funcional genérica independente de tecnologias para este domínio e é usada como base para outras recomendações que descrevem a arquitetura funcional de redes, a gerência, a avaliação de desempenho e a especificação funcional de equipamentos. Apesar de fornecer uma ferramenta ágil para a descrição da arquitetura, a apresentação dos conceitos é feita de forma textual, gerando confusão por conta de definições recursivas e exemplos não claros, que muitas vezes até mesmo se contradizem. Esses aspectos da recomendação a torna de difícil entendimento, podendo confundir o leitor. É importante que, devido sua fundamental relevância, essa recomendação seja livre desses problemas. Para tal, é proposta nesta dissertação a utilização de técnicas de modelagem conceitual baseadas em ontologias para a geração de um modelo de referência para a área de redes de transporte, a partir da Recomendação ITU-T G.805. Além dos principais conceitos da recomendação são também apresentadas as vantagens da criação de um modelo de referência em ontologias e as principais tecnologias utilizadas para este objetivo. São realizadas uma análise arquitetural e uma reestruturação dos componentes definidos pela recomendação e uma avaliação ontológica da mesma, verificando casos de incompletudes, ambiguidades e outras deficiências ontológicas e apontando soluções. Por fim, é apresentado o modelo de referência em ontologia desenvolvido para a Recomendação ITU-T G.805, incluindo o modelo conceitual e suas regras de derivação e de restrição.

Made available in DSpace on 2016-12-23T14:07:26Z (GMT). No. of bitstreams: 1 Pedro Paulo Favato Barcelos Cap 1 a 5.pdf: 1996678 bytes, checksum: 06570d7feadc8f768039f34ccc71400a (MD5) Previous issue date: 2011-04-07 / A recomendação ITU-T G.805 (ITU-T, 2000) é uma importante recomendação para redes de transporte, pois descreve uma arquitetura funcional genérica independente de tecnologias para este domínio e é usada como base para outras recomendações que descrevem a arquitetura funcional de redes, a gerência, a avaliação de desempenho e a especificação funcional de equipamentos. Apesar de fornecer uma ferramenta ágil para a descrição da arquitetura, a apresentação dos conceitos é feita de forma textual, gerando confusão por conta de definições recursivas e exemplos não claros, que muitas vezes até mesmo se contradizem. Esses aspectos da recomendação a torna de difícil entendimento, podendo confundir o leitor. É importante que, devido sua fundamental relevância, essa recomendação seja livre desses problemas. Para tal, é proposta nesta dissertação a utilização de técnicas de modelagem conceitual baseadas em ontologias para a geração de um modelo de referência para a área de redes de transporte, a partir da Recomendação ITU-T G.805. Além dos principais conceitos da recomendação são também apresentadas as vantagens da criação de um modelo de referência em ontologias e as principais tecnologias utilizadas para este objetivo. São realizadas uma análise arquitetural e uma reestruturação dos componentes definidos pela recomendação e uma avaliação ontológica da mesma, verificando casos de incompletudes, ambiguidades e outras deficiências ontológicas e apontando soluções. Por fim, é apresentado o modelo de referência em ontologia desenvolvido para a Recomendação ITU-T G.805, incluindo o modelo conceitual e suas regras de derivação e de restrição / The ITU-T Recommendation G.805 (ITU-T, 2000) is an important recommendation for transport networks. It describes a generic functional architecture that is independent of technology for this domain and it is used as the basis for recommendations that describe the functional architecture of networks, management, performance analysis and functional specification of equipment. Despite providing a flexible tool for the architecture description, the recommendation presents its concepts textually, leading to confusion because of recursive definitions and unclear examples that are often contradictory. These aspects of the recommendation make it difficult to understand and may confuse the reader. It is important that, due to its fundamental importance, this recommendation is free from these problems. For this purpose, this work proposes the use of ontology-based conceptual modeling techniques for the generation of a reference model for the transport network domain, based on the ITU-T Recommendation G.805. In addition to the recommendation main concepts, the advantages of creating an ontology-based reference model and the main technologies used for this purpose are also presented. An architectural analysis and a restructuring of the components defined by the recommendation are performed together with an ontological evaluation of it. Cases of incompleteness, ambiguities and other deficiencies are checked and solutions are pointed. Finally, the ontology-based reference model developed for the ITU-T Recommendation G.805 is presented, including the conceptual model and its derivation and restrictions rules

Made available in DSpace on 2018-08-02T00:01:06Z (GMT). No. of bitstreams: 1 tese_6875_Dissertação_Rafael_Campanharo_Favoreto_Versão_Final.pdf: 6258201 bytes, checksum: bed370fa569f7f0c60e71f03e2c28a5e (MD5) Previous issue date: 2014-08-22 / As demandas por serviços de telecomunicações multimídia, garantia de QoS (Quality of Service) e mecanismos de gerenciamento e controle direcionam a evolução da rede de núcleo para adoção da tecnologia OTN (Optical Transport Network) como solução de rede de transporte. Para conduzir a evolução da tecnologia OTN a ITU-T (International Telecommunication Union Telecommunications Standardization Sector) estabelece uma série de recomendações, dentre elas, as específicas para o plano de gerenciamento de redes. Essas, por sua vez, apresentam deficiências no que se referem ao levantamento do inventário dos objetos gerenciados definidos pelas normas da ITU-T. A ITU-T define a representação dos elementos de rede em caráter funcional, mais especificamente como módulos de Funções Atômicas. Contudo, a falta de clareza na implementação das Funções Atômicas e a carência de integração com as representações físicas dos elementos de rede implica na omissão, por parte dos diversos fabricantes, da implementação dos mecanismos recomendados em norma. Esta dissertação tem como objetivo geral apresentar uma proposta de solução para o levantamento do inventário dos elementos de rede em redes OTN abrangendo de forma integrada as representações funcional e física do elemento de rede, além de possibilitar aos diversos fabricantes a aderência às normas da ITU-T bem como oferecer ao operador de rede a configuração dos componentes de uma forma mais intuitiva.

Continuous Evaluation of Time Varying Speech Quality (CETVSQ) is a method of subjective assessment of transmitted speech quality for long speech sequences containing quality fluctuations in time. This method is modeled for continuous evaluation of long speech sequences based on two subjective tasks. First task is to assess the speech quality during the listening and second task is to assess the overall speech quality after listening to the speech sequences. The development of continuous evaluation of time varying speech quality was motivated by fact that speech quality degradations are often not constant and varies in time. In modern IP telephony and wireless networks, speech quality varies due to specific impairments such as packet loss, echo, handover in networks etc. Many other standard methods already exist, which are being used for subjective assessment of short speech sequences. These methods such as ITU-T Rec. P.800 are well suited for only time constant speech quality. In this thesis work, it was required to implement CETVSQ methodology, so that it could be possible to assess long speech sequences. An analog hardware slider is used for the continuous assessment of speech qualities, as well as for overall quality judgments. Instantaneous and overall quality judgments are being saved into Excel file. The results stored in the Excel file are analyzed by applying different statistical measures. In evaluation part of the thesis work, subjects’ scores are analyzed by applying statistical methods to identify several factors that have originated in the CETVSQ methodology. A subjective test had already been conducted according to P.800 ACR method. The long speech sequences were divided into 8 seconds short sequences and then assessed using P.800 ACR method. In this study, the long speech sequences are assessed using CETVSQ methodology and comparison is conducted between P.800 ACR and CETVSQ results. It has been revealed that if long speech sequences are divided into short segments and evaluated using P.800 ACR, then P.800 ACR results will be different from the results obtained from CETVSQ methodology. The necessity of CETVSQ methodology is proved by this study.

Current telecommunication infrastructure is undergoing significant changes. Such changes involve the type of traffic traveling through the network as well as the requirements imposed by the new traffic mix (e.g. strict delay control and low end-to-end delay). In this new networking scenario, the current infrastructure, which remained almost unchanged for the last several decades, is struggling to adapt, and its limitations in terms of power consumption, scalability, and economical viability have become more evident. In this dissertation we explore the potential advantages of using periodic data structures to handle efficiently bandwidth-intensive transactions, which constitute a significant portion of today's network traffic. We start by implementing an approach that can work as a standalone system aiming to provide the same advantages promised by all-optical approaches such as OBS and OFS. We show that our approach is able to provide similar advantages (e.g. energy efficiency, link utilization, and low computational load for the network hardware) while avoiding the drawbacks (e.g. use of optical buffers, inefficient resource utilization, and costly deployment), using commercially available hardware. Aware of the issues of large scale hardware redeployment, we adapt our approach to work within the current transport network architecture, reusing most of the hardware and protocols that are already in place, offering a more gradual evolutionary path, while retaining the advantages of our standalone system. We then apply our approach to Data Center Networks (DCNs), showing its ability to achieve significant improvements in terms of network performance stability, predictability, performance isolation, agility, and goodput with respect to popular DCN approaches. We also show our approach is able to work in concert with many proposed and deployed DCN architectures, providing DCNs with a simple, efficient, and versatile protocol to handle bandwidth-intensive applications within the DCs. / Graduate

Issue of this diploma thesis is focused on analysis of audio codecs used within IP telephony. Attention of teoretical part is given mostly to audio codecs according to ITU-T recommendations, but also to signaling protocols used here. For practical part of analysis is chosen router Cisco 2821 and IP phones Cisco 7975G. Configuration is done over operating system Cisco IOS. Chosen signaling protocol is SCCP. For analysis itself are chosen 2 analysers – L-580FX and Fluke NetTool. These are used in combination with program Wireshark. Analysed parameters are latency, packet lost, bandwidth, jitter and mean opinion score. Measured values are presented in graphs and tables and they are discussed. Next output of the thesis is laboratory excercise, which deals with analysis of audio codecs.

Orientador: Gelio Mendes Ferreira / Dissertação (mestrado) - Universidade Federal do ABC, Programa de Pós-Graduação em Engenharia da Informação, 2011

Progressive download video services, such as YouTube and podcasts, are responsible for a major part of the transmitted data volume in the Internet and it is expected, that they will also strongly affect mobile networks. Streaming video quality mainly depends on the sustainable throughput achieved during transmission. To ensure acceptable video quality in mobile networks (with limited capacity resources) the perceived quality by the customer (QoE) needs to be monitored by estimation. For that, the streaming video quality needs to be measured and monitored permanently. For TCP based progressive download we propose to extract the the video timestamps which are encoded within the payload of the TCP segments by decoding the video within the payload. The actual estimation is then done by play out buffer fill level calculations based on the TCP segment timestamp and their internal play out timestamp. The perceived quality for the user is derived from the number and duration of video stalls. Algorithms for decoding Flash Video, MP4 and WebM Video have already been implemented. After deriving the play out time it is compared to the timestamp of the respective TCP segment. The result of this comparison is an estimate of the fill level of the play out buffer in terms of play out time within the client. This estimation is done without access to the end device. The same measurement procedure can be applied for any TCP based progressive download Internet service. Video was simply taken as an example because of its current large share in traffic volume in operator networks.

Progressive download video services, such as YouTube, are responsible for a major part of the transmitted data volume in the Internet and it is expected, that they also will strongly affect mobile networks. Streaming video quality mainly depends on the sustainable throughput achieved during transmission. In order to achieve an acceptable video quality in mobile networks (with limited capacity resources), traffic engineering mechanisms have to be applied. For that, the streaming video quality needs to be measured and monitored permanently. Therefore, the video timestamps which are encoded within the payload of the TCP segments have to be extracted. For that it is necessary to decode the video within the transported payload. Algorithms for decoding Flash Video, MP4 and WebM Video have already been implemented as a demonstration implementation in support of the network based measurement contribution to SG12 by Chemnitz University for TCP encoded progressive download Internet services. In the demonstration, the derived play out buffering from the monitored traffic is being output internally. A second application is then used to graphically display the estimation result. The measurement and estimation is solely done within a measurement point of an operator network without access to the client’s end device.