Over the last decade, the internet industry has rapidly grown with regard to infrastructure and bandwidth. Widespread internet networks with large bandwidth connect people-to-people, people-to-machines, and machine-to-machine. Like other multimedia services, large bandwidth enables voice services to be provided over IP networks where network connectivity is not consistent. In this context, research on service quality monitoring is necessary to satisfy customers by providing consistent service quality. The major contribution of this dissertation is the development of three novel techniques to improve or measure voice quality over IP networks. This dissertation first addresses an adaptive playout buffer scheduling algorithm that enables systems to lossen delay jitter due to the legacy of packet-switched networks. The scheduling algorithm is operated by a desired quality of service, minimizing the end-to-end delay by adjusting playout delay times. Secondly, this dissertation also explores a parameter-based nonintrusive speech quality measure to monitor the quality of VoIP. During the lifetime of sound, the network parameters are estimated and used to predict the quality of speech. As a cognitive model, a machine-learning technique is exploited to map features in the feature space into the perceived speech quality scale space. Finally, this dissertation introduces a signal-based nonintrusive speech quality measure. Features for the proposed measurement are extracted from observations of the characteristics of natural speech sounds and artificial noises. The calculated features are mapped into the perceived speech quality scale. The proposed parameter-based measure achieves a high prediction accuracy while the signal-based measure reaches to a comparable performance to the official International Telecommunication Union (ITU) standard, P.563. The contributions described in this dissertation provides smart methodologies for monitoring or enhancement of VoIP service qualities. / text

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.

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This report describes a bachelor thesis in computer science performed at the system department within the product development unit for GSM (Global System for Mobile communication) at Ericsson in Linköping.GSM can be considered a mature technology but it is still updated with new features on a regular basis. This requires that the technology also continuously is being tested. One technique that is used to perform quality tests is called PESQ (Perceptual Evaluation of Speech Quality). PESQ is a standardized algorithm to perform speech quality tests; it gives a score based on how a human would percept speech quality.The purpose of this thesis was to enhance the speech quality analysis process used when testing the GSM network. One goal was to analyze if it was possible to record and inject sound into a phone call and verify the possibility to execute the PESQ algorithm using a smartphone. Another goal was to develop a smartphone application to perform the described tasks.In order to do so an application for Android was developed. The development process included an investigation whether the Android platform supplies the functionality required. The investiga-tion showed that no general solution for recording and injecting audio into a phone call could be found. Recording was possible on some smartphones, but the characteristics of the recorded audio were not as desired. No solution for injecting audio into a phone call could be found for any of the tested smartphones.However, even though no solution for recording and injecting was found, an application was developed. The application was decided to be more of a proof of concept on how an application of this kind would work. It would also be a good foundation for future development. Suggested future work in order to integrate the application into the existing testing environment is to find alternative solutions for recording and injecting, maybe with the help of smartphone manu-facturers.

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As the characteristics of mobile radio channels vary over time, transmit power must be controlled accordingly to ensure that the received signal level is within the receiver's sensitivity. As a consequence, modern mobile radio systems employ power control to regulate the received signal level such that it is neither less nor excessively larger than receiver sensitivity in order to maintain adequate service quality. In this context, speech quality measurement is an important aspect in the delivery of speech services as it will impact satisfaction of customers as well as the usage of precious system resources. A variety of techniques for speech quality measurement has been produced over the last few years as result of tireless research in the area of perceptual speech quality estimation. These are mainly based on psychoacoustic models of the human auditory systems. However, these techniques cannot be directly applied for real-time communication purposes as they typically require a copy of the transmitted and received speech signals for their operation. This thesis presents a novel technique of incorporating perceptual speech quality metrics with power control for mobile radio systems. The technique allows for standardized perceptual speech quality measurement algorithms to be used for in-service measurement of speech quality. The accuracy of the proposed Real-Time Perceptual Speech Quality Measurement (RTPSQM) technique with respect to measuring speech quality is first validated by extensive simulations. On this basis, RTPSQM is applied to power control in the Global System for Mobile (GSM) communication and the Universal Mobile Telecommunication System (UMTS). It is shown by simulations that the use of perceptual-based power control in GSM and UMTS outperforms conventional power control in terms of reducing the transmitter signal power required for providing adequate speech quality. This in turn facilitates the observed increase in system capacity and thus offers better utilization of available system resources. To enable an analytical performance assessment of perceptual speech quality metrics in power control, the mathematical frameworks for conventional and perceptual-based power control are derived. The derivations are performed for Code Division Multiple Access (CDMA) systems and kept as generic as possible. Numerical results are presented which could be used in a system design to readily find the Erlang capacity per cell for either of the considered power control algorithms.

Vivemos atualmente uma era de convergência de tecnologias, motivada por questões tanto econômicas como de caráter operacional, na qual os serviços de dados, voz e vídeo estão migrando rapidamente para uma plataforma IP. Particularmente considerando o paradigma da telefonia IP neste processo de convergência, ocorrem desafios tecnológicos, pois temos de um lado os usuários finais que já possuem uma referência sobre a qualidade da fala, fruto das décadas de uso do sistema telefônico tradicional, e na outra extremidade as operadoras de telecomunicações que em sua última milha dependem de redes estatísticas, sem mecanismos adequados para a garantia de QoS. Assim, se torna vital para o sucesso da operação a devida identificação das relações entre os diversos componentes existentes entre terminais e sua contribuição para a qualidade de fala, percebida pelo assinante, de forma a entregar um serviço com qualidade similar ao Sistema Telefônico Fixo Comutado. Neste contexto, este trabalho busca identificar por meio de técnicas de estatística multivariada uma correlação entre métricas objetivas de Qualidade de Serviços aplicáveis em redes IP e a qualidade subjetiva da fala predita pelo algoritmo Modelo-E definido na recomendação ITU-T G.107. Um método de coleta e análise estatística de informações foi desenvolvido para atingir o objetivo proposto. Para sua validação um ambiente de testes foi criado, dados de operação foram coletados e ferramentas computacionais foram aplicadas para o tratamento analítico e estatístico. Os resultados obtidos pelo método foram então aplicados em campo durante as etapas de testes e homologação de um PABX-IP-IMS desenvolvido para o mercado corporativo. A correlação entre os diversos fatores envolvidos, suas métricas e como todo este sistema impacta na qualidade relativa, percebida pelo usuário final permitirá aos provedores de serviços avaliarem quais as melhores estratégias a serem empregadas em seus segmentos de rede de forma a garantir a excelência no nível de serviço oferecido ao consumidor final. / We live now in an convergence of technologies era, driven by economic and operational issues, where the data services, voice and video are quickly moving to an IP platform. Particularly considering the paradigm of IP telephony in the process of convergence, there are technological challenges. We have subscribers who already have a reference about the quality of speech, derived from decades of using the traditional phone system. At the other end telecom operators that rely on statistical networks, with no possibility to guarantee QoS. So it becomes vital to the operations success the proper identification of the relationships between the various components between the terminals and their contribution to the speech quality perceived by the subscriber in order to deliver a quality service close to the PSTN. In this context, this study sought to identify a correlation between objective metrics for Quality of Service applicable to IP networks and subjective quality of speech predicted by the algorithm \"Model-E\" defined in ITU-T G.107 through multivariate statistical techniques. A method of collecting and analyzing statistical information was developed to achieve the proposed objective. To validate a test environment was created, operation data were collected and computational tools were applied to the analytical and statistical treatment. The results obtained by the method were then applied in the field during the stages of testing and approval of an IMS-IP-PBX designed for the corporate market. The correlation between the various factors involved, their metrics and how the whole system impacts on the quality perceived by end users will enable service providers to assess what the best strategies to use in their network segments to ensure an adequate level of service offered to consumers.

Hearing aids are tasked with the undesirable job of compensating an impaired, highly-nonlinear auditory system. Historically, these devices have either employed linear processing or relatively unsophisticated, nonlinear processing techniques. With increasingly more accurate models of the auditory system, expanding computational power, and many more objective measures which utilize these models, we are at a turning point in hearing aid design. Although subjective listener tests are often the most accepted methods for evaluating the quality and intelligibility of speech, they inherently treat the auditory system as a "black box." Conversely, model-based objective measures typically treat the auditory system as a cascade of physical processes. As a result, objective measures have the potential to provide more detailed information about how sound is processed and about where and why quality or intelligibility breaks down. Provided that we can generalize model-based objective measures, we can use the measures as tools for understanding how to best process degraded signals, and therefore, how to best design hearing aids. However, generalizability is a key requirement. Since many of the well-known objective measures have been developed for normal-hearing listeners in the context of audio codecs, we are unsure about the generalizability of these measures to predicting quality and intelligibility for hearing-impaired listeners with "unknown" datasets (i.e. a set on which it was not trained) and distortions which are specific to hearing aids. Relatively recently, however, Kates and Arehart (Journal of the Audio Engineering Society, 2010) proposed the Hearing Aid Speech Quality Index (HASQI), which is a model-based objective measure that predicts quality for normal-hearing and hearing-impaired listeners by taking into account many of the distortions which hearing aids introduce. HASQI solves many of our concerns of generalizability for predicting quality, but it still remains to test HASQI's ability to predict quality with datasets on which it was not trained. Thus, we explore the robustness of HASQI by testing its ability to predict quality for "unknown" de-noised speech, and we directly compare its performance to some other metrics in the literature.

Vivemos atualmente uma era de convergência de tecnologias, motivada por questões tanto econômicas como de caráter operacional, na qual os serviços de dados, voz e vídeo estão migrando rapidamente para uma plataforma IP. Particularmente considerando o paradigma da telefonia IP neste processo de convergência, ocorrem desafios tecnológicos, pois temos de um lado os usuários finais que já possuem uma referência sobre a qualidade da fala, fruto das décadas de uso do sistema telefônico tradicional, e na outra extremidade as operadoras de telecomunicações que em sua última milha dependem de redes estatísticas, sem mecanismos adequados para a garantia de QoS. Assim, se torna vital para o sucesso da operação a devida identificação das relações entre os diversos componentes existentes entre terminais e sua contribuição para a qualidade de fala, percebida pelo assinante, de forma a entregar um serviço com qualidade similar ao Sistema Telefônico Fixo Comutado. Neste contexto, este trabalho busca identificar por meio de técnicas de estatística multivariada uma correlação entre métricas objetivas de Qualidade de Serviços aplicáveis em redes IP e a qualidade subjetiva da fala predita pelo algoritmo Modelo-E definido na recomendação ITU-T G.107. Um método de coleta e análise estatística de informações foi desenvolvido para atingir o objetivo proposto. Para sua validação um ambiente de testes foi criado, dados de operação foram coletados e ferramentas computacionais foram aplicadas para o tratamento analítico e estatístico. Os resultados obtidos pelo método foram então aplicados em campo durante as etapas de testes e homologação de um PABX-IP-IMS desenvolvido para o mercado corporativo. A correlação entre os diversos fatores envolvidos, suas métricas e como todo este sistema impacta na qualidade relativa, percebida pelo usuário final permitirá aos provedores de serviços avaliarem quais as melhores estratégias a serem empregadas em seus segmentos de rede de forma a garantir a excelência no nível de serviço oferecido ao consumidor final. / We live now in an convergence of technologies era, driven by economic and operational issues, where the data services, voice and video are quickly moving to an IP platform. Particularly considering the paradigm of IP telephony in the process of convergence, there are technological challenges. We have subscribers who already have a reference about the quality of speech, derived from decades of using the traditional phone system. At the other end telecom operators that rely on statistical networks, with no possibility to guarantee QoS. So it becomes vital to the operations success the proper identification of the relationships between the various components between the terminals and their contribution to the speech quality perceived by the subscriber in order to deliver a quality service close to the PSTN. In this context, this study sought to identify a correlation between objective metrics for Quality of Service applicable to IP networks and subjective quality of speech predicted by the algorithm \"Model-E\" defined in ITU-T G.107 through multivariate statistical techniques. A method of collecting and analyzing statistical information was developed to achieve the proposed objective. To validate a test environment was created, operation data were collected and computational tools were applied to the analytical and statistical treatment. The results obtained by the method were then applied in the field during the stages of testing and approval of an IMS-IP-PBX designed for the corporate market. The correlation between the various factors involved, their metrics and how the whole system impacts on the quality perceived by end users will enable service providers to assess what the best strategies to use in their network segments to ensure an adequate level of service offered to consumers.

This master’s thesis deals with application of the ITU-T G.107 E-model for objective non-intrusive voice transmission quality measurements in LTE and Wi-Fi networks. The first part presents techniques used for voice quality measurements, particularly algorithm POLQA, analysis of the E-model and overview of the VoLTE and VoWiFi technologies. The main part of this paper consists of design of the R-factor calculation formula using parameters measured by Android OS powered devices. The algorithm design is based on extensive VoLTE measurements and its voice quality prediction successfulness is evaluated by comparison with POLQA measurements. The paper also presents implementation possibilities of the proposed algorithm on devices with Android OS.