Enhancement of perceived quality of service for voice over internet protocol systems

Voice over Internet Protocol (WIP) applications are becoming more and more popular in the telecommunication market. Packet switched V61P systems have many technical advantages over conventional Public Switched Telephone Network (PSTN), including its efficient and flexible use of the bandwidth, lower cost and enhanced security. However, due to the IP network's "Best Effort" nature, voice quality are not naturally guaranteed in the VoIP services. In fact, most current Vol]P services can not provide as good a voice quality as PSTN. IP Network impairments such as packet loss, delay and jitter affect perceived speech quality as do application layer impairment factors, such as codec rate and audio features. Current perceived Quality of Service (QoS) methods are mainly designed to be used in a PSTN/TDM environment and their performance in V6IP environment is unknown. It is a challenge to measure perceived speech quality correctly in V61P system and to enhance user perceived speech quality for VoIP system. The main goal of this project is to evaluate the accuracy of the existing ITU-T speech quality measurement method (Perceptual Evaluation of Speech Quality - PESQ) in mobile wireless systems in the context of V61P, and to develop novel and efficient methods to enhance the user perceived speech quality for emerging V61P services especially in mobile V61P environment. The main contributions of the thesis are threefold: (1) A new discovery of PESQ errors in mobile VoIP environment. A detailed investigation of PESQ performance in mobile VoIP environment was undertaken and included setting up a PESQ performance evaluation platform and testing over 1800 mobile-to-mobile and mobileto- PSTN calls over a period of three months. The accuracy issues of PESQ algorithm was investigated and main problems causing inaccurate PESQ score (improper time-alignment in the PESQ algorithm) were discovered . Calibration issues for a safe and proper PESQ testing in mobile environment were also discussed in the thesis. (2) A new, simple-to-use, V611Pjit ter buffer algorithm. This was developed and implemented in a commercial mobile handset. The algorithm, called "Play Late Algorithm", adaptively alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. It can be used as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance. Results show that the proposed algorithm can increase user perceived quality without consuming too much processing power when tested in live wireless VbIP networks. (3) A new QoS enhancement scheme. The new scheme combines the strengths of adaptive codec bit rate (i. e. AMR 8-modes bit rate) and speech priority marking (i. e. giving high priority for the beginning of a voiced segment). The results gathered on a simulation and emulation test platform shows that the combined method provides a better user perceived speech quality than separate adaptive sender bit rate or packet priority marking methods

A methodology for obtaining More Realistic Cross-Layer QoS Measurements in mobile networks: A VoIP over LTE Use Case

Los servicios de voz han sido durante mucho tiempo la primera fuente de ingresos para los operadores móviles. Incluso con el protagonismo creciente del tráfico de datos, los servicios de voz seguirán jugando un papel importante y no desaparecerán con la transición a redes basadas en el protocolo IP. Por otra parte, hace años que los principales actores en la industria móvil detectaron claramente que los usuarios no aceptarían una degradación en la calidad de los servicios de voz. Es por esto que resulta crítico garantizar la experiencia de usuario (QoE) en la transición a redes de nueva generación basadas en conmutación de paquetes. El trabajo realizado durante esta tesis ha buscado analizar el comportamiento y las dependencias de los diferentes servicios de Voz sobre IP (VoIP), así como identificar configuraciones óptimas, mejoras potenciales y metodologías que permitan asegurar niveles de calidad aceptables al mismo tiempo que se trate de minimizar los costes. La caracterización del rendimiento del tráfico de datos en redes móviles desde el punto de vista de los usuarios finales es un proceso costoso que implica la monitorización y análisis de un amplio rango de protocolos y parámetros con complejas dependencias. Para abordar desde la raíz este problema, se requiere realizar medidas que relacionen y correlen el comportamiento de las diferentes capas. La metodología de caracterización propuesta en esta tesis proporciona la posibilidad de recoger información clave para la resolución de problemas en las comunicaciones IP, relaciolándola con efectos asociados a la propagación radio, como cambios de celda o pérdida de enlaces, o con carga de la red y limitaciones de recursos en zonas geográficas específicas. Dicha metodología se sustenta en la utilización de herramientas nativas de monitorización y registro de información en smartphones, y la aplicación de cadenas de herramientas para la experimentación extensiva tanto en redes reales y como en entornos de prueba controlados. Con los resultados proporcionados por esta serie de herramientas, tanto operadores móviles y proveedores de servicio como desarrolladores móviles podrían ganar acceso a información sobre la experiencia real del usuario y sobre cómo mejorar la cobertura, optimizar los servicios y adaptar el funcionamiento de las aplicaciones y el uso de protocolos móviles basados en IP en este contexto. Las principales contribuciones de las herramientas y métodos introducidos en esta tesis son los siguientes: - Una herramienta de monitorización multicapa para smartphones Android, llamada TestelDroid, que permite la captura de indicadores clave de rendimiento desde el propio equipo de usuario. Asimismo proporciona la capacidad de generar tráfico de forma activa y de verificar el estado de alcanzabilidad del terminal, realizando pruebas de conectividad. - Una metodología de post-procesado para correlar la información presente en las diferentes capas de las medidas realizadas. De igual forma, se proporciona la opción a los usuarios de acceder directamente a la información sobre el tráfico IP y las medidas radio y de aplicar metodologías propias para la obtención de métricas. - Se ha realizado la aplicación de la metodología y de las herramientas usando como caso de uso el estudio y evaluación del rendimiento de las comunicaciones basadas en IP a bordo de trenes de alta velocidad. - Se ha contribuido a la creación de un entorno de prueba realista y altamente configurable para la realización de experimentos avanzados sobre LTE. - Se han detectado posibles sinergias en la utilización de instrumentación avanzada de I+D en el campo de las comunicaciones móviles, tanto para la enseñanza como para la investigación en un entorno universitario

A novel non-intrusive objective method to predict voice quality of service in LTE networks.

This research aimed to introduce a novel approach for non-intrusive objective measurement of voice Quality of Service (QoS) in LTE networks. While achieving this aim, the thesis established a thorough knowledge of how voice traffic is handled in LTE networks, the LTE network architecture and its similarities and differences to its predecessors and traditional ground IP networks and most importantly those QoS affecting parameters which are exclusive to LTE environments. Mean Opinion Score (MOS) is the scoring system used to measure the QoS of voice traffic which can be measured subjectively (as originally intended). Subjective QoS measurement methods are costly and time-consuming, therefore, objective methods such as Perceptual Evaluation of Speech Quality (PESQ) were developed to address these limitations. These objective methods have a high correlation with subjective MOS scores. However, they either require individual calculation of many network parameters or have an intrusive nature that requires access to both the reference signal and the degraded signal for comparison by software. Therefore, the current objective methods are not suitable for application in real-time measurement and prediction scenarios. A major contribution of the research was identifying LTE-specific QoS affecting parameters. There is no previous work that combines these parameters to assess their impacts on QoS. The experiment was configured in a hardware in the loop environment. This configuration could serve as a platform for future research which requires simulation of voice traffic in LTE environments. The key contribution of this research is a novel non-intrusive objective method for QoS measurement and prediction using neural networks. A comparative analysis is presented that examines the performance of four neural network algorithms for non-intrusive measurement and prediction of voice quality over LTE networks. In conclusion, the Bayesian Regularization algorithm with 4 neurons in the hidden layer and sigmoid symmetric transfer function was identified as the best solution with a Mean Square Error (MSE) rate of 0.001 and regression value of 0.998 measured for the testing data set

A VoIP Implementation on an Embedded Platform

This thesis presents a method of building an open VoIP telephone using off-the-shelf hardware. VoIP has become an important area of study in computer science and engineering, but many of the pieces are expensive and proprietary. We discuss the process of building an open IP telephone, the design decisions and difficulties, a performance evaluation of the different pieces of the system, and methods and suggestions for improving the overall system. The IP telephone is built on the Freescale 68HC12 microcontroller because of its analog and digital capabilities and the Linksys WRT54GL router for its embedded simplicity and network capabilities. The lightweight Embedded Xinu operating system was selected to build up the VoIP capabilities on top of a functional network stack. Our results show that building this VoIP device is viable. The completed IP telephone establishes a call between two routers using UDP network transport and μ-law companding. The roundtrip latency is less than two milliseconds, and the phone has a PESQ MOS (voice quality) of 3.13

Quality of media traffic over Lossy internet protocol networks: Measurement and improvement.

Voice over Internet Protocol (VoIP) is an active area of research in the world of communication. The high revenue made by the telecommunication companies is a motivation to develop solutions that transmit voice over other media rather than the traditional, circuit switching network. However, while IP networks can carry data traffic very well due to their besteffort nature, they are not designed to carry real-time applications such as voice. As such several degradations can happen to the speech signal before it reaches its destination. Therefore, it is important for legal, commercial, and technical reasons to measure the quality of VoIP applications accurately and non-intrusively. Several methods were proposed to measure the speech quality: some of these methods are subjective, others are intrusive-based while others are non-intrusive. One of the non-intrusive methods for measuring the speech quality is the E-model standardised by the International Telecommunication Union-Telecommunication Standardisation Sector (ITU-T). Although the E-model is a non-intrusive method for measuring the speech quality, but it depends on the time-consuming, expensive and hard to conduct subjective tests to calibrate its parameters, consequently it is applicable to a limited number of conditions and speech coders. Also, it is less accurate than the intrusive methods such as Perceptual Evaluation of Speech Quality (PESQ) because it does not consider the contents of the received signal. In this thesis an approach to extend the E-model based on PESQ is proposed. Using this method the E-model can be extended to new network conditions and applied to new speech coders without the need for the subjective tests. The modified E-model calibrated using PESQ is compared with the E-model calibrated using i ii subjective tests to prove its effectiveness. During the above extension the relation between quality estimation using the E-model and PESQ is investigated and a correction formula is proposed to correct the deviation in speech quality estimation. Another extension to the E-model to improve its accuracy in comparison with the PESQ looks into the content of the degraded signal and classifies packet loss into either Voiced or Unvoiced based on the received surrounding packets. The accuracy of the proposed method is evaluated by comparing the estimation of the new method that takes packet class into consideration with the measurement provided by PESQ as a more accurate, intrusive method for measuring the speech quality. The above two extensions for quality estimation of the E-model are combined to offer a method for estimating the quality of VoIP applications accurately, nonintrusively without the need for the time-consuming, expensive, and hard to conduct subjective tests. Finally, the applicability of the E-model or the modified E-model in measuring the quality of services in Service Oriented Computing (SOC) is illustrated

Ubiquitous Mesh Networking: application to mobile communication and information dissemination in a rural context

ICT has furthered the social and economic development of societies but, rural African communities have lagged behind due to issues such as sparse population, low household income, a lack of electricity and other basic infrastructure that make it unattractive for telecommunication service providers to extend service provision. Where the service is available, ubiquitous service coverage has not translated into ubiquitous access for individuals because of the associated costs. A community-wide WMN offering VoIP using fixed telephone handsets has been deployed as a viable alternative to the cellular service provider. The effectiveness of this WMN VoIP service springs from the mobile phone usage statistics which showed that the majority of calls made are intra-community. This dissertation has been an effort towards improved communication and access to information for the under-served communities. Key contributions include, mobile VoIP support, translation gateway deployment to make textual information accessible in voice form via the phone, IP-based radio for community information dissemination. The lack of electricity has been mitigated by the use of low-power devices. In order to circumvent the computational challenges posed by the processing and storage limitations of these devices, a decentralised system architecture whereby the processing and storage load are distributed across the mesh nodes has been proposed. High-performance equipment can be stationed at the closest possible place with electricity in the area and connectivity extended to the non-electrified areas using low-power mesh networking devices. Implementation techniques were investigated and performance parameters measured. The quality of service experienced by the user was assessed using objective methods and QoS correlation models. A MOS value of 4.29, i.e. very good, was achieved for the mobile VoIP call quality, with the underlying hardware supporting up to 15 point-to-point simultaneous calls using SIP and the G.711 based codec. Using the PEAQ algorithm to evaluate the IP-based radio, a PEAQ value of 4.15, i.e. good, was achieved. Streaming audio across the network reduces the available bandwidth by 8Kbps per client due to the unicast nature of streaming. Therefore, a multicast approach has been proposed for efficient bandwidth utilization. The quality of the text-to-voice service rendered by the translation gateway had a PESQ score of 1.6 i.e. poor. The poor performance can be attributed to the TTS engine implementation and also to the lack of robustness in the time-alignment module of the PESQ algorithm. The dissertation also proposes the use of the WMN infrastructure as a back-haul to isles of WSNs deployed in areas of interest to provide access to information about environmental variables useful in decision making

A study of mobile VoIP performance in wireless broadband networks

Voice service is to date still the killer mobile service and the main source for operator revenue for years to come. Additionally, voice service will evolve from circuit switched technologies towards packet based Voice over IP (VoIP). However, using VoIP over wireless networks different from 3GPP cellular technologies makes it also a disruptive technology in the traditional telecommunication sector. The focus of this dissertation is on determining mobile VoIP performance in different wireless broadband systems with current state of the art networks, as well as the potential disruption to cellular operators when mobile VoIP is deployed over different access networks. The research method is based on an empirical model. The model and experiments are well documented and based on industry standards for voice quality evaluation. The evaluation provides results from both experiments in a controlled laboratory setup as well as from live scenarios. The research scope is first, evaluate each network technology independently; second, investigate vertical handover mobility cases; third, determine other aspects directly affecting end user experience (e.g., call setup delay and battery lifetime). The main contribution of this work is a systematic examination of mobile VoIP performance and end user experience. The research results point out the main challenges for achieving call toll quality, and how derive the required changes and technological performance roadmap for improved VoIP service. That is, investigate how the performance and usability of mobile VoIP can eventually be improved to be a suitable substitute for circuit switched voice. In addition, we evaluate the potential disruption to cellular operators that mobile VoIP brings when deployed over other access networks. This research extends the available knowledge from simulations and provides an insight into actual end user experience, as well as the challenges of using embedded clients in handheld devices. In addition, we find several issues that are not visible or accounted for in simulations in regard to network parameters, required retransmissions and decreased battery lifetime. The conclusion is that although the network performance of several wireless networks is good enough for near toll quality voice in static scenarios, there are still a number of problems which make it currently unfeasible to use as a primary voice service. Moreover, under mobility scenarios performance is degraded. Finally, there are other issues apart from network performance such as energy consumption, hardware limitations and lack of supporting business models (e.g., for WiFi mesh) that further limit the possibility of rolling out mobile VoIP services

Speech quality prediction for voice over Internet protocol networks

Merged with duplicate record 10026.1/878 on 03.01.2017 by CS (TIS). Merged with duplicate record 10026.1/1657 on 15.03.2017 by CS (TIS)This is a digitised version of a thesis that was deposited in the University Library. If you are the author please contact PEARL Admin ([email protected]) to discuss options.IP networks are on a steep slope of innovation that will make them the long-term carrier of all types of traffic, including voice. However, such networks are not designed to support real-time voice communication because their variable characteristics (e.g. due to delay, delay variation and packet loss) lead to a deterioration in voice quality. A major challenge in such networks is how to measure or predict voice quality accurately and efficiently for QoS monitoring and/or control purposes to ensure that technical and commercial requirements are met. Voice quality can be measured using either subjective or objective methods. Subjective measurement (e.g. MOS) is the benchmark for objective methods, but it is slow, time consuming and expensive. Objective measurement can be intrusive or non-intrusive. Intrusive methods (e.g. ITU PESQ) are more accurate, but normally are unsuitable for monitoring live traffic because of the need for a reference data and to utilise the network. This makes non-intrusive methods(e.g. ITU E-model) more attractive for monitoring voice quality from IP network impairments. However, current non-intrusive methods rely on subjective tests to derive model parameters and as a result are limited and do not meet new and emerging applications. The main goal of the project is to develop novel and efficient models for non-intrusive speech quality prediction to overcome the disadvantages of current subjective-based methods and to demonstrate their usefulness in new and emerging VoIP applications. The main contributions of the thesis are fourfold: (1) a detailed understanding of the relationships between voice quality, IP network impairments (e.g. packet loss, jitter and delay) and relevant parameters associated with speech (e.g. codec type, gender and language) is provided. An understanding of the perceptual effects of these key parameters on voice quality is important as it provides a basis for the development of non-intrusive voice quality prediction models. A fundamental investigation of the impact of the parameters on perceived voice quality was carried out using the latest ITU algorithm for perceptual evaluation of speech quality, PESQ, and by exploiting the ITU E-model to obtain an objective measure of voice quality. (2) a new methodology to predict voice quality non-intrusively was developed. The method exploits the intrusive algorithm, PESQ, and a combined PESQ/E-model structure to provide a perceptually accurate prediction of both listening and conversational voice quality non-intrusively. This avoids time-consuming subjective tests and so removes one of the major obstacles in the development of models for voice quality prediction. The method is generic and as such has wide applicability in multimedia applications. Efficient regression-based models and robust artificial neural network-based learning models were developed for predicting voice quality non-intrusively for VoIP applications. (3) three applications of the new models were investigated: voice quality monitoring/prediction for real Internet VoIP traces, perceived quality driven playout buffer optimization and perceived quality driven QoS control. The neural network and regression models were both used to predict voice quality for real Internet VoIP traces based on international links. A new adaptive playout buffer and a perceptual optimization playout buffer algorithms are presented. A QoS control scheme that combines the strengths of rate-adaptive and priority marking control schemes to provide a superior QoS control in terms of measured perceived voice quality is also provided. (4) a new methodology for Internet-based subjective speech quality measurement which allows rapid assessment of voice quality for VoIP applications is proposed and assessed using both objective and traditional MOS test methods