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

Adaptive Playout Buffer Algorithm for Enhancing Perceived Quality of Streaming Applications

The end-to-end packet delay is an important performance parameter on the Internet, because it heavily affects the quality of realtime applications. Currently, however, because the packet transmission quality (e.g., transmission delay, jitter, packet loss) may vary dynamically, it is not easy to handle real-time traffic. For UDP-based real-time applications, a smoothing buffer (playout buffer) is typically used at the client to compensate for variable delays. The issue of playout control has been studied previously, and several algorithms for controlling the playout buffer have been proposed. These studies considered the network parameters (e.g., packet loss ratio and playout delay), but not the quality perceived by end users

Adaptive Playout Buffer Algorithm for Enhancing Perceived Quality of Streaming Applications

An end-to-end packet delay in the Internet is an important performance parameter, because it heavily affects the quality of real-time applications. In the current Internet, however, because the packet trans-mission qualities (e.g., transmission delays, jitters, packet losses) may vary dynamically, it is not easy to handle a real-time traffic. In UDP based real-time applications, a smoothing buffer (playout buffer) is typically used at a client host to compensate for variable delays. The issue of playout control has been studied by some previous works, and several algorithms controlling the playout buffer have been pro-posed. These studies have controlled the network parameters (e.g., packet loss ratio and playout delay), not considered the quality perceived by users. In this paper, we first clarify the relationship between Mean Opinion Score (MOS) of played audio and network parameters (e.g., packet loss, packet transmission delay, transmission rate). Next, utilizing the MOS function, we propose a new playout buffer algorithm considering user’s perceived quality of real-time applications. Our simulation and implementation tests show that it can enhance the perceived quality, compared with existing algorithms. 1 All correspondence should be directed to

Evaluación de algoritmos de control de retardo en voz sobre internet

Este proyecto de tesis de maestría trata acerca de los algoritmos de control de retardo en VoIP. Las aplicaciones interactivas sobre Internet se utilizan extensamente en nuestros días. Aplicaciones P2P como Skype, VoIPBuster han incorporado exitosamente la VoIP. Un algoritmo de control de playout implementa un buffer en el lado del receptor, así guarda los paquetes recibidos. Entonces, el algoritmo calcula un tiempo límite para cada paquete. Si el paquete se perdió en la red o llegó después de su tiempo límite esperado, el paquete se considera perdido en el receptor. Un algoritmo de playout considera el compro-miso entre las pérdidas y el retardo de tal forma que optimice la interactividad de la sesión de VoIP. Nos enfocamos en la clase de algoritmos que actualizan el retardo de playout al ini-cio de cada frase. Estudiamos un algoritmo NLMS originálmente propuesto por DeLeon y posteriormente modificado por Shallwani para probarlos extensivamente bajo las mismas condiciones de trabajo. Los resultados que obtenemos indican por un lado que el algoritmo de Shallwani puede tener errores de depuración, y por el otro lado, que la detección de picos de retardo puede mejorarse. Así, decidimos mejorar la detección de picos de retardo que propone Shallwani y comparar el desempeño de nuestro algoritmo con los algoritmos de DeLeon y de Shallwani. Encontramos, que para la mayoría de los casos, usando trazas reales de audio, que han sido muy utilizadas en otros trabajos, nuestro algoritmo se desempeña mejor

Rétroaction de la qualité de l'expérience pour améliorer la qualité de service

Résumé Pendant de nombreuses années l'évaluation des réseaux IP a été effectuée de façon objective en mesurant différents critères qui déterminent la qualité du réseau. Ces critères font référence à un ensemble de paramètres de qualité de service (QdS) que doit satisfaire le réseau dans le transport des paquets. Avec l'apparition des trafics multimédias tels que la voix sur IP et la vidéo, les paramètres de QdS, quoique fondamentaux, se révèlent être insuffisants puisqu'ils ne tiennent pas compte de ce qui se passe en amont et en aval du réseau. C'est ce qui explique l'émergence du concept de la qualité de l'expérience (QdE) dans les réseaux IP pour les applications temps réel et interactives. Ce concept, exprimant le niveau de satisfaction de l'usager, permet de tenir compte de ce dernier dans la boucle de service. Ainsi, il s'avère nécessaire de disposer de mécanismes pour contrôler la QdE des usagers et qui sont capables de réagir lorsqu'une dégradation se produit. Les mécanismes proposés jusqu'ici concernent, pour la plupart, les applications de voix sur IP ou la vidéo. Ils agissent le plus souvent au niveau des équipements terminaux particulièrement sur les codeurs ou les décodeurs en modifiant leur débit ou en faisant varier la taille du tampon de la gigue. De ce fait, ils ne sont appropriés que pour les applications multimédias et ne peuvent pas être appliqués à autres applications interactives, les jeux vidéo en ligne par exemple, auxquelles le concept de QdE s'est élargi. De plus, très peu de travaux proposent d'utiliser les métriques de QdE pour avoir de meilleurs mécanismes de QdS. Dans ce mémoire, un nouveau mécanisme est proposé qui, via des techniques utilisées pour faire de la QdS dans les réseaux IP, permet de réagir lorsque la QdE pour une application subit des dégradations. Ce mécanisme comporte deux aspects. Le premier aspect consiste à définir un cadre réaliste pour, d'une part, avoir la rétroaction de la QdE (feed-back) pour tout type d'applications, temps réel et non temps réel; et d'autre part, introduire le feed-back de la QdE (re-feed-back) dans le réseau de sorte que les différents nœuds puissent en être informés. La réinsertion du feed-back de la QdE se fait ‘in band’ à travers l'entête IP. Le second aspect consiste à proposer un nouveau mécanisme de classification des paquets par les routeurs internes à un domaine DiffServ. Ce mécanisme permet aux routeurs de tenir compte de la QdE lors de la mise en file d'attente des paquets d'un flot de trafic qui doit bénéficier d'une certaine garantie de QdE. L'information de la QdE sera accessible aux routeurs par le mécanisme de feed-back proposé qui l'écrira dans l'entête des paquets IP. Le mécanisme proposé a été simulé avec le simulateur NS-3. Les résultats obtenus démontrent que l'utilisation du feed-back de la métrique de la QdE dans DiffServ permet d'obtenir de meilleurs résultats de QdE, prouvant ainsi qu'il est possible et utile d'impliquer la QdE dans les mécanismes de QdS.----------Abstract For many years the evaluation of IP networks was carried out objectively by measuring different criteria that determine the quality of the network. Those criteria refer to a set of parameters of quality of service (QoS) that must be satisfied by the network while carrying packets. With the emergence of multimedia traffic such as Voice over IP (VoIP) and video, QoS parameters, though still very important, are yet not enough because they do not take into account what is happening at the end nodes. This has led to the emergence of the new concept of quality of experience (QoE) which expresses the user's satisfaction regarding a given comunication service. Thus it is necessary to develop mechanisms in IP netwoks allowing one to monitor and enhance the users' QoE. So far, proposed mechanisms concern mostly VoIP or video applications and act, most of the time, at the terminal equipment, particularly at the codec to adapt, if possible, the data transmission rate or to modify the jitter buffer size. So, they are only suitable for multimedia applications and they do not take into account other interactive applications like, online games for example, to which QoE concept has widened. In addition, very few works propose to use QoE metrics to enhance QoS mechanisms. In this master thesis, we propose a new mechanism QoS-based mechanism which is able to react when the QoE value of an application goes below a threshold. This mechanism has two aspects. The first aspect is to define a realistic framework to have the QoE information echoed to the receiver on the one hand and secondly to insert the QoE information in the network so that nodes on the path can use it in QoS mechanism. That is done `in band' through IP packet header. The second aspect consists of proposing a new mechanism to classify packets by routers inside a DiffServ domain. This mechanism enables routers to take into account the QoE info. The QoE information will be accessible by the routers by the proposed feedback mechanism. The proposed mechanism was simulated with the ns-3 simulator. The results show that the use of feedback of the QoE info in DiffServ achieves better QoE results proving that it is possible and useful to involve the QoE in QoS mechanisms

Measuring And Improving Internet Video Quality Of Experience

Streaming multimedia content over the IP-network is poised to be the dominant Internet traffic for the coming decade, predicted to account for more than 91% of all consumer traffic in the coming years. Streaming multimedia content ranges from Internet television (IPTV), video on demand (VoD), peer-to-peer streaming, and 3D television over IP to name a few. Widespread acceptance, growth, and subscriber retention are contingent upon network providers assuring superior Quality of Experience (QoE) on top of todays Internet. This work presents the first empirical understanding of Internet’s video-QoE capabilities, and tools and protocols to efficiently infer and improve them. To infer video-QoE at arbitrary nodes in the Internet, we design and implement MintMOS: a lightweight, real-time, noreference framework for capturing perceptual quality. We demonstrate that MintMOS’s projections closely match with subjective surveys in accessing perceptual quality. We use MintMOS to characterize Internet video-QoE both at the link level and end-to-end path level. As an input to our study, we use extensive measurements from a large number of Internet paths obtained from various measurement overlays deployed using PlanetLab. Link level degradations of intra– and inter–ISP Internet links are studied to create an empirical understanding of their shortcomings and ways to overcome them. Our studies show that intra–ISP links are often poorly engineered compared to peering links, and that iii degradations are induced due to transient network load imbalance within an ISP. Initial results also indicate that overlay networks could be a promising way to avoid such ISPs in times of degradations. A large number of end-to-end Internet paths are probed and we measure delay, jitter, and loss rates. The measurement data is analyzed offline to identify ways to enable a source to select alternate paths in an overlay network to improve video-QoE, without the need for background monitoring or apriori knowledge of path characteristics. We establish that for any unstructured overlay of N nodes, it is sufficient to reroute key frames using a random subset of k nodes in the overlay, where k is bounded by O(lnN). We analyze various properties of such random subsets to derive simple, scalable, and an efficient path selection strategy that results in a k-fold increase in path options for any source-destination pair; options that consistently outperform Internet path selection. Finally, we design a prototype called source initiated frame restoration (SIFR) that employs random subsets to derive alternate paths and demonstrate its effectiveness in improving Internet video-QoE