IMPROVING QoS OF VoWLAN VIA CROSS-LAYER BASED ADAPTIVE APPROACH

Voice over Internet Protocol (VoIP) is a technology that allows the transmission of voice packets over Internet Protocol (IP). Recently, the integration of VoIP and Wireless Local Area Network (WLAN), and known as Voice over WLAN (VoWLAN), has become popular driven by the mobility requirements ofusers, as well as by factor of its tangible cost effectiveness. However, WLAN network architecture was primarily designed to support the transmission of data, and not for voice traffic, which makes it lack ofproviding the stringent Quality ofService (QoS) for VoIP applications. On the other hand, WLAN operates based on IEEE 802.11 standards that support Link Adaptive (LA) technique. However, LA leads to having a network with multi-rate transmissions that causes network bandwidth variation, which hence degrades the voice quality. Therefore, it is important to develop an algorithm that would be able to overcome the negative effect of the multi-rate issue on VoIP quality. Hence, the main goal ofthis research work is to develop an agent that utilizes IP protocols by applying a Cross-Layering approach to eliminate the above-mentioned negative effect. This could be expected from the interaction between Medium Access Control (MAC) layer and Application layer, where the proposed agent adapts the voice packet size at the Application layer according to the change of MAC transmission data rate to avoid network congestion from happening. The agent also monitors the quality of conversations from the periodically generated Real Time Control Protocol (RTCP) reports. If voice quality degradation is detected, then the agent performs further rate adaptation to improve the quality. The agent performance has been evaluated by carrying out an extensive series ofsimulation using OPNET Modeler. The obtained results of different performance parameters are presented, comparing the performance ofVoWLAN that used the proposed agent to that ofthe standard network without agent. The results ofall measured quality parameters hav

VOIP WITH ADAPTIVE RATE IN MULTI- TRANSMISSION RATE WIRELESS LANS

“Voice over Internet Protocol (VoIP)” is a popular communication technology that plays a vital role in term of cost reduction and flexibility. However, like any emerging technology, there are still some issues with VoIP, namely providing good Quality of Service (QoS), capacity consideration and providing security. This study focuses on the QoS issue of VoIP, specifically in “Wireless Local Area Networks (WLAN)”. IEEE 802.11 is the most popular standard of wireless LANs and it offers different transmission rates for wireless channels. Different transmission rates are associated with varying available bandwidth that shall influence the transmission of VoIP traffic

Measures of quality of experience based on the E-model during a VoIP call

Orientador: Yuzo IanoTese (doutorado) ¿ Universidade Estadual de Campinas, Faculdade de Engenharia Elétrica e de ComputaçãoResumo: A tecnologia de voz sobre protocolo de internet (voz sobre IP) ou simplesmente, VoIP (Voice Over Internet Protocol) ganha novos usuários decorrente do cenário atual de mercado de convergência de redes de dados e telecomunicações. Entretanto, o sucesso desta tecnologia depende fortemente da qualidade do sinal da fala percebida pelo usuário, uma medida subjetiva, também conhecida como Qualidade de Experiência (Quality of Experience - QoE). Esta Tese propõe e valida um mecanismo de observação de variáveis de ambiente durante uma comunicação VoIP tendo como saída um número que representa a QoE não referenciada (sem o sinal transmitido, somente o sinal recebido) ao utilizar o Modelo-E com o propósito de selecionar os melhores parâmetros disponíveis de configuração que afetam o fluxo de voz e ao mesmo tempo obter a melhor qualidade de chamada possível dentro de um determinado cenário de rede. O resultado obtido fez a ligação entre uma medida objetiva, o parâmetro R gerado pelo Modelo-E, e uma medida subjetiva estimada, o MOS (Mean Opinion Score), durante uma chamada VoIP, não se limitando a medida em si. Todavia, cobriu-se todo o cenário para a medição e a comparação com sistemas padrões de medição de qualidade, formando uma base de conhecimento com os resultados obtidos. O método utilizado de estimativa de qualidade da fala foi comparado em diferentes codecs de voz padrão ITU-T (PCMU, GSM, G.723, G.729, G.726-32), testados em uma topologia de rede que sofreu distorções, como diferentes situações de perdas de dados (0,0%, 1,0%, 2,0%, 2,5%, 3,0%, 5,0%, 7,5%, 10,0%). Uma análise de regressão foi utilizada para permitir uma melhor compreensão do impacto das condições de rede e codecs sobre a QoE do serviço VoIP medido. Foi utilizado um suíte de testes padronizados para medição da QoE nos arquivos de voz recebidos e transmitidos durante os testes baseados em testes referenciados (com os sinais transmitidos e recebidos) nos padrões ITU-T P.863 (Perceptual Objective Listening Quality Assessment - POLQA) e ITU-T P.862 (Perceptual Evaluation of Speech Quality - PESQ) e os resultados foram comparados com os obtidos pelo método não referenciado proposto para medida de QoE. Para os resultados do codec testado foi aplicado o método de regressão linear sendo a variável independente as medidas de QoE obtidas pelo método proposto e a variável dependente foram os resultados obtidos pelos algoritmos PESQ e POLQA. Para todos os codecs testados, o Coeficiente de Determinação (R2) entre o método proposto e os resultados obtidos pelo algoritmo PESQ foram superiores a 0,90 indicando uma forte correlação linear. Já entre o método proposto e o algoritmo POLQA, para os codecs PCMU, GSM e G.723, os resultados de R2 foram superiores a 0,973, indicando uma correlação muito forte. R2 para o codec G.726-32 foi de 0,88 indicando uma correlação forte. Já para o codec G.729, R2 ficou em 0,67 indicando que o modelo linear pode não ser o mais adequado para explicar a relação entre os resultados do método proposto e os valores obtidos pelo algoritmo POLQAAbstract: The technology of Voice over Internet Protocol (Voice over IP or simply VoIP) is present in our personal and professional lives. The number of VoIP users increases day after day due to the current scenario of convergence of data and telecommunications networks. However, the success of this technology depends on the speech signal quality as perceived by the user, a subjective measure as function of the user's point of view, also known as the Quality of Experience (QoE). This thesis proposes and validates an environment variable observation mechanism during a VoIP communication having as output a number that represents the QoE not referenced (without the transmitted signal, only the received signal) of the call, using the E-Model, in order to select the best available parameter settings that affect voice flow of the current VoIP call and at the same time gets the best call quality as possible within a given network scenario. The result relates an objective measurement, the R parameter generated by the E-Model, to the estimated subjective measurement, MOS, during a VoIP call, not limited to the measurement itself. However, it covered the whole scenario for measurement and comparison with quality measurement standards systems, forming a knowledge base with the results. The method of speech quality estimation was compared in different standard voice codec's ITU-T (PCMU, GSM, G.723, G.729, G.726-32) tested in a network topology that has suffered distortions, as different situations of data loss (0.0%, 1.0%, 2.0%, 2.5%, 3.0%, 5.0%, 7.5%, 10.0%). A regression analysis was used to allow a better understanding of the impact of network conditions and codec's on the VoIP service QoE measured. In this thesis it was used a suite of standardized tests for measuring QoE in voice files received and transmitted during testing based on referenced tests (with the transmitted and received signals) in the ITU-T P.863 standard (Perceptual Objective Listening Quality Assessment - POLQA) and ITU-T P.862 (Perceptual Evaluation of Speech Quality - PESQ) and the results were compared with those obtained by the no referenced measure method of QoE proposed in this thesis. The linear regression was applied in order to analyze the results of the tested codec. The independent variable was the QoE measurements obtained by the proposed method to measure QoE and the dependent variable were the results of PESQ and POLQA algorithms. For all tested codec's, the Coefficient of Determination (R2) between the proposed method and the results of the PESQ algorithm was higher than 0.90 indicates a strong linear correlation. For PCMU, G.723, GSM codec's R2 was greater than 0.973 indicating a strong correlation between the results of proposed method and the results of POLQA algorithm. R2 for G.726-32 codec was 0.88 that indicates a high correlation. For G.729 codec, R2 was 0.67 that indicates the linear model may not be the most appropriate to explain the relationship between the results of the proposed method and values obtained by POLQA algorithmDoutoradoTelecomunicações e TelemáticaDoutor em Engenharia Elétric

VOIP WITH ADAPTIVE RATE IN MULTI- TRANSMISSION RATE WIRELESS LANS

“Voice over Internet Protocol (VoIP)” is a popular communication technology that plays a vital role in term of cost reduction and flexibility. However, like any emerging technology, there are still some issues with VoIP, namely providing good Quality of Service (QoS), capacity consideration and providing security. This study focuses on the QoS issue of VoIP, specifically in “Wireless Local Area Networks (WLAN)”. IEEE 802.11 is the most popular standard of wireless LANs and it offers different transmission rates for wireless channels. Different transmission rates are associated with varying available bandwidth that shall influence the transmission of VoIP traffic

Multimedia

The nowadays ubiquitous and effortless digital data capture and processing capabilities offered by the majority of devices, lead to an unprecedented penetration of multimedia content in our everyday life. To make the most of this phenomenon, the rapidly increasing volume and usage of digitised content requires constant re-evaluation and adaptation of multimedia methodologies, in order to meet the relentless change of requirements from both the user and system perspectives. Advances in Multimedia provides readers with an overview of the ever-growing field of multimedia by bringing together various research studies and surveys from different subfields that point out such important aspects. Some of the main topics that this book deals with include: multimedia management in peer-to-peer structures & wireless networks, security characteristics in multimedia, semantic gap bridging for multimedia content and novel multimedia applications

Context-awareness for ubiquitous media service delivery in next generation networks

Les récentes avancées technologiques permettent désormais la fabrication de terminaux mobiles de plus en plus compacts et dotés de plusieurs interfaces réseaux. Le nouveau modèle de consommation de médias se résume par le concept "Anytime, Anywhere, Any Device" et impose donc de nouvelles exigences en termes de déploiement de services ubiquitaires. Cependant la conception et le developpement de réseaux ubiquitaires et convergents de nouvelles générations soulèvent un certain nombre de défis techniques. Les standards actuels ainsi que les solutions commerciales pourraient être affectés par le manque de considération du contexte utilisateur. Le ressenti de l'utilisateur concernant certains services multimédia tels que la VoIP et l'IPTV dépend fortement des capacités du terminal et des conditions du réseau d'accès. Cela incite les réseaux de nouvelles générations à fournir des services ubiquitaires adaptés à l'environnement de l'utilisateur optimisant par la même occasion ses resources. L'IP Multimedia Subsystem (IMS) est une architecture de nouvelle génération qui centralise l'accès aux services et permet la convergence des réseaux fixe/mobile. Néanmoins, l'évolution de l'IMS est nécessaire sur les points suivants :- l'introduction de la sensibilité au contexte utilisateur et de la PQoS (Perceived QoS) : L'architecture IMS ne prend pas en compte l'environnement de l'utilisateur, ses préférences et ne dispose pas d'un méchanisme de gestion de PQOS. Pour s'assurer de la qualité fournit à l'utilisateur final, des informations sur l'environnement de l'utilisateur ainsi que ses préférences doivent transiter en cœur de réseau afin d'y être analysés. Ce traitement aboutit au lancement du service qui sera adapté et optimisé aux conditions observées. De plus pour le service d'IPTV, les caractéristiques spatio-temporelles de la vidéo influent de manière importante sur la PQoS observée côté utilisateur. L'adaptation des services multimédias en fonction de l'évolution du contexte utilisateur et de la nature de la vidéo diffusée assure une qualité d'expérience à l'utilisateur et optimise par la même occasion l'utilisation des ressources en cœur de réseau.- une solution de mobilité efficace pour les services conversationnels tels que la VoIP : Les dernières publications 3GPP fournissent deux solutions de mobilité: le LTE proposeMIP comme solution de mobilité alors que l'IMS définit une mobilité basée sur le protocoleapplicatif SIP. Ces standards définissent le système de signalisation mais ne s'avancent pas sur la gestion du flux média lors du changement d'interface réseau. La deuxième section introduit une étude comparative détaillée des solutions de mobilité dans les NGNs.Notre première contribution est la spécification de l'architecture globale de notre plateforme IMS sensible au contexte utilisateur réalisée au sein du projet Européen ADAMANTIUM. Nous détaillons tout d'abord le serveur MCMS intelligent placé dans la couche application de l'IMS. Cet élément récolte les informations de qualité de services à différents équipements réseaux et prend la décision d'une action sur l'un de ces équipements. Ensuite nous définissons un profil utilisateur permettant de décrire son environnement et de le diffuser en coeur de réseau. Une étude sur la prédiction de satisfaction utilisateur en fonction des paramètres spatio-temporels de la vidéo a été réalisée afin de connaître le débit idéal pour une PQoS désirée.Notre deuxième contribution est l'introduction d'une solution de mobilité adaptée aux services conversationnels (VoIP) tenant compte du contexte utilisateur. Notre solution s'intègre à l'architecture IMS existante de façon transparente et permet de réduire le temps de latence du handover. Notre solution duplique les paquets de VoIP sur les deux interfaces actives pendant le temps de la transition. Parallèlement, un nouvel algorithme de gestion de mémoire tampon améliore la qualité d'expérience pour le service de VoIP.The latest advances in technology have already defied Moore s law. Thanks to research and industry, hand-held devices are composed of high processing embedded systems enabling the consumption of high quality services. Furthermore, recent trends in communication drive users to consume media Anytime, Anywhere on Any Device via multiple wired and wireless network interfaces. This creates new demands for ubiquitous and high quality service provision management. However, defining and developing the next generation of ubiquitous and converged networks raise a number of challenges. Currently, telecommunication standards do not consider context-awareness aspects for network management and service provisioning. The experience felt by the end-user consuming for instance Voice over IP (VoIP) or Internet Protocol TeleVision (IPTV) services varies depending mainly on user preferences, device context and network resources. It is commonly held that Next Generation Network (NGN) should deliver personalized and effective ubiquitous services to the end user s Mobile Node (MN) while optimizing the network resources at the network operator side. IP Multimedia Subsystem (IMS) is a standardized NGN framework that unifies service access and allows fixed/mobile network convergence. Nevertheless IMS technology still suffers from a number of confining factors that are addressed in this thesis; amongst them are two main issues :The lack of context-awareness and Perceived-QoS (PQoS):-The existing IMS infrastructure does not take into account the environment of the user ,his preferences , and does not provide any PQoS aware management mechanism within its service provisioning control system. In order to ensure that the service satisfies the consumer, this information need to be sent to the core network for analysis. In order to maximize the end-user satisfaction while optimizing network resources, the combination of a user-centric network management and adaptive services according to the user s environment and network conditions are considered. Moreover, video content dynamics are also considered as they significantly impact on the deduced perceptual quality of IPTV services. -The lack of efficient mobility mechanism for conversational services like VoIP :The latest releases of Third Generation Partnership Project (3GPP) provide two types of mobility solutions. Long-Term Evolution (LTE) uses Mobile IP (MIP) and IMS uses Session Initiation Protocol (SIP) mobility. These standards are focusing on signaling but none of them define how the media should be scheduled in multi-homed devices. The second section introduces a detailed study of existing mobility solutions in NGNs. Our first contribution is the specification of the global context-aware IMS architecture proposed within the European project ADAptative Management of mediA distributioN based on saTisfaction orIented User Modeling (ADAMANTIUM). We introduce the innovative Multimedia Content Management System (MCMS) located in the application layer of IMS. This server combines the collected monitoring information from different network equipments with the data of the user profile and takes adaptation actions if necessary. Then, we introduce the User Profile (UP) management within the User Equipment (UE) describing the end-user s context and facilitating the diffusion of the end-user environment towards the IMS core network. In order to optimize the network usage, a PQoS prediction mechanism gives the optimal video bit-rate according to the video content dynamics. Our second contribution in this thesis is an efficient mobility solution for VoIP service within IMS using and taking advantage of user context. Our solution uses packet duplication on both active interfaces during handover process. In order to leverage this mechanism, a new jitter buffer algorithm is proposed at MN side to improve the user s quality of experience. Furthermore, our mobility solution integrates easily to the existing IMS platform.BORDEAUX1-Bib.electronique (335229901) / SudocSudocFranceF