Os constantes avanços nas áreas de transmissão e processamento de dados ao longo dos últimos anos permitiram a criação de diversas aplicações e serviços baseados em dados multimídia, como streaming de vídeo, videoconferências, aulas remotas e IPTV. Além disso, avanços nas demais áreas da computação e engenharias, possibilitaram a construção de uma enorme diversidade de dispositivos de acesso a esses serviços, desde computadores pessoais até celulares, para citar os mais utilizados atualmente. Muitas dessas aplicações e dispositivos estão amplamente difundidos hoje em dia, e, ao mesmo tempo em que a tecnologia avança, os usuários tornam-se mais exigentes, buscando sempre melhor qualidade nos serviços que utilizam. Devido à grande variedade de redes e dispositivos atuais, uma dificuldade existente é possibilitar o acesso universal a uma transmissão. Uma alternativa criada é utilizar transmissão de vídeo escalável com IP multicast e controlada por mecanismos para adaptabilidade e controle de congestionamento. O produto final dessas transmissões mulimídia são os próprios dados multimídia (vídeo e áudio, principalmente) que o usuário está recebendo, portanto a qualidade destes dados é fundamental para um bom desempenho do sistema e satisfação dos usuários. Este trabalho apresenta um estudo de avaliações subjetivas de qualidade aplicadas em sequências de vídeo codificadas através da extensão escalável do padrão H.264 (SVC). Foi executado um conjunto de testes para avaliar, principalmente, os efeitos da instabilidade da transmissão (variação do número de camadas de vídeo recebidas) e a influência dos três métodos de escalabilidade (espacial, temporal e de qualidade) na qualidade dos vídeos. As definições foram baseadas em um sistema de transmissão em camadas com utilização de protocolos para adaptabilidade e controle de congestionamento. Para execução das avaliações subjetivas foi feito o uso da metodologia ACR-HRR e recomendações das normas ITU-R Rec. BT.500 e ITU-T Rec. P.910. Os resultados mostram que, diferente do esperado, a instabilidade não provoca grandes alterações na qualidade subjetiva dos vídeos e que o método de escalabilidade temporal tende a apresentar qualidade bastante inferior aos outros métodos. As principais contribuições deste trabalho estão nos resultados obtidos nas avaliações, além da metodologia utilizada durante o desenvolvimento do trabalho (definição do plano de avaliação, uso das ferramentas como o JSVM, seleção do material de teste, execução das avaliações, entre outros), das aplicações desenvolvidas, da definição de alguns trabalhos futuros e de possíveis objetivos para avaliações de qualidade. / The constant advances in multimedia processing and transmission over the past years have enabled the creation of several applications and services based on multimedia data, such as video streaming, teleconference, remote classes and IPTV. Futhermore, a big variety of devices, that goes from personal computers to mobile phones, are now capable of receiving these transmissions and displaying the multimedia data. Most of these applications are widely adopted nowadays and, at the same time the technology advances, the user are becoming more demanding about the quality of the services they use. Given the diversity of devices and networks available today, one of the big challenges of these multimedia systems is to be able to adapt the transmission to the receivers' characteristics and conditions. A suitable solution to provide this adaptation is the integration of scalable video coding with layered transmission. As the final product in these multimedia systems are the multimedia data that is presented to the user, the quality of these data will define the performace of the system and the users' satisfaction. This paper presents a study of subjective quality of scalable video sequences, coded using the scalable extension of the H.264 standard (SVC). A group of experiments was performed to measure, primarily, the efeects that the transmission instability (variations in the number of video layers received) has in the video quality and the relationship between the three scalability methods (spatial, temporal and quality) in terms of subjective quality. The decisions taken to model the tests were based on layered transmission systems that use protocols for adaptability and congestion control. To run the subjective assessments we used the ACR-HRR methodology and recommendations given by ITU-R Rec. BT.500 and ITU-T Rec. P.910. The results show that the instability modelled does not causes significant alterations on the overall video subjective quality if compared to a stable video and that the temporal scalability usually produces videos with worse quality than the spatial and quality methods, the latter being the one with the better quality. The main contributions presented in this work are the results obtained in the subjective assessments. Moreover, are also considered as contributions the methodology used throughout the entire work (including the test plan definition, the use of tools as JSVM, the test material selection and the steps taken during the assessment), some applications that were developed, the definition of future works and the specification of some problems that can also be solved with subjective quality evaluations.

In this dissertation, a fully scalable video coding system is proposed. This system achieves full temporal, resolution, and fidelity scalability by combining mesh-based motion-compensated temporal filtering, multihypothesis motion compensation, and an embedded 3D wavelet-coefficient coder. The first major contribution of this work is the introduction of the redundant-wavelet multihypothesis paradigm into motion-compensated temporal filtering, which is achieved by deploying temporal filtering in the domain of a spatially redundant wavelet transform. A regular triangle mesh is used to track motion between frames, and an affine transform between mesh triangles implements motion compensation within a lifting-based temporal transform. Experimental results reveal that the incorporation of redundant-wavelet multihypothesis into mesh-based motion-compensated temporal filtering significantly improves the rate-distortion performance of the scalable coder. The second major contribution is the introduction of a sliding-window implementation of motion-compensated temporal filtering such that video sequences of arbitrarily length may be temporally filtered using a finite-length frame buffer without suffering from severe degradation at buffer boundaries. Finally, as a third major contribution, a novel 3D coder is designed for the coding of the 3D volume of coefficients resulting from the redundant-wavelet based temporal filtering. This coder employs an explicit estimate of the probability of coefficient significance to drive a nonadaptive arithmetic coder, resulting in a simple software implementation. Additionally, the coder offers the possibility of a high degree of vectorization particularly well suited to the data-parallel capabilities of modern general-purpose processors or customized hardware. Results show that the proposed coder yields nearly the same rate-distortion performance as a more complicated coefficient coder considered to be state of the art.

With the advances in video compression and networking techniques, the last ten years have witnessed an explosive growth of video applications over the Internet. However, the service model of the current best-effort network was never engineered to handle video traffic and, as a result, video applications still suffer from varying and unpredictable network conditions, in terms of bandwidth, packet loss and delay. To address these problems, a lot of innovative techniques have been proposed and researched. Among them, scalable video coding is a promising one to cope with the dynamics of the available bandwidth and heterogeneous terminals. This work aims at improving the efficacy of scalable video transport over IP networks. In this work, we first propose an optimal interleaving scheme combined with motion-compensated fine granularity scalability video source coding and unequal loss protection schemes, under an imposed delay constraint. The network is modeled as a packet-loss channel with random delays. The motion compensation prediction, ULP allocation and the depth of the interleaver are jointly optimized based on the network status and the delay constraint. We then proceed to investigate the multiple path transport technique. A unified approach which incorporates adaptive motion compensation prediction, multiple description coding and unequal multiple path allocation, is proposed to improve both the robustness and error resilience property of the video coding and transmission system, while the delivered video quality is improved simultaneously. To analytically investigate the efficacy of error resilient transport schemes for progressively encoded sources, including unequal loss protection, best-effort and FEC transport schemes, we develop evaluation and optimization approaches for these transport schemes. In this part of the work, the network is modeled as an M/D/1/K queue, and then a comprehensive queueing analysis is provided. Armed with these results, the efficacy of these transport schemes for progressively encoded sources are investigated and compared.

With the proliferation of smart devices that require Internet connectivity anytime, anywhere, and the recent technological advances that make it possible, current networked systems will have to provide a various range of services, such as content distribution, in a wide range of settings, including wireless environments. Wireless links may experience temporary losses, however, TCP, the de facto protocol for robust unicast communications, reacts by reducing the congestion window drastically and injecting less traffic in the network. Consequently the wireless links are underutilized and the overall performance of the TCP protocol in wireless environments is poor. As content delivery (i.e. multicasting) services, such as BBC iPlayer, become popular, the network needs to support the reliable transport of the data at high rates, and with specific delay constraints. A typical approach to deliver content in a scalable way is to rely on peer-to-peer technology (used by BitTorrent, Spotify and PPLive), where users share their resources, including bandwidth, storage space, and processing power. Still, these systems suffer from the lack of incentives for resource sharing and cooperation, and this problem is exacerbated in the presence of heterogenous users, where a tit-for-tat scheme is difficult to implement. Due to the issues highlighted above, current network architectures need to be changed in order to accommodate the users¿ demands for reliable and quality communications. In other words, the emergent need for advanced modes of information transport requires revisiting and improving network components at various levels of the network stack. The innovative paradigm of network coding has been shown as a promising technique to change the design of networked systems, by providing a shift from how data flows traditionally move through the network. This shift implies that data flows are no longer kept separate, according to the ¿store-and-forward¿ model, but they are also processed and mixed in the network. By appropriately combining data by means of network coding, it is expected to obtain significant benefits in several areas of network design and architecture. In this thesis, we set out to show the benefits of including network coding into three communication paradigms, namely point-topoint communications (e.g. unicast), point-to-multipoint communications (e.g. multicast), and multipoint-to-multipoint communications (e.g. peer-to-peer networks). For the first direction, we propose a network coding-based multipath scheme and show that TCP unicast sessions are feasible in highly volatile wireless environments. For point-to-multipoint communications, we give an algorithm to optimally achieve all the rate pairs from the rate region in the case of degraded multicast over the combination network. We also propose a system for live streaming that ensures reliability and quality of service to heterogenous users, even if data transmissions occur over lossy wireless links. Finally, for multipoint-to-multipoint communications, we design a system to provide incentives for live streaming in a peer-to-peer setting, where users have subscribed to different levels of quality. Our work shows that network coding enables a reliable transport of data, even in highly volatile environments, or in delay sensitive scenarios such as live streaming, and facilitates the implementation of an efficient incentive system, even in the presence of heterogenous users. Thus, network coding can solve the challenges faced by next generation networks in order to support advanced information transport.

This research investigates scalable video communications and its applications to video streaming, conferencing and 3DTV. Scalable video coding (SVC) is a layer-based encoding scheme that provides spatial, temporal and quality scalability. Heterogeneity of the Internet and clients' operating environment necessitate the adaptation of media content to ensure a satisfactory multimedia experience. SVC's layer structure allows the extraction of partial bitstreams at reduced spatial, quality and temporal resolutions that adjust the media bitrate at a fine granularity to changes in network state. The main focus of this research work is in developing such extraction algorithms in the context of SVC. Based on a combination of metadata computations and prediction mechanisms, these algorithms evaluate the quality contribution of each layer in the SVC bitstream and make extraction decisions that are aimed at maximizing video quality while operating within the available bandwidth resources. These techniques are applied in two-way interaction and one-way streaming of 2D and 3D content. Depending on the delay tolerance of these applications, rate-distortion optimized extraction algorithms are proposed. For conferencing applications, the extraction decisions are made over single frames and frame pairs due to tight end-to-end delay constraints. The proposed extraction algorithms for 3D content streaming maximize the overall perceived 3D quality based on human stereoscopic perception. When compared to current extraction methods, the new algorithms offer better video quality at a given bitrate while performing lesser number of metadata computations in the post-encoding phase. The solutions proposed for each application achieve the recurring goal of maintaining the best possible level of end-user quality of multimedia experience in spite of network impairments.

Les applications en temps réel, telles que les systèmes de conférence multi-utilisateurs, ont des exigences de qualité de service élevées pour garantir une qualité d'expérience décente. De nos jours, la plupart de ces conférences sont effectuées sur des appareils sans fil. Ainsi, les appareils mobiles hétérogènes et la dynamique du réseau doivent être correctement gérés pour fournir une bonne qualité d’expérience. Dans cette thèse, nous proposons deux algorithmes pour construire et gérer des sessions de conférence basées sur un réseau défini par logiciel qui utilise à la fois la distribution multicast et l’adaptation de flux. Le premier algorithme configure la conférence téléphonique en créant des arborescences de multidiffusion pour chaque participant. Ensuite, il place de manière optimale les emplacements et les règles d’adaptation des flux sur le réseau afin de minimiser la consommation de bande passante. Nous avons créé deux versions de cet algorithme: le premier, basé sur les arborescences les plus courtes, minimise la latence, tandis que le second, basé sur les arborescences, minimise la consommation de bande passante. Le deuxième algorithme adapte les arborescences de multidiffusion en fonction des modifications du réseau qui se produisent pendant un appel. Il ne recalcule pas les arbres, mais ne déplace que les emplacements et les règles d’adaptation des flux. Cela nécessite un calcul très faible au niveau du contrôleur, ce qui rend notre proposition rapide et hautement réactive. Des résultats de simulation étendus confirment l'efficacité de notre solution en termes de temps de traitement et d'économies de bande passante par rapport aux systèmes de conférence existants basés sur une unité de contrôle multipoint et une multidiffusion de couche d'application. / Real-time applications, such as Multi-party conferencing systems, have strong Quality of Service requirements for ensuring a decent Quality of Experience. Nowadays, most of these conferences are performed on wireless devices. Thus, heterogeneous mobile devices and network dynamics must be properly managed to provide a good quality of experience. In this thesis, we propose two algorithms for building and maintaining conference sessions based on Software-Defined Network that uses both multicast distribution and streams adaptation. The first algorithm set up the conference call by building multicast trees for each participant. Then, it optimally places the stream adaptation locations and rules inside the network in order to minimize the bandwidth consumption. We have created two versions of this algorithm: the first one, based on the shortest path trees is minimizing the latency, while the second one, based on spanning trees is minimizing the bandwidth consumption. The second algorithm adapts the multicast trees according to the network changes occurring during a call. It does not recompute the trees, but only relocates the locations and rules of stream adaptation. It requires very low computation at the controller, thus making our proposal fast and highly reactive. Extensive simulation results confirm the efficiency of our solution in terms of processing time and bandwidth savings compared to existing conferencing systems based on a Multipoint Control Unit and Application Layer Multicast.

[EN] This thesis dissertation tackles the problem concerning provision of video streaming services over mobile wireless ad hoc networks. Such networks are characterized by their versatility and flexibility, features that make them particularly suited to be used in many scenarios. However, some limitations inherited of the wireless channel and the mobility of the nodes make difficult to guarantee certain degree of quality of service, which is a required condition to the multimedia applications. Furthermore, with the massive demand of video content, it has become very necessary for mobile ad hoc networks to have an efficient routing and quality of services mechanisms to support this traffic. This is because video streaming services require network to provide sufficient bandwidth and an upper bound in delay, jitter and loss rate. Consequently, traditional best-effort protocols are not adequate. The main contribution of this thesis is the development of a comprehensive routing protocol that has a feedback scheme in order to provide information to the application about the network conditions. This protocol has a cross-layer architecture and it incorporates three important enhancements. Firstly, a new route recovery strategy, which provides a mechanism to detect the link failures in a route and re-establish the connections taking into account the conditions of quality of service that have been established during the previous route discovery phase. Secondly, an algorithm for the estimation of the available bandwidth along the route, information that is sent to application layer in order to apply an adaptation procedure that adjusts the bit rate of the video source. This rate-adaptive strategy is performed exploiting the layered scheme of the scalable video coding. In particular, the adaptive method removes, from the scalable video stream, those layers that could not be efficiently supported by network since their bitrates exceed the available bandwidth. The third main feature of the proposed routing protocol is a gateway discovery algorithm to improve the interconnectivity between mobile ad hoc networks and infrastructure-based networks. This algorithm incorporates available bandwidth as a metric during the gateway selection and a dynamic adaptation of some operational parameters such as the size of the proactive area and the frequency of the advertisement messages. Additionally, in order to solve the lack of a software tool to simulate rate-adaptive transmission of scalable video, a new simulation framework had be implemented. This simulation tool is an open source software freely available and, thus, it represents other contribution of this thesis. The results reveal performance improvements in terms of packet delay, dropped packets and the number of link failures while a more efficient use of the available bandwidth is obtained. In terms of video transmission, the results prove that the combined use of the proposed protocol and the scalable video coding provides an efficient platform for supporting rate-adaptive video streaming over mobile ad hoc networks. / [ES] Esta tesis aborda los problemas relacionados con los servicios de video en modo streaming sobre las redes móviles ad hoc. Este tipo de redes se caracterizan por su versatilidad y flexibilidad, lo cual las hace especialmente adecuadas para ser utilizadas en diversos escenarios. Sin embargo, algunas limitaciones inherentes a los enlaces inalámbricos y a la movilidad de los nodos, hace difícil garantizar cierto nivel de calidad de servicio, lo cual es una condición necesaria para el transporte de flujos multimedia. Además, con la masiva demanda de videos desde los dispositivos móviles, hace aún más necesario asegurar un encaminamiento eficiente y un cierto nivel de calidad de servicio en las redes móviles ad hoc. Por lo tanto, los tradicionales protocolos que funcionan bajo el modelo del "mejor esfuerzo" no son adecuados. Esto se debe principalmente a que las aplicaciones multimedia necesitan que la red asegure suficiente ancho de banda y unos valores máximos de retardo, jitter, y tasa de pérdidas. La principal contribución de esta tesis es el desarrollo de un protocolo de encaminamiento que contiene un esquema de realimentación que le permite informar a la aplicación sobre las condiciones de la red. Este protocolo tiene una arquitectura cross-layer e incorpora tres importantes mejoras. Primero, una nueva estrategia de mantenimiento y recuperación de rutas que provee mecanismos para detectar los fallos de conectividad y el posterior re-establecimiento de las conexiones, teniendo en cuenta las condiciones de calidad de servicio que fueron establecidas durante la etapa inicial del descubrimiento de las rutas. Segundo, un algoritmo para la estimación del ancho de banda disponible a lo largo de la ruta, información que es enviada a la capa de aplicación para aplicar un proceso de adaptación que ajusta la tasa de envío de datos de la fuente. Esta estrategia adaptativa de la tasa aprovecha el esquema por capas de la codificación escalable de video. En particular, el método adaptativo elimina del flujo de video escalable, aquellas capas que no pueden ser transmitidas por la red ya que su tasa de bits supera el ancho de banda disponible. La tercera mejora incluida en el protocolo propuesto es un algoritmo de descubrimiento de gateways para mejorar la interconectividad entre las redes móviles ad hoc y las redes basadas en infraestructura. Dicho algoritmo utiliza el ancho de banda disponible para seleccionar el mejor gateway, así mismo, realiza una adaptación dinámica de algunos parámetros operacionales como el alcance y la frecuencia de los mensajes anuncio. Adicionalmente, ha sido desarrollada una herramienta software para simular la transmisión adaptativa de video escalable sobre redes móviles ad hoc. Esta herramienta de simulación es un software de código abierto y constituye otra contribución más de esta tesis. Los resultados muestran mejoras en el funcionamiento de las redes relacionadas con el retardo, la tasa de pérdidas de paquetes y el número de fallos en la conectividad. Simultáneamente, se obtiene un uso más eficiente del ancho de banda. En relación a la calidad del video transmitido, los resultados demuestran que la utilización del protocolo propuesto junto con la codificación de video escalable, provee un eficiente sistema para la transmisión adaptativa de video escalable sobre redes móviles ad hoc. / [CAT] Aquesta tesi aborda els problemes relacionats amb els serveis de vídeo en mode streaming sobre les xarxes mòbils ad hoc. Aquest tipus de xarxes es caracteritzen per la seva versatilitat i flexibilitat, la qual cosa les fa especialment adequades per a ser utilitzades en diversos escenaris. No obstant això, algunes limitacions inherents als enllaços sense fils i a la mobilitat dels nodes, fa difícil garantir cert nivell de qualitat de servei, cosa que és una condició necessària per al transport de fluxos multimèdia. A més, amb la massiva demanda de vídeos des dels dispositius mòbils, fa encara més necessari assegurar un encaminament eficient i un cert nivell de qualitat de servei en les xarxes mòbils ad hoc. Per tant, els tradicionals protocols que funcionen sota el model del "millor esforç" no són adequats. Això es deu principalment al fet que les aplicacions multimèdia necessiten que la xarxa asseguri suficient ample de banda i uns valors màxims de retard, jitter, i taxa de pèrdues. La principal contribució d'aquesta tesi és el desenvolupament d'un protocol d'encaminament que conté un esquema de realimentació que li permet informar l'aplicació sobre les condicions de la xarxa. Aquest protocol té una arquitectura cross-layer i incorpora tres importants millores. Primer, una nova estratègia de manteniment i recuperació de rutes que proveeix mecanismes per detectar les fallades de connectivitat i el posterior re-establiment de les connexions, tenint en compte les condicions de qualitat de servei que van ser establertes durant l'etapa inicial del descobriment de les rutes. Segon, un algoritme per a l'estimació de l'ample de banda disponible al llarg de la ruta, informació que és enviada a la capa d'aplicació per aplicar un procés d'adaptació que ajusta la taxa d'enviament de dades de la font. Aquesta estratègia adaptativa de la taxa aprofita l'esquema per capes de la codificació escalable de vídeo. En particular, el mètode adaptatiu elimina del flux de vídeo escalable aquelles capes que no poden ser transmeses per la xarxa ja que la seva taxa de bits supera l'ample de banda disponible. La tercera millora inclosa en el protocol proposat és un algoritme de descobriment de gateways per millorar la interconnectivitat entre les xarxes mòbils ad hoc i les xarxes basades en infraestructura. Aquest algoritme utilitza l'ample de banda disponible per seleccionar el millor gateway, així mateix, realitza una adaptació dinàmica d'alguns paràmetres operacionals com l'abast i la freqüència dels missatges anunci. Addicionalment, ha estat desenvolupada una eina programari per a simular la transmissió adaptativa de vídeo escalable sobre xarxes mòbils ad hoc. Aquesta eina de simulació és un programari de codi obert i constitueix una altra contribució més d'aquesta tesi. Els resultats mostren millores en el funcionament de les xarxes relacionades amb el retard, la taxa de pèrdues de paquets i el nombre de fallades en la connectivitat. Simultàniament, se n'obté un ús més eficient de l'ample de banda. En relació a la qualitat del vídeo transmès, els resultats demostren que la utilització del protocol proposat juntament amb la codificació de vídeo escalable, proveeix un eficient sistema per a la transmissió adaptativa de vídeo escalable sobre xarxes mòbils ad hoc. / Castellanos Hernández, WE. (2015). Quality of Service Routing and Mechanisms for Improving Video Streaming over Mobile Wireless Ad hoc Networks [Tesis doctoral]. Editorial Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/53238 / TESIS

This thesis is concerned with maximizing the coding efficiency, random accessibility and visual performance of scalable compressed video. The unifying theme behind this work is the use of finely embedded localized coding structures, which govern the extent to which these goals may be jointly achieved. The first part focuses on scalable volumetric image compression. We investigate 3D transform and coding techniques which exploit inter-slice statistical redundancies without compromising slice accessibility. Our study shows that the motion-compensated temporal discrete wavelet transform (MC-TDWT) practically achieves an upper bound to the compression efficiency of slice transforms. From a video coding perspective, we find that most of the coding gain is attributed to offsetting the learning penalty in adaptive arithmetic coding through 3D code-block extension, rather than inter-frame context modelling. The second aspect of this thesis examines random accessibility. Accessibility refers to the ease with which a region of interest is accessed (subband samples needed for reconstruction are retrieved) from a compressed video bitstream, subject to spatiotemporal code-block constraints. We investigate the fundamental implications of motion compensation for random access efficiency and the compression performance of scalable interactive video. We demonstrate that inclusion of motion compensation operators within the lifting steps of a temporal subband transform incurs a random access penalty which depends on the characteristics of the motion field. The final aspect of this thesis aims to minimize the perceptual impact of visible distortion in scalable reconstructed video. We present a visual optimization strategy based on distortion scaling which raises the distortion-length slope of perceptually significant samples. This alters the codestream embedding order during post-compression rate-distortion optimization, thus allowing visually sensitive sites to be encoded with higher fidelity at a given bit-rate. For visual sensitivity analysis, we propose a contrast perception model that incorporates an adaptive masking slope. This versatile feature provides a context which models perceptual significance. It enables scene structures that otherwise suffer significant degradation to be preserved at lower bit-rates. The novelty in our approach derives from a set of "perceptual mappings" which account for quantization noise shaping effects induced by motion-compensated temporal synthesis. The proposed technique reduces wavelet compression artefacts and improves the perceptual quality of video.

Multimedia services and applications became the driving force in the development and widespread deployment of wireless broadband access technologies and high speed local area networks. Mobile phone service providers are offering wide range of multimedia applications over high speed wireless data networks. People can watch live TV, stream on-demand video clips and place videotelephony calls using multimedia capable mobile devices. Mobile devices will soon support capturing and displaying high definition video. Similar evolution is also occurring in the local area domain. The video receiver or storage devices were conventionally connected to display devices using cables. By using wireless local area networking (WLAN) technologies, convenient and cable-free connectivity can be achieved. Media over wireless home networks prevents the cable mess and provides mobility to portable TVs. However, there still exit challenges for improving the quality-of-service (QoS) of multimedia applications. Conventional service architectures, network structures and protocols lack to provide a robust distribution medium since most of them are not designed considering the high data rate and real-time transmission requirements of digital video. In this thesis the challenges of wireless video streaming are addressed in two main categories. Streaming protocol level issues constitute the first category. We will refer to the collection of network protocols that enable transmitting digital compressed video from a source to a receiver as the streaming protocol. The objective of streaming protocol solutions is the high quality video transfer between two networked devices. Novel application-layer video bit-rate adaptation methods are designed for handling short- and long-term bandwidth variations of the wireless local area network (WLAN) links. Both transrating and scalable video coding techniques are used to generate video bit-rate flexibility. Another contribution of this thesis study is an error control method that dynamically adjusts the forward error correction (FEC) rate based on channel bit-error rate (BER) estimation and video coding structure. The second category is the streaming service level issues, which generally surface in large scale systems. Service system solutions target to achieve system scalability and provide low cost / high quality service to consumers. Peer-to-peer assisted video streaming technologies are developed to reduce the load of video servers. Novel video file segment caching strategies are proposed for more efficient peer-to-peer collaboration.

Abstract This dissertation studies the impact of heterogeneous wireless networks on the design and implementation of mobile video streaming services. The aim is to enable Quality of Service (QoS) sensitive video streaming services to take full advantage of the access diversity of heterogeneous networks in order to optimize their operation in terms of quality and efficiency of network resource usage. This nevertheless requires support beyond the layered communication architecture of today’s Internet. The thesis proposes an architecture for end-to-end cross-layer signaling and control for video streaming systems. The architecture supports extensive context informa- tion transfer in heterogeneous networks; thus, enabling the efficient management of video stream adaptation and user terminal mobility in a diverse and dynamic network environment. This thesis also studies and proposes cross-layer enhancements for adaptive video streaming and mobility management functions enabled by the cross- layer architecture. These include cross-layer video adaptation, congestion-triggered handovers, and concurrent utilization of multiple access networks in the video stream transport. For the video adaptation and multipath transmission, the flexible adaptation and transmission capabilities of the novel scalable video coding technology are used. Regarding the mobility management, the proposed solutions essentially enhance the handover decision-making of the Mobile IP protocol to better support QoS-sensitive video streaming. Finally, the thesis takes a holistic view on the application adaptation and mobility management, and proposes a solution for coordinated control of these two operations in order to achieve end-to-end optimization. The resulting mobile video streaming system architecture and the cross-layer control algorithms are evaluated using network simulations and real prototypes. Based on the results, the proposed mechanisms can be seen to be viable solutions for improving video streaming performance in heterogeneous wireless networks. They require changes in the communication end-points and the access network but support gradual deployment. This allows service providers and operators to select only a subset of the proposed mechanisms for implementation. The results also support the need for cross-layer signaling and control in facilitating efficient management and utilization of heterogeneous wireless networks. / Tiivistelmä Väitöskirja tutkii langattomien ja heterogeenisen verkkoympäristöjen vaikutusta erityisesti mobiilikäyttöön suunnattuihin suoratoistovideopalveluihin (streaming). Työn tavoitteena on löytää keino optimoida verkkoyhteyden palvelunlaadulle (QoS) herkän suoratoistovideon toiminta sekä videopalvelun laadun että verkon tiedonsiirtokapasiteetin käytön osalta. Tämä tapahtuu mahdollistamalla heterogeenisten verkkojen tehokas käyttö suoratoistovideopalvelujen tapauksessa. Tavoitellut parannukset vaativat kuitenkin muutoksia nykyiseen kerroksittaiseen Internet-arkkitehtuuriin. Väitöskirjassa esitetään arkkitehtuuri protokollakerrosten välisen tiedon (cross-layer) välitykseen ja hyödyntämiseen suoratoistovideopalvelujen tiedonsiirron kontrolloinnissa. Arkkitehtuuria voidaan käyttää laaja-alaiseen kontekstitiedon välitykseen tietoverkoissa, mikä mahdollistaa tehokkaan videopalvelun adaptoinnin ja päätelaitteen liikkuvuudenhallinnan heterogeenisissa verkoissa, joissa palvelunlaatu vaihtelee. Väitöskirja myös ehdottaa erilaisia ratkaisuja videopalvelun adaptoinnin ja tiedonsiirron parantamiseksi arkkitehtuuria hyödyntämällä. Näihin lukeutuvat usealle protokollakerrokselle toteutettu videon adaptointi, verkkoyhteyden ruuhkautumiseen reagoiva yhteydensiirto sekä usean verkkoyhteyden samanaikainen käyttö videopalvelun tiedonsiirrossa. Videon adaptoinnissa ja siirrossa hyödynnetään uutta skaalautuvaa videonkoodausteknologiaa, joka mahdollistaa vaaditun, joustavan videobittivirran muokkauksen. Liikkuvuudenhallinnan osalta työssä keskitytään pääosin kehittämään Mobile IP -protokollan päätöksentekoa suoratoistovideopalvelujen tapauksessa. Lopuksi väitöskirjassa esitetään kokonaisvaltainen ja koordinoitu ratkaisu videopalvelun adaptoinnin sekä päätelaitteen liikkuvuuden hallintaan päästä päähän -optimoinnin saavuttamiseksi. Tuloksena esitetyt järjestelmäarkkitehtuuri ja protokollakerrosten välistä tietoa käyttävät hallinta-algoritmit evaluoitiin simulaatioiden ja oikeiden prototyyppien avulla. Tulokset osoittavat, että ehdotettuja menetelmiä voidaan käyttää parantamaan suoratoistovideopalvelujen suorituskykyä heterogeenisissa verkoissa. Ratkaisut vaativat muutoksia verkko- ja palveluarkkitehtuureihin, mutta niiden asteittainen tai osittainen käyttöönotto on mahdollista. Tulokset osoittavat myös protokollakerrosten välisen tiedon tarpeellisuuden langattomien ja heterogeenisten verkkojen tehokkaassa käytössä.