The IP Multimedia Subsystem (IMS) has gone from just a step in the evolution of the GSM cellular architecture control core, to being the de-facto framework for Next Generation Network (NGN) implementations and deployments by operators world-wide, not only cellular mobile communications operators, but also fixed line, cable television, and alternative operators. With this transition from standards documents to the real world, engineers in these new multimedia communications companies need to face the task of making these new networks secure against threats and real attacks that were not a part of the previous generation of networks. We present the IMS and other competing frameworks, we analyze the security issues, we present the topic of Security Patterns, we introduce several new patterns, including the basis for a Generic Network pattern, and we apply these concepts to designing a security architecture for a fictitious 3G operator using IMS for the control core. / by Jose M. Ortiz-Villajos. / Thesis (M.S.C.S.)--Florida Atlantic University, 2009. / Includes bibliography. / Electronic reproduction. Boca Raton, Fla., 2009. Mode of access: World Wide Web.

In a wireless and mobile communication paradigm, distribution and sharing of video content often occurs over unfriendly network environments constrained by lack of sufficient bandwidth, and prone to jitter, delay and packet losses. The research presented in this thesis proposed an assortment of application-specific optimizations designed to enable high-quality video communication over bandwidth constrained and unreliable channels. This assortment of solutions, termed herein as the Application Specific Video Coding and Delivery (ASVCD) toolkit, comprises of content and network adaptive approaches such as Region of Interest (ROI) video coding, Multiple Representation Coding (MRC), and Multiple Representation Coding of the Region of Interest (ROI + MRC). Thus, the effectiveness of ROI based video-coding in facilitating diagnostically lossless delivery of surgical videos over very low bandwidth channels was studied in this thesis. Furthermore, to facilitate error resilient video delivery over channels prone to burst losses and signal loss intervals, the MRC scheme was presented in this thesis. Finally, the thesis proposed a scheme for unequal protection of the ROI in the video by using the MRC scheme to effectively enable a distance learning application. To summarize, the ASVCD toolkit contributed in enabling high-quality video communications applications to become seamless and pervasive.

Video frame rate up-conversion is an important issue for multimedia systems in achieving better video quality and motion portrayal. Motion-compensated methods offer better quality interpolated frames since the interpolation is performed along the motion trajectory. In addition, computational complexity, regularity, and memory bandwidth are important for a real-time implementation. Motion-compensated frame rate up-conversion (MC-FRC) is composed of two main parts: motion estimation (ME) and motion-compensated frame interpolation (MCFI). Since ME is an essential part of MC-FRC, a new fast motion estimation (FME) algorithm capable of producing sub-sample motion vectors at low computational-complexity has been developed. Unlike existing FME algorithms, the developed algorithm considers the low complexity sub-sample accuracy in designing the search pattern for FME. The developed FME algorithm is designed in such a way that the block distortion measure (BDM) is modeled as a parametric surface in the vicinity of the integer-sample motion vector; this modeling enables low computational-complexity sub-sample motion estimation without pixel interpolation. MC-FRC needs more accurate motion trajectories for better video quality; hence, a novel true-motion estimation (TME) algorithm targeting to track the projected object motion has been developed for video processing applications, such as motion-compensated frame interpolation (MCFI), deinterlacing, and denoising. Developed TME algorithm considers not only the computational complexity and regularity but also memory bandwidth. TME is obtained by imposing implicit and explicit smoothness constraints on block matching algorithm (BMA). In addition, it employs a novel adaptive clustering algorithm to keep the low-complexity at reasonable levels yet enable exploiting more spatiotemporal neighbors. To produce better quality interpolated frames, dense motion field at the interpolation instants are obtained for both forward and backward motion vectors (MVs); then, bidirectional motion compensation using forward and backward MVs is applied by mixing both elegantly.

Wireless multimedia sensor networks (WMSNs) are networks of interconnected devices that allow retrieving video and audio streams, still images, and scalar data from the environment. In a densely deployed WMSN, there exists correlation among the observations of camera sensors with overlapped coverage areas, which introduces substantial data redundancy in the network. In this dissertation, efficient communication schemes are designed for WMSNs by leveraging the correlation of visual information observed by camera sensors. First, a spatial correlation model is developed to estimate the correlation of visual information and the joint entropy of multiple correlated camera sensors. The compression performance of correlated visual information is then studied. An entropy-based divergence measure is proposed to predict the compression efficiency of performing joint coding on the images from correlated cameras. Based on the predicted compression efficiency, a clustered coding technique is proposed that maximizes the overall compression gain of the visual information gathered in WMSNs. The correlation of visual information is then utilized to design a network scheduling scheme to maximize the lifetime of WMSNs. Furthermore, as many WMSN applications require QoS support, a correlation-aware QoS routing algorithm is introduced that can efficiently deliver visual information under QoS constraints. Evaluation results show that, by utilizing the correlation of visual information in the communication process, the energy efficiency and networking performance of WMSNs could be improved significantly.

The objective of this thesis is to formulate a generic dictionary learning method with the guiding principle that states: Efficient representations lead to efficient estimations. The fundamental idea behind using transforms or dictionaries for signal representation is to exploit the regularity within data samples such that the redundancy of the representation is minimized subject to a level of fidelity. This observation translates to rate-distortion cost in compression literature, where a transform that has the lowest rate-distortion cost provides a more efficient representation than the others. In our work, rather than using as an analysis tool, the rate-distortion cost is utilized to improve the efficiency of transforms. For this, an iterative optimization method is proposed, which seeks an orthonormal transform that reduces the expected value of rate-distortion cost of an ensemble of data. Due to the generic nature of the new optimization method, one can design a set of orthonormal transforms either in the original signal domain or on the top of a transform-domain representation. To test this claim, several image codecs are designed, which use block-, lapped- and wavelet-transform structures. Significant increases in compression performances are observed compared to original methods. An extension of the proposed optimization method for video coding gave us state-of-the-art compression results with separable transforms. Also using the robust statistics, an explanation to the superiority of new design over other learning-based methods such as Karhunen-Loeve transform is provided. Finally, the new optimization method and the minimization of the "oracle" risk of diagonal estimators in signal estimation is shown to be equal. With the design of new diagonal estimators and the risk-minimization-based adaptation, a new image denoising algorithm is proposed. While these diagonal estimators denoise local image patches, by formulation the optimal fusion of overlapping local denoised estimates, the new denoising algorithm is scaled to operate on large images. In our experiments, the state-of-the-art results for transform-domain denoising are achieved.

xi, 284 p. A print copy of this thesis is available through the UO Libraries. Search the library catalog for the location and call number. / My dissertation delves into the recent articulation of popular nationalism in Mainland China, with particular emphasis on the changes that globalization and transnationalism have brought about to the representation of the Chinese nation in sentimental terms. Complementing the rich existing literature of Chinese nationalism that focuses mainly on the pre-1949 period, my study explores the less-treaded contemporary era characterized by the new historical condition of postsocialism, which features a residual of the socialist past as well as its reinvention under new overwhelming trends of globalization. Postsocialism and its consequences-the deepening of a neoliberalist economic refonn, the state-intellectual promotion of cultural economy, the emergence of a dominant consumer culture, etc.-have produced new issues existing scholarship on Chinese nationalism has yet to address. One such issue is how the paradoxical entity of the "nation" in time and space has been fragmented by the accretion of diversified voices from a wide spectrum of Chinese society. In postsocialist China, the agents imagining the nation include not only regulars like the state and intellectuals, but also new players like mass-media elites and netizens (wangmin). I argue that these voices of different social forces that break up the hegemony of the state in representing the nation-the result of which being not that the state is excluded from this enterprise but that it now tells only part of the story-become expressed as modes of national sentiments (minzu qinggan) when the nation is imagined under the historical condition of postsocialism. My study then explores in detail the fashioning and refashioning of contemporary Chinese subjectivity, as it relates through the joining of national sentiments to the literal and figurative body of the nation and the social power structure, by analyzing these specific voices in a broad range of popular texts from TV, film, and the Internet. The detailed examination includes four chapters dealing with specific modes of national sentiments articulated by the intellectuals, the state, the mass-media elites, and the netizens, respectively. / Committee in charge: Tze-lan Sang, Co-Chairperson, East Asian Languages & Literature; David Leiwei Li, Co-Chairperson, English; Maram Epstein, Member, East Asian Languages & Literature; Bryna Goodman, Outside Member, History

The Internet Protocol Multimedia Subsystem (IMS) is a component of the 3G mobile network that has been specified by standards development organisations such as the 3GPP (3rd Generation Partnership Project) and ETSI (European Telecommunication Standards Institute). IMS seeks to guarantee that the telecommunication network of the future provides subscribers with seamless access to services across disparate networks. In order to achieve this, it defines a service architecture that hosts application servers that provide subscribers with value added services. Typically, an application server bundles all the functionality it needs to execute the services it delivers, however this view is currently being challenged. It is now thought that services should be synthesised from simple building blocks called service capabilities. This decomposition would facilitate the re-use of service capabilities across multiple services and would support the creation of new services that could not have originally been conceived. The shift from monolithic services to those built from service capabilities poses a challenge to the current service model in IMS. To accommodate this, the 3GPP has defined an entity known as a service capability interaction manager (SCIM) that would be responsible for managing the interactions between service capabilities in order to realise complex services. Some of these interactions could potentially lead to undesirable results, which the SCIM must work to avoid. As an added requirement, it is believed that the network should allow policies to be applied to network services which the SCIM should be responsible for enforcing. At the time of writing, the functional and structural architecture of the SCIM has not yet been standardised. This thesis explores the current serv ice architecture of the IMS in detail. Proposals that address the structure and functions of the SCIM are carefully compared and contrasted. This investigation leads to the presentation of key aspects of the SCIM, and provides solutions that explain how it should interact with service capabilities, manage undesirable interactions and factor user and network operator policies into its execution model. A modified design of the IMS service layer that embeds the SCIM is subsequently presented and described. The design uses existing IMS protocols and requires no change in the behaviour of the standard IMS entities. In order to develop a testbed for experimental verification of the design, the identification of suitable software platforms was required. This thesis presents some of the most popular platforms currently used by developers such as the Open IMS Core and OpenSER, as well as an open source, Java-based, multimedia communication platform called Mobicents. As a precursor to the development of the SCIM, a converged multimedia service is presented that describes how a video streaming application that is leveraged by a web portal was implemented for an IMS testbed using Mobicents components. The Mobicents SIP Servlets container was subsequently used to model an initial prototype of the SCIM, using a mUlti-component telephony service to illustrate the proposed service execution model. The design focuses on SIP-based services only, but should also work for other types of IMS application servers as well.

With the rapid adoption of multimedia network technologies, a number of companies and standards bodies are introducing technologies that enhance user experience in networked multimedia environments. These technologies focus on device discovery, connection management, control, and monitoring. This study focused on control and monitoring. Multimedia networks make it possible for devices that are part of the same network to reside in different physical locations. These devices contain parameters that are used to control particular features, such as speaker volume, bass, amplifier gain, and video resolution. It is often necessary for changes in one parameter to affect other parameters, such as a synchronised change between volume and bass parameters, or collective control of multiple parameters. Thus, relationships are required between the parameters. In addition, some devices contain parameters, such as voltage, temperature, and audio level, that require constant monitoring to enable corrective action when thresholds are exceeded. Therefore, a mechanism for monitoring networked devices is required. This thesis proposes relationships that are essential for the proper functioning of a multimedia network and that should, therefore, be incorporated in standard form into a protocol, such that all devices can depend on them. Implementation mechanisms for these relationships were created. Parameter grouping and monitoring capabilities within mixing console implementations and existing control protocols were reviewed. A number of requirements for parameter grouping and monitoring were derived from this review. These requirements include a formal classification of relationship types, the ability to create relationships between parameters with different underlying value units, the ability to create relationships between parameters residing on different devices on a network, and the use of an event-driven mechanism for parameter monitoring. These requirements were the criteria used to govern the implementation mechanisms that were created as part of this study. Parameter grouping and monitoring mechanisms were implemented for the XFN protocol. The mechanisms implemented fulfil the requirements derived from the review of capabilities of mixing consoles and existing control protocols. The formal classification of relationship types was implemented within XFN parameters using lists that keep track of the relationships between each XFN parameter and other XFN parameters that reside on the same device or on other devices on the network. A common value unit, known as the global unit, was defined for use as the value format within value update messages between XFN parameters that have relationships. Mapping tables were used to translate the global unit values to application-specific (universal) units, such as decibels (dB). A mechanism for bulk parameter retrieval within the XFN protocol was augmented to produce an event-driven mechanism for parameter monitoring. These implementation mechanisms were applied to an XFN-protocol-compliant graphical control application to demonstrate their usage within an end user context. At the time of this study, the XFN protocol was undergoing standardisation within the Audio Engineering Society. The AES-64 standard has now been approved. Most of the implementation mechanisms resulting from this study have been incorporated into this standard.

In the Morder-Client Food Service, a waiter/waitresses orders meals via a wireless handheld device. By reading the order items listed on the screen of a personal computer which is in a kitchen, the cook can make the correct dishes. This system can improve service quality and efficiency.

[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