Information-theoretic arguments focus on modeling the reliability of information transmission, assuming availability of infinite data at sources, thus ignoring randomness in message generation times at the respective sources. However, in information transport networks, not only is reliable transmission important, but also stability, i.e., finiteness of mean delay in- curred by messages from the time of generation to the time of successful reception. Usually, delay analysis is done separately using queueing-theoretic arguments, whereas reliable information transmission is studied using information theory. In this thesis, we investigate these two important aspects of data communication jointly by suitably combining models from these two fields. In particular, we model scheduled communication of messages , that arrive in a random process, (i) over multiaccess channels, with either independent decoding or joint decoding, and (ii) over degraded broadcast channels. The scheduling policies proposed permit up to a certain maximum number of messages for simultaneous transmission. In the first part of the thesis, we develop a multi-class discrete-time processor-sharing queueing model, and then investigate the stability of this queue. In particular, we model the queue by a discrete-time Markov chain defined on a countable state space, and then establish (i) a sufficient condition for c-regularity of the chain, and hence positive recurrence and finiteness of stationary mean of the function c of the state, and (ii) a sufficient condition for transience of the chain. These stability results form the basis for the conclusions drawn in the thesis. The second part of the thesis is on multiaccess communication with random message arrivals. In the context of independent decoding, we assume that messages can be classified into a fixed number of classes, each of which specifies a combination of received signal power, message length, and target probability of decoding error. Each message is encoded independently and decoded independently. In the context of joint decoding, we assume that messages can be classified into a fixed number of classes, each of which specifies a message length, and for each of which there is a message queue. From each queue, some number of messages are encoded jointly, and received at a signal power corresponding to the queue. The messages are decoded jointly across all queues with a target probability of joint decoding error. For both independent decoding and joint decoding, we derive respective discrete- time multiclass processor-sharing queueing models assuming the corresponding information-theoretic models for the underlying communication process. Then, for both the decoding schemes, we (i) derive respective outer bounds to the stability region of message arrival rate vectors achievable by the class of stationary scheduling policies, (ii) show for any mes- sage arrival rate vector that satisfies the outer bound, that there exists a stationary “state-independent” policy that results in a stable system for the corresponding message arrival process, and (iii) show that the stability region of information arrival rate vectors, in the limit of large message lengths, equals an appropriate information-theoretic capacity region for independent decoding, and equals the information-theoretic capacity region for joint de-coding. For independent decoding, we identify a class of stationary scheduling policies, for which we show that the stability region in the limit of large maximum number of simultane-ous transmissions is independent of the received signal powers, and each of which achieves a spectral efficiency of 1 nat/s/Hz in the limit of large message lengths. In the third and last part of the thesis, we show that the queueing model developed for multiaccess channels with joint decoding can be used to model communication over degraded broadcast channels, with superposition encoding and successive decoding across all queues. We then show respective results (i), (ii), and (iii), stated above.

Τη σημερινή εποχή γινόμαστε καθημερινά μάρτυρες μίας ταχέως αναπτυσσόμενης αγοράς, που δεν είναι άλλη από αυτή των κινητών πολυμεσικών εφαρμογών, όπως του Mobile TV και του Mobile Streaming. Υπηρεσίες όπως αυτές έχουν ή αναμένεται να έχουν υψηλή διείσδυση στη βιομηχανία της κινητής πολυμεσικής επικοινωνίας. Για να ικανοποιηθούν οι απαιτήσεις αυτών των υπηρεσιών για υψηλές ταχύτητες μετάδοσης, ο οργανισμός 3rd Generation Partnership Project (3GPP) ανέπτυξε το Long Term Evolution Advanced (LTE-A), μία τεχνολογία η οποία αποτελεί την εξέλιξη των κινητών τηλεπικοινωνιακών τεχνολογιών 3ης γενιάς. Το LTE-A χρησιμοποιεί την τεχνολογία Orthogonal Frequency Division Multiple Access (OFDMA). Η συγκεκριμένη τεχνολογία μπορεί να προσφέρει νέες υψηλής χωρητικότητας ευρυζωνικές εφαρμογές και υπηρεσίες, ενώ παρέχει αποτελεσματική, από πλευράς κόστους, καθολική κάλυψη. Επιπλέον, ο οργανισμός 3GPP εισήγαγε την τεχνολογία Multimedia Broadcast / Multicast Service (MBMS), ως μέσο πανεκπομπής και πολυεκπομπής πληροφοριών στους χρήστες κινητών, με το Mobile TV να είναι η κύρια υπηρεσία που παρέχεται. Η υποδομή του LTE-A προσφέρει στο MBMS την επιλογή να χρησιμοποιήσει ένα uplink κανάλι για την αλληλεπίδραση μεταξύ της υπηρεσίας και του χρήστη, η οποία στα συνήθη δίκτυα πανεκπομπής δεν είναι απλό θέμα. Στο πλαίσιο των LTE-A συστημάτων, το MBMS έχει εξελιχθεί σε e-MBMS (το "e-" αντιστοιχεί στη λέξη evolved, δηλαδή εξελιγμένο). Αυτό θα επιτευχθεί μέσα από την αυξημένη απόδοση της ασύρματης διεπαφής που περιλαμβάνει μία νέα τεχνολογία μετάδοσης που ονομάζεται MBMS over Single Frequency Network (MBSFN). Κατά τη λειτουργία του MBSFN, τα MBMS δεδομένα μεταδίδονται ταυτόχρονα μέσω του αέρα από πολλαπλά κελιά τα οποία είναι αυστηρά χρονο-συγχρονισμένα. Το σύνολο των κελιών που λαμβάνουν αυτά τα δεδομένα, καλείται MBSFN περιοχή. Δεδομένου ότι οι MBSFN μεταδόσεις ενισχύουν σημαντικά το λόγο σήματος προς παρεμβολή και θόρυβο, μπορούν να οδηγήσουν σε σημαντικές βελτιώσεις στη φασματική απόδοση σε σύγκριση με την πολυεκπομπή μέσω των συστημάτων 3ης γενιάς. Αυτό είναι εξαιρετικά επωφελές στα όρια των κελιών, όπου οι μεταδόσεις (που στα συστήματα 3ης γενιάς, όπως το Universal Mobile Telecommunications System - UMTS, θεωρούνται ως παρεμβολή) μεταφράζονται σε χρήσιμη ενέργεια σήματος και ως εκ τούτου η ισχύς του λαμβανόμενου σήματος είναι αυξημένη, ενώ την ίδια στιγμή η ισχύς παρεμβολής μειώνεται σε μεγάλο βαθμό. Για να αξιοποιηθούν πλήρως τα πλεονεκτήματα της τεχνολογίας MBSFN και να βελτιωθεί η φασματική απόδοση, θα πρέπει να επιλεχθεί με προσοχή το σχήμα διαμόρφωσης και κωδικοποίησης για τη μετάδοση των δεδομένων. Η σχέση μεταξύ της απόδοσης του MBSFN και της επιλογής του σχήματος διαμόρφωσης και κωδικοποίησης έχει μελετηθεί διεξοδικά σε προηγούμενες ερευνητικές εργασίες. Ωστόσο οι περισσότερες (αν όχι όλες) από τις εργασίες αυτές επικεντρώνονται μόνο στην πλευρά των χρηστών και ως εκ τούτου δεν μπορούν να θεωρηθούν πλήρεις. Μερικές φορές ο στόχος του παρόχου μπορεί να είναι η μεγιστοποίηση της φασματικής απόδοσης σε όλους τους χρήστες της τοπολογίας ή η παροχή της υπηρεσίας σε όλους τους χρήστες ανεξάρτητα από τις συνθήκες που βιώνουν. Επίσης, οι περισσότερες από αυτές τις εργασίες καθορίζουν το σχήμα διαμόρφωσης και κωδικοποίησης κατά τις MBSFN μεταδόσεις εξετάζοντας μόνο την περίπτωση της μετάδοσης από ένα πομπό σε ένα δέκτη και δεν εξετάζουν τα οφέλη που μπορούν να προσφέρουν οι τεχνικές Multiple Input Multiple Output (MIMO) στη συνολική απόδοση του συστήματος. Ο στόχος της παρούσας εργασίας είναι να επεκτείνει τις προηγούμενες ερευνητικές εργασίες και, επιπλέον, να προτείνει μια λύση στο πρόβλημα της επιλογής του σχήματος διαμόρφωσης και κωδικοποίησης. Προς την κατεύθυνση αυτή, αναλύουμε πρώτα μία διαδικασία τριών βημάτων η οποία επιλέγει το σχήμα διαμόρφωσης και κωδικοποίησης και υπολογίζει τη φασματική απόδοση στην περίπτωση ενός μόνο χρήστη. Στη συνέχεια, ακολουθεί η γενίκευση της υπόθεσης ενός χρήστη και προτείνονται τρεις προσεγγίσεις που επιλέγουν το σχήμα κωδικοποίησης για την μετάδοση των MBSFN δεδομένων σε σενάρια πολλαπλών χρηστών. Οι προσεγγίσεις αξιολογούνται για τρεις διαφορετικούς τρόπους μετάδοσης, έτσι ώστε να εξεταστεί η επίδραση των τεχνικών MIMO στην επιλογή σχήματος διαμόρφωσης για διαφορετικές κατανομές χρηστών. Τα αποτελέσματα της αξιολόγησης δείχνουν ότι, ανάλογα με το στόχο που έχει θέσει ο πάροχος (π.χ. μεγιστοποίηση της φασματικής απόδοσης ή επίτευξη μίας συγκεκριμένης τιμής φασματικής απόδοσης) κάθε προσέγγιση θα μπορούσε να οδηγήσει σε βελτιωμένη απόδοση. / Today we are witnesses of a rapidly increasing market for mobile multimedia applications, such as Mobile TV and Mobile Streaming. Services like these have or are expected to have high penetration in the mobile multimedia communications industry. In order to confront such high requirements for services that demand higher data rates, the 3rd Generation Partnership Project (3GPP) developed the Long Term Evolution Advanced (LTE-A) technology which constitutes the evolution of the 3rd Generation (3G) mobile telecommunications technologies. LTE-A utilizes Orthogonal Frequency Division Multiple Access (OFDMA). This radio technology is optimized to enhance networks by enabling new high capacity mobile broadband applications and services, while providing cost efficient ubiquitous mobile coverage. In addition, 3GPP has introduced the Multimedia Broadcast/Multicast Service (MBMS) as a means to broadcast and multicast information to mobile users, with Mobile TV being the main service offered. LTE-A infrastructure offers to MBMS an option to use an uplink channel for interaction between the service and the user, which is not a straightforward issue in usual broadcast networks. In the context of LTE-A systems, the MBMS will evolve into e-MBMS (“e-” stands for evolved). This will be achieved through the increased performance of the air interface that will include a new transmission scheme called MBMS over Single Frequency Network (MBSFN). In MBSFN operation, MBMS data are transmitted simultaneously over the air from multiple tightly time-synchronized cells. A group of those cells, which are targeted to receive these data, is called MBSFN area. Since the MBSFN transmission greatly enhances the Signal to Interference plus Noise Ratio (SINR), the MBSFN transmission mode leads to significant improvements in Spectral Efficiency (SE) in comparison to multicasting over 3G systems. This is extremely beneficial at the cell edge, where transmissions (which in 3G systems, like Universal Mobile Telecommunications System - UMTS, are considered as inter-cell interference) are translated into useful signal energy and hence the received signal strength is increased, while at the same time the interference power is largely reduced. In order to fully exploit the benefits of MBSFN and to improve its performance in terms of SE, the Modulation and Coding Scheme (MCS) for the transmission of the data should be carefully selected. The relationship between MBSFN performance and MCS selection has been thoroughly studied in previous research works; however most (if not all) of these works focus only on the users’ side and therefore may not be sufficient. Sometimes the operator’s goal may be the maximization of the SE over all users of the topology or the provision of the service to all the users irrespectively of the conditions that they experience. In addition, most of these works determine the MCS scheme for MBSFN considering only the case of single antenna transmissions and they do not examine the benefits that Multiple Input Multiple Output (MIMO) transmissions may offer on the overall performance. The goal of this thesis is to extend the previous research works and, furthermore, to tackle the problems addressed. To this direction, we first analyze a 3-step procedure that selects the MCS and calculates the SE in the case of a single user. Then, we generalize the single-user case and we propose three approaches that select the MCS for the delivery of the MBSFN data in multiple-users scenarios. The approaches are evaluated for three different transmission modes, so as to examine the impact of multiple antennas techniques on the MCS selection, and for different users’ distributions. The evaluation results indicate that depending on the target that the operator may set (i.e. SE maximization or achievement of a specific SE) each approach could lead to improved performance.

許多學者認為開放市場、降低管制是提升電視產業競爭力的不二法門，然而，國家介入與電視產業績效之間的實證研究仍相當缺乏，因此本研究的目的旨在討論國家干涉程度和節目產製能力間的關係。本研究採用歷史制度途徑的分析，解釋同樣被歸類為發展型國家的台灣與韓國，為何在1997年金融風暴後選擇不同的轉型路徑。研究結果顯示，台灣電視產業因國家干涉程度低，放任市場過度競爭而導致產製能力下滑，另一方面，韓國電視產業則受惠於國家高度干預，因寡佔結構而有效維持產製能力和經營績效。總結而言，本研究證明韓國在發展型國家的基礎上成功轉型升級，而台灣則是揚棄原本的發展模式，向新自由主義靠近。 / Many researchers have often suggested that the best way to improve broadcast industry growth is always through deregulation and opening up to competition. However, research which has empirically documented the link between state intervention in market and performance of broadcast industry is scant. Therefore, the aim of this research attempts to explore how state intervention and program productivity are related. Historical institutionalism has been used to explain the different transformation paths that Taiwan and Korean, the typical examples of the developmental states, have taken after financial crisis in 1997. Results of this study showed competitive market structure shaped by weak state intervention had contributed to productivity decline in Taiwan. And, strong state intervention in Korea did lead to stable program productivity and better industry performance. To conclude, the Korean developmental state is successfully transformed, on the other hand, Taiwan has transformed developmental state into neoliberal state.

In Intelligent Transportation Systems (ITS), disseminating warning messages in a timely and efficient way through wireless short-range communications can save many lives and reduce traffic congestion. A geographical broadcast protocol provides data delivery to specified geographical areas, using multi-hop communications if needed. Among the main challenges for such protocols are forwarder selection and the reduction of the number of hops required to reach and cover the destination area.  In this thesis we propose an efficient geographical broadcast protocol called Preferred and Contention Based Forwarding (PCBF) and evaluate it through simulations. PCBF uses a combination of contention-based forwarding and selecting preferred forwarders also found in other protocols like Emergency Message Dissemination for Vehicular Environments (EMDV). Since the preferred forwarder is allowed to immediately forward the packet (evading contention among other potential forwarders), this approach reduces end-to-end delays. Notable extensions of PCBF compared to EMDV are the use of direct negative acknowledgements in case of unnecessary rebroadcasts and the use of forwarders outside the target region.  Our simulation results show that the PCBF protocol outperforms selected other protocols in terms of end-to-end delay, re-broadcast overhead and reliability in both sparse and dense networks.

La rapide progression des technologies sans fil au cours de ces dernières années a vu naître de nouveaux systèmes de communication dont les réseaux véhiculaires. Ces réseaux visent à intégrer les nouvelles technologies de l’information et de la communication dans le domaine automobile en vue d’améliorer la sécurité et le confort sur le réseau routier. Offrir un accès Internet aux véhicules et à leurs occupants peut sans doute aider à anticiper certains dangers sur la route tout en rendant plus agréables les déplacements à bord des véhicules. Le déploiement de ce service nécessite que des messages soient échangés entre les véhicules. Le routage constitue un élément crucial dans un réseau, car définissant la façon dont les différentes entités échangent des messages. Le routage dans les VANETS constitue un grand défi car ces derniers sont caractérisés par une forte mobilité entraînant une topologie très dynamique. Des protocoles ont été proposés pour étendre Internet aux réseaux véhiculaires. Toutefois, la plupart d’entre eux nécessitent un coût élevé de messages de contrôle pour l’établissement et le maintien des communications. Ceci a pour conséquence la saturation de la bande passante entrainant ainsi une baisse de performance du réseau. Nous proposons dans ce mémoire, un protocole de routage qui s’appuie sur des passerelles mobiles pour étendre Internet aux réseaux véhiculaires. Le protocole prend en compte la mobilité des véhicules et la charge du réseau pour l’établissement et le maintien des routes. / The fast progression of wireless technologies has motivated the emergence of new communications system called VANETS (Vehicular Adhoc Networks). VANETS enable vehicles on the roadway to communicate with each other and with road infrastructure using wireless capabilities. The applications of VANETS include improving safety and comfort on the road. For example, by providing Internet to vehicles, traveling can be safer and more comfortable. To provide Internet connectivity, messages need to be exchanged between the vehicles. However, it is hard to design an efficient routing protocol for connecting vehicles to Internet with a reasonable cost due to high mobility in VANETS. Although, several existing routing protocols have been proposed in the open literature to extend Internet to VANETS, they generate considerable overhead. This leads to unfairly consumption of bandwidth decreasing network performance. We design a routing protocol to connect vehicles to Internet through mobile gateways with the objective to make efficient use of the network bandwidth. Indeed, the protocol significantly reduces the communication overhead required to establish and maintain the routes relying on the mobility of the gateways and the network’s load.

In recent years Content Based Information Retrieval (CBIR) has received a lot of research attention, starting with audio, followed by images and video. Video ngerprinting is a CBIR technique that creates a digital descriptor, also known as a ngerprint, for videos based on its content. These ngerprints are then saved to a database and used to detect unknown videos by comparing the unknown video's ngerprint to the ngerprints in the database to get a match. Many techniques have already been proposed with various levels of success, but most of the existing techniques focus mainly on robustness and neglect the speed of implementation. In this dissertation a novel video ngerprinting technique will be developed with the main focus on detecting advertisements in a television broadcast. Therefore the system must be able to process the incoming video stream in real-time and detect all the advertisements that are present. Even though the algorithm has to be fast, it still has to be robust enough to handle a moderate amount of distortions. These days video ngerprinting still holds many challenges as it involves characterizing videos, made up of sequences of images, e ectively. This means the algorithm must somehow imitate the inherent ability of humans to recognize a video almost instantly. The technique uses the content of the video to derive a ngerprint, thus the features used by the ngerprinting algorithm should be robust to distortions that don't a ect content according to humans. / Thesis (MIng (Computer and Electronic Engineering))--North-West University, Potchefstroom Campus, 2013

In recent years Content Based Information Retrieval (CBIR) has received a lot of research attention, starting with audio, followed by images and video. Video ngerprinting is a CBIR technique that creates a digital descriptor, also known as a ngerprint, for videos based on its content. These ngerprints are then saved to a database and used to detect unknown videos by comparing the unknown video's ngerprint to the ngerprints in the database to get a match. Many techniques have already been proposed with various levels of success, but most of the existing techniques focus mainly on robustness and neglect the speed of implementation. In this dissertation a novel video ngerprinting technique will be developed with the main focus on detecting advertisements in a television broadcast. Therefore the system must be able to process the incoming video stream in real-time and detect all the advertisements that are present. Even though the algorithm has to be fast, it still has to be robust enough to handle a moderate amount of distortions. These days video ngerprinting still holds many challenges as it involves characterizing videos, made up of sequences of images, e ectively. This means the algorithm must somehow imitate the inherent ability of humans to recognize a video almost instantly. The technique uses the content of the video to derive a ngerprint, thus the features used by the ngerprinting algorithm should be robust to distortions that don't a ect content according to humans. / Thesis (MIng (Computer and Electronic Engineering))--North-West University, Potchefstroom Campus, 2013

The increasing popularity of Internet bandwidth-intensive applications prompts us to consider followingproblem: How to compute efficient collective communication schemes on large-scale platform?The issue of designing a collective communication in the context of a large scale distributed networkis a difficult and a multi-level problem. A lot of solutions have been extensively studied andproposed. But a new, comprehensive and systematic approach is required, that combines networkmodels and algorithmic design of solutions.In this work we advocate the use of models that are able to capture real-life network behavior,but also are simple enough that a mathematical analysis of their properties and the design of optimalalgorithms is achievable.First, we consider the problem of the measuring available bandwidth for a given point-topointconnection. We discuss how to obtain reliable datasets of bandwidth measurements usingPlanetLab platform, and we provide our own datasets together with the distributed software usedto obtain it. While those datasets are not a part of our model per se, they are necessary whenevaluating the performance of various network algorithms. Such datasets are common for latencyrelatedproblems, but very rare when dealing with bandwidth-related ones.Then, we advocate for a model that tries to accurately capture the capabilities of a network,named LastMile model. This model assumes that essentially the congestion happens at the edgesconnecting machines to the wide Internet. It has a natural consequence in a bandwidth predictionalgorithm based on this model. Using datasets described earlier, we prove that this algorithm is ableto predict with an accuracy comparable to best known network prediction algorithm (DistributedMatrix Factorization) available bandwidth between two given nodes. While we were unable toimprove upon DMF algorithm in the field of point-to-point prediction, we show that our algorithmhas a clear advantage coming from its simplicity, i.e. it naturally extends to the network predictionsunder congestion scenario (multiple connections sharing a bandwidth over a single link). We areactually able to show, using PlanetLab datasets, that LastMile prediction is better in such scenarios.In the third chapter, we propose new algorithms for solving the large scale broadcast problem.We assume that the network is modeled by the LastMile model. We show that under thisassumption, we are able to provide algorithms with provable, strong approximation ratios. Takingadvantage of the simplicity and elasticity of the model, we can even extend it, so that it captures theidea of connectivity artifacts, in our case firewalls preventing some nodes to communicate directlybetween each other. In the extended case we are also able to provide approximation algorithmswith provable performance.The chapters 1 to 3 form three successful steps of our program to develop from scratch amathematical network communication model, prove it experimentally, and show that it can beapplied to develop algorithms solving hard problems related to design of communication schemesin networks.In the chapter 4 we show how under different network cost models, using some simplifyingassumptions on the structure of network and queries, one can design very efficient communicationschemes using simple combinatorial techniques. This work is complementary to the previous chapter in the sense that previously when designing communication schemes, we assumed atomicityof connections, i.e. that we have no control over routing of simple connections. In chapter 4 weshow how to solve the problem of an efficient routing of network request, given that we know thetopology of the network. It shows the importance of instantiating the parameters and the structureof the network in the context of designing efficient communication schemes.

The importance of communication is vital in wartime. The capability of having all the position information of the allied and enemy forces in a single Tactical Information Display System (TIDS), maintains a great advantage for deciding what to do before the enemy reacts. A Naval Information Distributing System (NIDS) is developed for building an effective communication infrastructure between the war ships. In the designed network, besides the mobile platforms (ships), some fixed platforms (land stations) are used to transfer the information coming from these mobile platforms to all the other platforms. To demonstrate the performance and effectiveness of the Naval Information Distribution System, a discrete event simulation model is developed on a Geographic Information System. The goal of this thesis is to describe and experimentally evaluate an effective and feasible information sharing and routing system for Navy.

Network programming is one of the most important applications in Wireless Sensor Networks as It provides an efficient way to update program Images running on sensor nodes without physical access to them. Securing these updates, however, remains a challenging and important issue, given the open deployment environment of sensor nodes. Though several security schemes have been proposed to impose the authenticity and Integrity protection on network programming applications, they are either energy Inefficient as they tend to use digital signature or lacks the data confidentiality. In addition, due to the absence of secure memory management in the current sensor hardware, the attacker could inject malicious code into the program flash by exploiting buffer overflow In the memory despite the secure code dissemination. The contribution of this thesis Is to provide two software-based security protocols and one hardware-based remote attestation protocol for network programming application. Our first protocol deploys multiple one-way key chains for a multi-hop sensor network. The scheme Is shown to be lower In computational, power consumption and communication costs yet still able to secure multi??hop propagation of program images. Our second protocol utilizes an Iterative hash structure to the data packets in network programming application, ensuring the data confidentiality and authenticity. In addition, we Integrated confidentiality and DoS-attack-resistance in a multi??hop code dissemination protocol. Our final solution is a hardware-based remote attestation protocol for verification of running codes on sensor nodes. An additional piece of tamper-proof hardware, Trusted Platform Module (TPM), is imposed into the sensor nodes. It secures the sensitive information (e.g., the session key) from attackers and monitors any platform environment changes with the Internal registers. With these features of TPM, the code Injection attack could be detected and removed when the contaminated nodes are challenged in our remote attestation protocol. We implement the first two software-based protocols with Deluge as the reference network programming protocol in TinyOS, evaluate them with the extensive simulation using TOSSIM and validate the simulation results with experiments using Tmote. We implement the remote attestation protocol on Fleck, a sensor platform developed by CSIRO that Integrates an Atmel TPM chip.