A Novel Approach for Implementing Worldwide Interoperability for Microwave Access for Video Surveillance

Video surveillance applications have experienced an increase in demand over the last decade. Surveillance systems can easily be found in places such as commercial offices, banks and traffic intersections, parks and recreational areas. Surveillance applications have the potential to be implemented on a WiMAX (Worldwide Interoperability for Microwave Access) network. Moreover, WiMAX devices have been used widely in the market and WiMAX-based video surveillance products have also been available. As a radio technology, WiMAX is a wireless broadband system that offers greater capacity than WiFi networks and wider coverage than cellular networks. The acceptance of WiMAX in the market, the availability of WiMAX products and its technology excellence, contribute to the possibility of implementing it for surveillance application. However, since WiMAX is designed to accommodate various applications with different quality of service (QoS) requirements, dedicated surveillance network implementation of WiMAX may not achieve optimum performance, as all Subscriber Stations (SSs) generate the same QoS requirements. In the medium access (MAC) layer, this thesis proposes a bandwidth allocation scheme that considers the QoS uniformity of the traffic sources. The proposed bandwidth allocation scheme comprises a simplified bandwidth allocation architecture, a packet-aware bandwidth request mechanism and packet-aware scheduling algorithms. The simplified architecture maximizes resources in the Base Station (BS), deactivates unnecessary services and minimizes the processing delay. The proposed bandwidth request mechanism reduces bandwidth grant and transmission delays. The proposed scheduling algorithms prioritize bandwidth granting access to a request that contains important packet(s). The proposed methods in the MAC layer are designed to be applied to existing devices in the market, without the necessity to change hardware. The transport protocol should be able to deliver video with sufficient quality while maintaining low delay connectivity. The proposed transport layer protocol is therefore designed to improve the existing user datagram protocol (UDP) performance by retransmitting packet loss selectively to increase the received video quality, and utilizing MAC support to achieve low delay connectivity. In order to overcome the limitations of the lower layers, this thesis employs a rateless code instead of transport layer redundancy in the application layer. Moreover, this thesis proposes post-decoding error concealment techniques as the last means to overcome packet loss. To evaluate the performances of the proposed methods, simulations are carried out using NS-2 simulator on Linux platform. The proposed methods are compared to existing works to measure their effectiveness. To facilitate the implementation of the transport layer protocols in practical scenarios, UDP packet modification is applied for each transport layer protocol.Indonesian Directorate General of Higher Education (DGHE/DIKTI

Modeling Dynamics of Video Request Routing in Mobile Networks using Adaptive Scheme

3G and 4G equipment development has melodramatically increasing the mobile internet in the recent years. The devices like laptops, cellular mobiles and tablets using the mobile broadband internet like rise steeply. The most popular mobile presentation is the video streaming in the application of hypermedia. In a cost effective way, the big challenge is the quality to make obtainable these services to users. The above task is achievable by means of emerging the LTE (Long Term Evolution) in the world of mobile. With low latency and high data rates in the applications of multimedia the effective services is provided by the LTE equipment features. In this paper, we study and analyze the Quality of Experience (QoE) at the end user for Video on Demand (VoD) over the LTE network. To achieve this, we streamed High Definition (HD) videos based on H.264/AVC and these videos are delivered from foundation to destination using Transport Control Protocol (TCP) and User Datagram Protocol (UDP). Specifically, our study is about QoEassessment in terms of delay variation, packet loss metrics and provides performance assessment to characterize the impact of conveyance layer protocol in video streaming over radio systems like LTE

Resource Management in Multimedia Networked Systems

Error-free multimedia data processing and communication includes providing guaranteed services such as the colloquial telephone. A set of problems have to be solved and handled in the control-management level of the host and underlying network architectures. We discuss in this paper \u27resource management\u27 at the host and network level, and their cooperation to achieve global guaranteed transmission and presentation services, which means end-to-end guarantees. The emphasize is on \u27network resources\u27 (e.g., bandwidth, buffer space) and \u27host resources\u27 (e.g., CPU processing time) which need to be controlled in order to satisfy the Quality of Service (QoS) requirements set by the users of the multimedia networked system. The control of the specified resources involves three actions: (1) properly allocate resources (end-to-end) during the multimedia call establishment, so that traffic can flow according to the QoS specification; (2) control resource allocation during the multimedia transmission; (3) adapt to changes when degradation of system components occurs. These actions imply the necessity of: (a) new services, such as admission services, at the hosts and intermediate network nodes; (b) new protocols for establishing connections which satisfy QoS requirements along the path from send to receiver(s), such as resource reservation protocol; (c) new control algorithms for delay, rate and error control; (d) new resource monitoring protocols for reporting system changes, such as resource administration protocol; (e) new adaptive schemes for dynamic resource allocation to respond to system changes; and (f) new architectures at the hosts and switches to accommodate the resource management entities. This article gives an overview of services, mechanisms and protocols for resource management as outlined above

Optimization and Performance Analysis of High Speed Mobile Access Networks

The end-to-end performance evaluation of high speed broadband mobile access networks is the main focus of this work. Novel transport network adaptive flow control and enhanced congestion control algorithms are proposed, implemented, tested and validated using a comprehensive High speed packet Access (HSPA) system simulator. The simulation analysis confirms that the aforementioned algorithms are able to provide reliable and guaranteed services for both network operators and end users cost-effectively. Further, two novel analytical models one for congestion control and the other for the combined flow control and congestion control which are based on Markov chains are designed and developed to perform the aforementioned analysis efficiently compared to time consuming detailed system simulations. In addition, the effects of the Long Term Evolution (LTE) transport network (S1and X2 interfaces) on the end user performance are investigated and analysed by introducing a novel comprehensive MAC scheduling scheme and a novel transport service differentiation model

Quality-driven resource utilization methods for video streaming in wireless communication networks

This research is focused on the optimisation of resource utilisation in wireless mobile networks with the consideration of the users’ experienced quality of video streaming services. The study specifically considers the new generation of mobile communication networks, i.e. 4G-LTE, as the main research context. The background study provides an overview of the main properties of the relevant technologies investigated. These include video streaming protocols and networks, video service quality assessment methods, the infrastructure and related functionalities of LTE, and resource allocation algorithms in mobile communication systems. A mathematical model based on an objective and no-reference quality assessment metric for video streaming, namely Pause Intensity, is developed in this work for the evaluation of the continuity of streaming services. The analytical model is verified by extensive simulation and subjective testing on the joint impairment effects of the pause duration and pause frequency. Various types of the video contents and different levels of the impairments have been used in the process of validation tests. It has been shown that Pause Intensity is closely correlated with the subjective quality measurement in terms of the Mean Opinion Score and this correlation property is content independent. Based on the Pause Intensity metric, an optimised resource allocation approach is proposed for the given user requirements, communication system specifications and network performances. This approach concerns both system efficiency and fairness when establishing appropriate resource allocation algorithms, together with the consideration of the correlation between the required and allocated data rates per user. Pause Intensity plays a key role here, representing the required level of Quality of Experience (QoE) to ensure the best balance between system efficiency and fairness. The 3GPP Long Term Evolution (LTE) system is used as the main application environment where the proposed research framework is examined and the results are compared with existing scheduling methods on the achievable fairness, efficiency and correlation. Adaptive video streaming technologies are also investigated and combined with our initiatives on determining the distribution of QoE performance across the network. The resulting scheduling process is controlled through the prioritization of users by considering their perceived quality for the services received. Meanwhile, a trade-off between fairness and efficiency is maintained through an online adjustment of the scheduler’s parameters. Furthermore, Pause Intensity is applied to act as a regulator to realise the rate adaptation function during the end user’s playback of the adaptive streaming service. The adaptive rates under various channel conditions and the shape of the QoE distribution amongst the users for different scheduling policies have been demonstrated in the context of LTE. Finally, the work for interworking between mobile communication system at the macro-cell level and the different deployments of WiFi technologies throughout the macro-cell is presented. A QoEdriven approach is proposed to analyse the offloading mechanism of the user’s data (e.g. video traffic) while the new rate distribution algorithm reshapes the network capacity across the macrocell. The scheduling policy derived is used to regulate the performance of the resource allocation across the fair-efficient spectrum. The associated offloading mechanism can properly control the number of the users within the coverages of the macro-cell base station and each of the WiFi access points involved. The performance of the non-seamless and user-controlled mobile traffic offloading (through the mobile WiFi devices) has been evaluated and compared with that of the standard operator-controlled WiFi hotspots

Quality of service optimization of multimedia traffic in mobile networks

Mobile communication systems have continued to evolve beyond the currently deployed Third Generation (3G) systems with the main goal of providing higher capacity. Systems beyond 3G are expected to cater for a wide variety of services such as speech, data, image transmission, video, as well as multimedia services consisting of a combination of these. With the air interface being the bottleneck in mobile networks, recent enhancing technologies such as the High Speed Downlink Packet Access (HSDPA), incorporate major changes to the radio access segment of 3G Universal Mobile Telecommunications System (UMTS). HSDPA introduces new features such as fast link adaptation mechanisms, fast packet scheduling, and physical layer retransmissions in the base stations, necessitating buffering of data at the air interface which presents a bottleneck to end-to-end communication. Hence, in order to provide end-to-end Quality of Service (QoS) guarantees to multimedia services in wireless networks such as HSDPA, efficient buffer management schemes are required at the air interface. The main objective of this thesis is to propose and evaluate solutions that will address the QoS optimization of multimedia traffic at the radio link interface of HSDPA systems. In the thesis, a novel queuing system known as the Time-Space Priority (TSP) scheme is proposed for multimedia traffic QoS control. TSP provides customized preferential treatment to the constituent flows in the multimedia traffic to suit their diverse QoS requirements. With TSP queuing, the real-time component of the multimedia traffic, being delay sensitive and loss tolerant, is given transmission priority; while the non-real-time component, being loss sensitive and delay tolerant, enjoys space priority. Hence, based on the TSP queuing paradigm, new buffer managementalgorithms are designed for joint QoS control of the diverse components in a multimedia session of the same HSDPA user. In the thesis, a TSP based buffer management algorithm known as the Enhanced Time Space Priority (E-TSP) is proposed for HSDPA. E-TSP incorporates flow control mechanisms to mitigate congestion in the air interface buffer of a user with multimedia session comprising real-time and non-real-time flows. Thus, E-TSP is designed to provide efficient network and radio resource utilization to improve end-to-end multimedia traffic performance. In order to allow real-time optimization of the QoS control between the real-time and non-real-time flows of the HSDPA multimedia session, another TSP based buffer management algorithm known as the Dynamic Time Space Priority (D-TSP) is proposed. D-TSP incorporates dynamic priority switching between the real-time and non-real-time flows. D-TSP is designed to allow optimum QoS trade-off between the flows whilst still guaranteeing the stringent real-time component’s QoS requirements. The thesis presents results of extensive performance studies undertaken via analytical modelling and dynamic network-level HSDPA simulations demonstrating the effectiveness of the proposed TSP queuing system and the TSP based buffer management schemes

Content-Aware Multimedia Communications

The demands for fast, economic and reliable dissemination of multimedia information are steadily growing within our society. While people and economy increasingly rely on communication technologies, engineers still struggle with their growing complexity. Complexity in multimedia communication originates from several sources. The most prominent is the unreliability of packet networks like the Internet. Recent advances in scheduling and error control mechanisms for streaming protocols have shown that the quality and robustness of multimedia delivery can be improved significantly when protocols are aware of the content they deliver. However, the proposed mechanisms require close cooperation between transport systems and application layers which increases the overall system complexity. Current approaches also require expensive metrics and focus on special encoding formats only. A general and efficient model is missing so far. This thesis presents efficient and format-independent solutions to support cross-layer coordination in system architectures. In particular, the first contribution of this work is a generic dependency model that enables transport layers to access content-specific properties of media streams, such as dependencies between data units and their importance. The second contribution is the design of a programming model for streaming communication and its implementation as a middleware architecture. The programming model hides the complexity of protocol stacks behind simple programming abstractions, but exposes cross-layer control and monitoring options to application programmers. For example, our interfaces allow programmers to choose appropriate failure semantics at design time while they can refine error protection and visibility of low-level errors at run-time. Based on some examples we show how our middleware simplifies the integration of stream-based communication into large-scale application architectures. An important result of this work is that despite cross-layer cooperation, neither application nor transport protocol designers experience an increase in complexity. Application programmers can even reuse existing streaming protocols which effectively increases system robustness.Der Bedarf unsere Gesellschaft nach kostengünstiger und zuverlässiger Kommunikation wächst stetig. Während wir uns selbst immer mehr von modernen Kommunikationstechnologien abhängig machen, müssen die Ingenieure dieser Technologien sowohl den Bedarf nach schneller Einführung neuer Produkte befriedigen als auch die wachsende Komplexität der Systeme beherrschen. Gerade die Übertragung multimedialer Inhalte wie Video und Audiodaten ist nicht trivial. Einer der prominentesten Gründe dafür ist die Unzuverlässigkeit heutiger Netzwerke, wie z.B.~dem Internet. Paketverluste und schwankende Laufzeiten können die Darstellungsqualität massiv beeinträchtigen. Wie jüngste Entwicklungen im Bereich der Streaming-Protokolle zeigen, sind jedoch Qualität und Robustheit der Übertragung effizient kontrollierbar, wenn Streamingprotokolle Informationen über den Inhalt der transportierten Daten ausnutzen. Existierende Ansätze, die den Inhalt von Multimediadatenströmen beschreiben, sind allerdings meist auf einzelne Kompressionsverfahren spezialisiert und verwenden berechnungsintensive Metriken. Das reduziert ihren praktischen Nutzen deutlich. Außerdem erfordert der Informationsaustausch eine enge Kooperation zwischen Applikationen und Transportschichten. Da allerdings die Schnittstellen aktueller Systemarchitekturen nicht darauf vorbereitet sind, müssen entweder die Schnittstellen erweitert oder alternative Architekturkonzepte geschaffen werden. Die Gefahr beider Varianten ist jedoch, dass sich die Komplexität eines Systems dadurch weiter erhöhen kann. Das zentrale Ziel dieser Dissertation ist es deshalb, schichtenübergreifende Koordination bei gleichzeitiger Reduzierung der Komplexität zu erreichen. Hier leistet die Arbeit zwei Beträge zum aktuellen Stand der Forschung. Erstens definiert sie ein universelles Modell zur Beschreibung von Inhaltsattributen, wie Wichtigkeiten und Abhängigkeitsbeziehungen innerhalb eines Datenstroms. Transportschichten können dieses Wissen zur effizienten Fehlerkontrolle verwenden. Zweitens beschreibt die Arbeit das Noja Programmiermodell für multimediale Middleware. Noja definiert Abstraktionen zur Übertragung und Kontrolle multimedialer Ströme, die die Koordination von Streamingprotokollen mit Applikationen ermöglichen. Zum Beispiel können Programmierer geeignete Fehlersemantiken und Kommunikationstopologien auswählen und den konkreten Fehlerschutz dann zur Laufzeit verfeinern und kontrolliere

Concurrent multipath transmission to improve performance for multi-homed devices in heterogeneous networks

Recent network technology developments have led to the emergence of a variety of access network technologies - such as IEEE 802.11, wireless local area network (WLAN), IEEE 802.16, Worldwide Interoperability for Microwave Access (WIMAX) and Long Term Evolution (LTE) - which can be integrated to offer ubiquitous access in a heterogeneous network environment. User devices also come equipped with multiple network interfaces to connect to the different network technologies, making it possible to establish multiple network paths between end hosts. However, the current connectivity settings confine the user devices to using a single network path at a time, leading to low utilization of the resources in a heterogeneous network and poor performance for demanding applications, such as high definition video streaming. The simultaneous use of multiple network interfaces, also called bandwidth aggregation, can increase application throughput and reduce the packets' end-to-end delays. However, multiple independent paths often have heterogeneous characteristics in terms of offered bandwidth, latency and loss rate, making it challenging to achieve efficient bandwidth aggregation. For instance, striping the flow's packets over multiple network paths with different latencies can cause packet reordering, which can significantly degrade performance of the current transport protocols. This thesis proposes three new solutions to mitigate the effects of network path heterogeneity on the performance of various concurrent multipath transmission settings. First, a network layer solution is proposed to stripe packets of delay-sensitive and high-bandwidth applications for concurrent transmission across multiple network paths. The solution leverages the paths' latency heterogeneity to reduce packet reordering, leading to minimal reordering delay, which improves performance of delay-sensitive applications. Second, multipath video streaming is developed for H.264 scalable video, where the reference video packets are adaptively assigned to low loss network paths to reduce drifting errors, thus combatting H.264 video distortion effectively. Finally, a new segment scheduling framework - which carefully considers path heterogeneity - is incorporated into the IETF Multipath TCP to improve throughput performance. The proposed solutions have been validated using a series of simulation experiments. The results reveal that the proposed solutions can enable efficient bandwidth aggregation for concurrent multipath transmission over heterogeneous network paths