Contribution to quality of user experience provision over wireless networks

The widespread expansion of wireless networks has brought new attractive possibilities to end users. In addition to the mobility capabilities provided by unwired devices, it is worth remarking the easy configuration process that a user has to follow to gain connectivity through a wireless network. Furthermore, the increasing bandwidth provided by the IEEE 802.11 family has made possible accessing to high-demanding services such as multimedia communications. Multimedia traffic has unique characteristics that make it greatly vulnerable against network impairments, such as packet losses, delay, or jitter. Voice over IP (VoIP) communications, video-conference, video-streaming, etc., are examples of these high-demanding services that need to meet very strict requirements in order to be served with acceptable levels of quality. Accomplishing these tough requirements will become extremely important during the next years, taking into account that consumer video traffic will be the predominant traffic in the Internet during the next years. In wired systems, these requirements are achieved by using Quality of Service (QoS) techniques, such as Differentiated Services (DiffServ), traffic engineering, etc. However, employing these methodologies in wireless networks is not that simple as many other factors impact on the quality of the provided service, e.g., fading, interferences, etc. Focusing on the IEEE 802.11g standard, which is the most extended technology for Wireless Local Area Networks (WLANs), it defines two different architecture schemes. On one hand, the infrastructure mode consists of a central point, which manages the network, assuming network controlling tasks such as IP assignment, routing, accessing security, etc. The rest of the nodes composing the network act as hosts, i.e., they send and receive traffic through the central point. On the other hand, the IEEE 802.11 ad-hoc configuration mode is less extended than the infrastructure one. Under this scheme, there is not a central point in the network, but all the nodes composing the network assume both host and router roles, which permits the quick deployment of a network without a pre-existent infrastructure. This type of networks, so called Mobile Ad-hoc NETworks (MANETs), presents interesting characteristics for situations when the fast deployment of a communication system is needed, e.g., tactics networks, disaster events, or temporary networks. The benefits provided by MANETs are varied, including high mobility possibilities provided to the nodes, network coverage extension, or network reliability avoiding single points of failure. The dynamic nature of these networks makes the nodes to react to topology changes as fast as possible. Moreover, as aforementioned, the transmission of multimedia traffic entails real-time constraints, necessary to provide these services with acceptable levels of quality. For those reasons, efficient routing protocols are needed, capable of providing enough reliability to the network and with the minimum impact to the quality of the service flowing through the nodes. Regarding quality measurements, the current trend is estimating what the end user actually perceives when consuming the service. This paradigm is called Quality of user Experience (QoE) and differs from the traditional Quality of Service (QoS) approach in the human perspective given to quality estimations. In order to measure the subjective opinion that a user has about a given service, different approaches can be taken. The most accurate methodology is performing subjective tests in which a panel of human testers rates the quality of the service under evaluation. This approach returns a quality score, so-called Mean Opinion Score (MOS), for the considered service in a scale 1 - 5. This methodology presents several drawbacks such as its high expenses and the impossibility of performing tests at real time. For those reasons, several mathematical models have been presented in order to provide an estimation of the QoE (MOS) reached by different multimedia services In this thesis, the focus is on evaluating and understanding the multimedia-content transmission-process in wireless networks from a QoE perspective. To this end, firstly, the QoE paradigm is explored aiming at understanding how to evaluate the quality of a given multimedia service. Then, the influence of the impairments introduced by the wireless transmission channel on the multimedia communications is analyzed. Besides, the functioning of different WLAN schemes in order to test their suitability to support highly demanding traffic such as the multimedia transmission is evaluated. Finally, as the main contribution of this thesis, new mechanisms or strategies to improve the quality of multimedia services distributed over IEEE 802.11 networks are presented. Concretely, the distribution of multimedia services over ad-hoc networks is deeply studied. Thus, a novel opportunistic routing protocol, so-called JOKER (auto-adJustable Opportunistic acK/timEr-based Routing) is presented. This proposal permits better support to multimedia services while reducing the energy consumption in comparison with the standard ad-hoc routing protocols.Universidad Politécnica de CartagenaPrograma Oficial de Doctorado en Tecnologías de la Información y Comunicacione

Towards the Quality of Service for VoIP traffic in IEEE 802.11 Wireless Networks

The usage of voice over IP (VoIP) traffic in IEEE 802.11 wireless networks is expected to increase in the near future due to widely deployed 802.11 wireless networks and VoIP services on fixed lines. However, the quality of service (QoS) of VoIP traffic in wireless networks is still unsatisfactory. In this thesis, I identify several sources for the QoS problems of VoIP traffic in IEEE 802.11 wireless networks and propose solutions for these problems. The QoS problems discussed can be divided into three categories, namely, user mobility, VoIP capacity, and call admission control. User mobility causes network disruptions during handoffs. In order to reduce the handoff time between Access Points (APs), I propose a new handoff algorithm, Selective Scanning and Caching, which finds available APs by scanning a minimum number of channels and furthermore allows clients to perform handoffs without scanning, by caching AP information. I also describe a new architecture for the client and server side for seamless IP layer handoffs, which are caused when mobile clients change the subnet due to layer 2 handoffs. I also present two methods to improve VoIP capacity for 802.11 networks, Adaptive Priority Control (APC) and Dynamic Point Coordination Function (DPCF). APC is a new packet scheduling algorithm at the AP and improves the capacity by balancing the uplink and downlink delay of VoIP traffic, and DPCF uses a polling based protocol and minimizes the bandwidth wasted from unnecessary polling, using a dynamic polling list. Additionally, I estimated the capacity for VoIP traffic in IEEE 802.11 wireless networks via theoretical analysis, simulations, and experiments in a wireless test-bed and show how to avoid mistakes in the measurements and comparisons. Finally, to protect the QoS for existing VoIP calls while maximizing the channel utilization, I propose a novel admission control algorithm called QP-CAT (Queue size Prediction using Computation of Additional Transmission), which accurately predicts the impact of new voice calls by virtually transmitting virtual new VoIP traffic

Energy Efficient Packet Size Optimization for Wireless Ad Hoc Networks

PhDEnergy efficiency is crucial for ad hoc networks because of limited energy stored in the battery. Recharging the nodes frequently is sometimes not possible. Therefore, proper energy utilization is paramount. One possible solution of increasing energy efficiency is to optimize the transmitted packet size. But, we claim that only optimal packet size can not boost the energy efficiency in the noisy channel due to high packet loss rate and overhead. Hence, to reduce the overhead size and packet loss, compression and Forward Error Correction (FEC) code are used as remedy. However, every method has its own cost. For compression and FEC, the costs are computation energy cost and extra processing time. Therefore, to estimate the energy-optimize packet size with FEC or compression, processing energy cost and delay need to be considered for precise estimation. Otherwise, for delay sensitive real time applications (such as: VoIP, multimedia) over ad hoc network, energy efficient optimal packet size can be overestimated. We will investigate without degrading the Quality of Service (QoS) with these two different techniques FEC and compression, how much energy efficiency can be achieved by using the energy efficient optimal packet size for different scenarios such as: single hop, multi-hop, multiple source congested network etc. This thesis also shows the impact of time variable channel, packet fragmentation, packet collision on the optimal packet size and energy efficiency. Our results show that, for larger packets, error correction improves the energy efficiency in multi-hop networks only for delay tolerant applications. Whereas for smaller packets, compression is more energy efficient most of the cases. For real-time application like VoIP the scope of increasing the energy efficiency by optimizing packet after maintaining all the constraints is very limited. However, it is shown that, in many cases, optimal packet size improves energy efficiency significantly and also reduces the overall packet loss

Design and Analysis of Medium Access Control Protocols for Broadband Wireless Networks

The next-generation wireless networks are expected to integrate diverse network architectures and various wireless access technologies to provide a robust solution for ubiquitous broadband wireless access, such as wireless local area networks (WLANs), Ultra-Wideband (UWB), and millimeter-wave (mmWave) based wireless personal area networks (WPANs), etc. To enhance the spectral efficiency and link reliability, smart antenna systems have been proposed as a promising candidate for future broadband access networks. To effectively exploit the increased capabilities of the emerging wireless networks, the different network characteristics and the underlying physical layer features need to be considered in the medium access control (MAC) design, which plays a critical role in providing efficient and fair resource sharing among multiple users. In this thesis, we comprehensively investigate the MAC design in both single- and multi-hop broadband wireless networks, with and without infrastructure support. We first develop mathematical models to identify the performance bottlenecks and constraints in the design and operation of existing MAC. We then use a cross-layer approach to mitigate the identified bottleneck problems. Finally, by evaluating the performance of the proposed protocols with analytical models and extensive simulations, we determine the optimal protocol parameters to maximize the network performance. In specific, a generic analytical framework is developed for capacity study of an IEEE 802.11 WLAN in support of non-persistent asymmetric traffic flows. The analysis can be applied for effective admission control to guarantee the quality of service (QoS) performance of multimedia applications. As the access point (AP) becomes the bottleneck in an infrastructure based WLAN, we explore the multiple-input multiple-output (MIMO) capability in the future IEEE 802.11n WLANs and propose a MIMO-aware multi-user (MU) MAC. By exploiting the multi-user degree of freedom in a MIMO system to allow the AP to communicate with multiple users in the downlink simultaneously, the proposed MU MAC can minimize the AP-bottleneck effect and significantly improve the network capacity. Other enhanced MAC mechanisms, e.g., frame aggregation and bidirectional transmissions, are also studied. Furthermore, different from a narrowband system where simultaneous transmissions by nearby neighbors collide with each other, wideband system can support multiple concurrent transmissions if the multi-user interference can be properly managed. Taking advantage of the salient features of UWB and mmWave communications, we propose an exclusive region (ER) based MAC protocol to exploit the spatial multiplexing gain of centralized UWB and mmWave based wireless networks. Moreover, instead of studying the asymptotic capacity bounds of arbitrary networks which may be too loose to be useful in realistic networks, we derive the expected capacity or transport capacity of UWB and mmWave based networks with random topology. The analysis reveals the main factors affecting the network (transport) capacity, and how to determine the best protocol parameters to maximize the network capacity. In addition, due to limited transmission range, multi-hop relay is necessary to extend the communication coverage of UWB networks. A simple, scalable, and distributed UWB MAC protocol is crucial for efficiently utilizing the large bandwidth of UWB channels and enabling numerous new applications cost-effectively. To address this issue, we further design a distributed asynchronous ER based MAC for multi-hop UWB networks and derive the optimal ER size towards the maximum network throughput. The proposed MAC can significantly improve both network throughput and fairness performance, while the throughput and fairness are usually treated as a tradeoff in other MAC protocols

Progressive transmission of medical images

A novel adaptive source-channel coding scheme for progressive transmission of medical images with a feedback system is therefore proposed in this dissertation. The overall design includes Discrete Wavelet Transform (DWT), Embedded Zerotree Wavelet (EZW) coding, Joint Source-Channel Coding (JSCC), prioritization of region of interest (RoI), variability of parity length based on feedback, and the corresponding hardware design utilising Simulink. The JSCC can achieve an efficient transmission by incorporating unequal error projection (UEP) and rate allocation. An algorithm is also developed to estimate the number of erroneous data in the receiver. The algorithm detects the address in which the number of symbols for each subblock is indicated, and reassigns an estimated correct data according to a decision making criterion, if error data is detected. The proposed system has been designed based on Simulink which can be used to generate netlist for portable devices. A new compression method called Compressive Sensing (CS) is also revisited in this work. CS exhibits many advantages in comparison with EZW based on our experimental results. DICOM JPEG2000 is an efficient coding standard for lossy or lossless multi-component image coding. However, it does not provide any mechanism for automatic RoI definition, and is more complex compared to our proposed scheme. The proposed system significantly reduces the transmission time, lowers computation cost, and maintains an error-free state in the RoI with regards to the above provided features. A MATLAB-based TCP/IP connection is established to demonstrate the efficacy of the proposed interactive and adaptive progressive transmission system. The proposed system is simulated for both binary and symmetric channel (BSC) and Rayleigh channel. The experimental results confirm the effectiveness of the design.EThOS - Electronic Theses Online ServiceGBUnited Kingdo

System level modelling and design of hypergraph based wireless system area networks for multi-computer systems

This thesis deals with issues pertaining the wireless multicomputer interconnection networks namely topology and Medium Access Control (MAC). It argues that new channel assignment technique based on regular low-dimensional hypergraph networks, the dual radio wireless hypermesh, represents a promising alternative high-performance wireless interconnection network for the future multicomputers to shared communication medium networks and/or ordinary wireless mesh networks, which have been widely used in current wireless networks. The focus of this work is on improving the network throughput while maintaining a relatively low latency of a wireless network system. By means of a Carrier Sense Multiple Access (CSMA) based design of the MAC protocol and based on the desirable features of hypermesh network topology a relatively high performance network has been introduced. Compared to the CSMA shared communication channel model, which is currently the de facto MAC protocol for most of wireless networks, our design is shown to achieve a significant increase in network throughput with less average network latency for large number of communication nodes. SystemC model of the proposed wireless hypermesh, validated through mathematical models, are then introduced. The analysis has been incorporated in the proper SystemC design methodology which facilitates the integration of communication modelling into the design modelling at the early stages of the system development. Another important application of SystemC modelling techniques is to perform meaningful comparative studies of different protocols, or new implementations to determine which communication scenario performs better and the ability to modify models to test system sensitivity and tune performance. Effects of different design parameters (e.g., packet sizes, number of nodes) has been carried out throughout this work. The results shows that the proposed structure has out perform the existing shared medium network structure and it can support relatively high number of wireless connected computers than conventional networks

Progressive transmission of medical images

A novel adaptive source-channel coding scheme for progressive transmission of medical images with a feedback system is therefore proposed in this dissertation. The overall design includes Discrete Wavelet Transform (DWT), Embedded Zerotree Wavelet (EZW) coding, Joint Source-Channel Coding (JSCC), prioritization of region of interest (RoI), variability of parity length based on feedback, and the corresponding hardware design utilising Simulink. The JSCC can achieve an efficient transmission by incorporating unequal error projection (UEP) and rate allocation. An algorithm is also developed to estimate the number of erroneous data in the receiver. The algorithm detects the address in which the number of symbols for each subblock is indicated, and reassigns an estimated correct data according to a decision making criterion, if error data is detected. The proposed system has been designed based on Simulink which can be used to generate netlist for portable devices. A new compression method called Compressive Sensing (CS) is also revisited in this work. CS exhibits many advantages in comparison with EZW based on our experimental results. DICOM JPEG2000 is an efficient coding standard for lossy or lossless multi-component image coding. However, it does not provide any mechanism for automatic RoI definition, and is more complex compared to our proposed scheme. The proposed system significantly reduces the transmission time, lowers computation cost, and maintains an error-free state in the RoI with regards to the above provided features. A MATLAB-based TCP/IP connection is established to demonstrate the efficacy of the proposed interactive and adaptive progressive transmission system. The proposed system is simulated for both binary and symmetric channel (BSC) and Rayleigh channel. The experimental results confirm the effectiveness of the desig