Many efforts have been carried out to provide transmission reliability in the history of communication systems. As the demand for real-time applications increased, providing a reliable communication in a timely manner for such applications is strongly desired. Considering timing constraints makes the issue of achieving reliability more difficult. This thesis concentrates on providing reliability for real-time communication in embedded networks by achieving a timing analysis and using the ARQ concept. What is carried out in this thesis is providing retransmission in a real-time manner for embedded networks according to application request. The thesis work focuses on one packet retransmission over a point to point link, but the concept is rich and can be extended to cover application request in real-time embedded networks. Two methods have been fulfilled, and a simulation has been done on the timing analysis focusing on the performance in accepting real-time traffic in the form of separate channels for each application request. The protocol combines ARQ and a scheduling algorithm as a base to support retransmission for hard real-time applications in embedded networks.

The INSPIRE initiative requires that every European member state establishes a national SDI and related web services. Among these a WFS must be implemented to query spatial data. Therefore several Quality of Service specifications must be fullled by this web service, among these are performance, capacity and availability. In this study work a WFS will be implemented for Nature SDI plus. In this European project the WFS will provide spatial data from several data specications of ANNEX I and III. The main object is to test the conformance to the INSPIRE specification. Therefore load and performance tests will be performed.

The emerging vehicular ad-hoc networks (VANETs) offer a variety of applications and new potential markets related to safety, convenience and entertainment, however, they suffer from a number of challenges not shared so deeply by other types of existing networks, particularly, in terms of mobility of nodes, and end-to-end quality of service (QoS) provision. Although several existing works in the literature have attempted to provide efficient protocols at different layers targeted mostly for safety applications, there remain many barriers to be overcome in order to constrain the widespread use of such networks for non-safety applications, specifically, for video streaming: 1) impact of high speed mobility of nodes on end-to-end QoS provision; 2) cross-layer protocol design while keeping low computational complexity; 3) considering customer-oriented QoS metrics in the design of protocols; and 4) maintaining seamless single-hop and multi-hop connection between the destination vehicle and the road side unit (RSU) while network is moving. This thesis addresses each of the above limitations in design of cross-layer protocols for video streaming application. 1) An adaptive MAC retransmission limit selection scheme is proposed to improve the performance of IEEE 802.11p standard MAC protocol for video streaming applications over VANETs. A multi-objective optimization framework, which jointly minimizes the probability of playback freezes and start-up delay of the streamed video at the destination vehicle by tuning the MAC retransmission limit with respect to channel statistics as well as packet transmission rate, is applied at road side unit (RSU). Two-hop transmission is applied in zones in which the destination vehicle is not within the transmission range of any RSU. In the multi-hop scenario, we discuss the computation of access probability used in the MAC adaptation scheme and propose a cross-layer path selection scheme; 2) We take advantage of similarity between multi-hop urban VANETs in dense traffic conditions and mesh connected networks. First, we investigate an application-centric routing scheme for video streaming over mesh connected overlays. Next, we introduce the challenges of urban VANETs compared to mesh networks and extend the proposed scheme in mesh network into a protocol for urban VANETs. A classification-based method is proposed to select an optimal path for video streaming over multi-hop mesh networks. The novelty is to translate the path selection over multi-hop networks to a standard classification problem. The classification is based on minimizing average video packet distortion at the receiving nodes. The classifiers are trained offline using a vast collection of video sequences and wireless channel conditions in order to yield optimal performance during real time path selection. Our method substantially reduces the complexity of conventional exhaustive optimization methods and results in high quality (low distortion). Next, we propose an application-centric routing scheme for real-time video transmission over urban multi-hop vehicular ad-hoc network (VANET) scenarios. Queuing based mobility model, spatial traffic distribution and prob- ability of connectivity for sparse and dense VANET scenarios are taken into consideration in designing the routing protocol. Numerical results demonstrate the gain achieved by the proposed routing scheme versus geographic greedy forwarding in terms of video frame distortion and streaming start-up delay in several urban communication scenarios for various vehicle entrance rate and traffic densities; and 3) finally, the proposed quality-driven routing scheme for delivering video streams is combined with a novel IP management scheme. The routing scheme aims to optimize the visual quality of the transmitted video frames by minimizing the distortion, the start-up delay, and the frequency of the streaming freezes. As the destination vehicle is in motion, it is unrealistic to assume that the vehicle will remain connected to the same access router (AR) for the whole trip. Mobile IP management schemes can benefit from the proposed multi-hop routing protocol in order to adapt proxy mobile IPv6 (PMIPv6) for multi-hop VANET for video streaming applications. The proposed cross-layer protocols can significantly improve the video streaming quality in terms of the number of streaming freezes and start-up delay over VANETs while achieving low computational complexity by using pattern classification methods for optimization.

The objective of this research is to investigate and develop an efficient and seamless data transport protocol for a heterogeneous wireless mobile network. In next-generation network, most of heterogeneous wireless mobile networks will be combined and complementarily constitute a hierarchical network. To integrate different networks, many challenging issues should be solved. In this thesis, an efficient and seamless data transport mechanisms are explored. We investigate the problems that the current transport control protocol (TCP) will experience within the heterogeneous mobile network. In a heterogeneous network, a mobile host experiences drastic changes in network condition during a session. Traditional TCP struggles with abrupt network changes by intersystem handoff and cannot work efficiently in this environment. We propose a TCP scheme to be tailored to the heterogeneous mobile network to support seamless data transport. In the proposed scheme, a TCP is informed the impending handoff events and works differently based on a handoff type. Simulation results present the proposed algorithm improves throughput, stabilizes data transmission rapidly, and provides a seamless data transfer. We also propose an adaptive resource management scheme within a 3G cellular network based on a users priority level to reduce the call dropping and blocking rates. In a heterogeneous network, a network that provides smaller bandwidth may struggle with handed-off calls being served with a higher bandwidth. Therefore, a resource management algorithm should be defined so that an ongoing call is not dropped by a handoff and provides seamless data transfer. We propose an adaptive resource management scheme based on downgrading the quality of some existing services in a 3G cellular network. We analyze the system capacity, call blocking rate and call dropping rate of the proposed algorithm, and simulate the performance variation of the downgraded traffic. The results show that the proposed scheme increases system capacity, and decreases the call dropping rate at the cost of small delay of the downgraded data traffic.

In order to achieve the storage QoS and power saving, we proposed a fast data migration/transmission scheme and a power saving algorithm for Dataenode management. The fast data migration/ transmission scheme consists of three mechanisms. First, it uses multicast to improve the network bandwidth and solve the I/O and bandwidth bottlenecks. Then, a network coding is used to increase the network throughput and retain high fault tolerance. Third, it uses a user/Dataenode connection management to prevent missing the important message and collocates with CPU & I/O bound scheduling to make data evenly stored in the system. Experimental results show the proposed fast data migration/transmission improves 56% and 85% efficiency in the upload bandwidth and the response time. The proposed power saving algorithm applies the Kalman filter first and then add with the pattern analysis to predict the system workload to adjust the number of Dataenodes dynamically in order to save power. Experimental results show that the proposed power saving algorithm for Dataenode management achieves more than 92.97% accuracy in the workload prediction and improves 52.25% in power consumption with 3.82% error rate.

Driven by the ambition for ubiquitous networking, wireless networks had gained substantial technical advances in recent years. Using radio signals in air as data links, wireless networks can get rid of the tangling of wired cables. However, due to the inherent limitations of wireless channels and legacy protocol design, users of wireless networks today still suffer from the problems on low bandwidth and high error rates. The seven-layer Open System Interconnection (OSI) model was originally designed with wired network environments in mind. Following the seven-layer OSI model, each layer is responsible for handling specific tasks without communicating with each other. Due to the relative stability of wired channels, the strictly-layered approach works well in wired network environments. However, its adequacy is a controversy in wireless environments, since wireless networks have completely different characteristics from its wired counterparts. In wireless environments, channel conditions are highly time-varying and are affected by many factors. External interference or signal degradation may lead to severe packet loss. Even signal-to-noise ratios are fine, transmissions may still fail due to collisions when contention-based MAC protocols are adopted. Conventional protocols developed with wired network environments in mind cannot appropriately response to the characteristics of wireless channels and may make wrong reactions. For these reasons, a flexible framework to capture the rapid change conditions of wireless channels and respond to them immediately is necessary. In this dissertation, we design a cross-layer framework with the consideration of wireless network characteristics. By the coordination between the involved layers, the cross-layer framework can adapt to wireless channel conditions and significantly improve QoS in wireless networks. In order to reduce collision probabilities in wireless networks, we propose a novel protocol named Wait-and-Transmit, which effectively alleviates contentions in wireless networks. By reducing collision probabilities of wireless networks, transmission delays can be shortened and throughputs can be significantly improved. Aiming at the transmission paths containing at least one wireless link, a flexible and efficient cross-layer transmission scheme is also present in this dissertation, which separates the rapid change conditions such as collision probabilities from the relatively stable conditions and well responds to these changes. The proposed approaches significantly improve the performance of wireless networks. We believe that these approaches can contribute to the development of wireless networking.

Motivated by the demand of increasing data rate in wireless communication, multiple-antenna communication is becoming a key technology in the next generation wireless system. This dissertation considers three different aspects of multipleantenna communication. The first part is signal detection in the multiple-input multiple-output (MIMO) communication. Some low complexity near optimal detectors are designed based on an improved version of Bell Laboratories Layered Space-Time (BLAST) architecture detection and an iterative space alternating generalized expectation-maximization (SAGE) algorithm. The proposed algorithms can almost achieve the performance of optimal maximum likelihood detection. Signal detections without channel knowledge (noncoherent) and with co-channel interference are also investigated. Novel solutions are proposed with near optimal performance. Secondly, the error exponent of the distributed multiple-antenna communication (relay) in the windband regime is computed. Optimal power allocation between the source and relay node, and geometrical relay node placement are investigated based on the error exponent analysis. Lastly, the quality of service (QoS) of MIMO/single-input single- output(SISO) communication is studied. The tradeoff of the end-to-end distortion and transmission buffer delay is derived. Also, the SNR exponent of the distortion is computed for MIMO communication, which can provide some insights of the interplay among time diversity, space diversity and the spatial multiplex gain.

The increasing demand for real-time multimedia applications and the lack of quality of service (QoS) support in public best-effort or Internet Protocol (IP) networks has prompted many researchers to propose improvements on the QoS of such networks. This research aims to improve the QoS of real-time multimedia applications in public best-effort networks, without modifying the core network infrastructure or the existing codecs of the original media applications. A source buffering control is studied based on a fluid model developed for a single flow transported over a best-effort network while allowing for flow reversal. It is shown that this control is effective for QoS improvement only when there is sufficient flow reversal or packet reordering in the network. An alternate control strategy based on predictive multi-path switching is studied where only two paths are considered as alternate options. Initially, an emulation study is performed, exploring the impact of path loss rate and traffic delay signal frequency content on the proposed control. The study reveals that this control strategy provides the best QoS improvement when the average comprehensive loss rates of the two paths involved are between 5% and 15%, and when the delay signal frequency content is around 0.5 Hz. Linear and nonlinear predictors are developed using actual network data for use in predictive multi-path switching control. The control results show that predictive path switching is better than no path switching, yet no one predictor developed is best for all cases studied. A voting based control strategy is proposed to overcome this problem. The results show that the voting based control strategy results in better performance for all cases studied. An actual voice quality test is performed, proving that predictive path switching is better than no path switching. Despite the improvements obtained, predictive path switching control has some scalability problems and other shortcomings that require further investigation. If there are more paths available to choose from, the increasing overhead in probing traffic might become unacceptable. Further, if most of the VoIP flows on the Internet use this control strategy, then the conclusions of this research might be different, requiring modifications to the proposed approach. Further studies on these problems are needed.

IEEE 802.11 has become the standard of medium access control (MAC) in wireless ad hoc networks. However, due to the embedded binary exponential backoff algorithm, the packet delay and jitter incurred by access collisions and frame retransmissions may grow drastically. The lack of time-constraint considerations in IEEE 802.11 makes it very difficult to provide QoS (Quality of Service) guarantees for multimedia services. Therefore, a lot of research works focusing on priority-based MAC protocols for wireless ad hoc networks have been proposed. Yet, no standards come out until now. This thesis presents a priority-based MAC scheme in wireless ad hoc networks, which not only provides differentiated services, but also improves the QoS limitations of the previously proposed schemes. The main idea of the proposed JMAC (jamming-based MAC) mechanism is that traffic flows with different priorities can be differentiated by transmitting jamming noises of different lengths to interfere with one another. The one with the longest length of jamming noise can start data transmission. Besides, in our design, priority can be dynamically adjusted to allow each MH to change its contention priority and the length of jamming noise in accordance with network congestions. To implement the proposed JMAC, three modules are developed in this thesis: Collision Avoidance, Starvation Prevention, and Deadlock Prevention. For the purpose of evaluation, we perform simulations on the well-known network simulator, NS-2. Our scheme is compared with the EDCF (enhanced distributed coordination function) of IEEE 802.11e¡]draft¡^and one of the existing works. The simulation results demonstrate the effectiveness and superiority of our scheme.

UDP (User Datagram Protocol) and RTP (Real-time Transport Protocol), using fixed bit rate to convey data every time period, are the most pervasive transport protocols for multimedia traffic in communications networks. However, unexpected packet delay/jitter may occur when network becomes congested or channel interference remains unresolved. To reduce packet delay and packet loss for real-time traffic in a hybrid network from wired to wireless ad-hoc, this thesis presents RTP rate control with an ad-hoc gateway to dynamically adjust the transmission rate according to network conditions. With the proposed scheme, a source node can distinguish the two network conditions, congestion and interference, by monitoring RTCP (RTP control protocol) packets regularly reported from destination nodes and the associated ad-hoc gateway. Based on the RTCP reports, a sender node can dynamically change its encoding bit rate to improve the quality of real-time traffic. For the purpose of demonstration, we implement the proposed adaptive rate control scheme on a Linux platform for SIP-phone communications. The experimental results have shown that our proposed scheme not only relieves traffic congestion but also increases the number of received data even in the case of severe channel interference.