The use of microcontrollers and FPGAs is getting more and more wide spread in electronic designs. A recent developmenthas been to implement microcontrollers onboard the FPGA, there are a lot of benefits but also disadvantages with this. Often the microcontroler requires a lot of resources in the expensive FPGA. This is where PicoBlaze, a microcontroller provided by Xilinx, fits in. It is designed with one main object, keep it as small and powerful as possible. In this report PicoBlaze is evaluated and documented. Two implementations have been done. One smaller to show how to use PicoBlaze and one larger implementation of an Ethernet network interface. The function of the implementations have been verified on a experiment board utilizing a Virtex-II FPGA. The conclusion is that PicoBlaze is a very powerful microcontroller in comparison to the resources it uses on the FPGA. It uses only a little more than 80 slices on a Virtex II FPGA. This is its main advantage, the disadvantages of PicoBlaze is its limited program memory and the limited address space.

The objective of the thesis is primarily to conform and encode a digital signal to be controlled by software. The idea is to transport today’s tracesignal through the IP-network instead of cable-net. In this report I will suggest a possible solution to get a fully passed result with no peculiar time delay. I will also suggest some ideas for a future work with the new tracesystem.

POTS (Plain Old Telephony Service) systems have traditionally been connected via synchronous connections. When installing new nodes in the telephone network, they may sometimes be connected via packet networks such as Ethernet. Ethernet is an asynchronous network which means that nodes connected to the network don’t have access to the same clock frequency if it is not provided in some other way. If two nodes have different clock frequency, the receiver’s buffer will eventually overflow or starve. While not being a severe problem for telephony, devices used for data transmission, e.g. modems and fax will not be able to function properly. To avoid this it is necessary to synchronize the nodes. This thesis investigates methods to synchronize nodes connected over Ethernet by simulating them in Matlab. The simulations show that under certain circumstances it is possible to produce a clock signal conforming to relevant standards.

Expanding the global Internet to include mobile devices is an exciting area of current research. Because of the vast size of the Internet, and because the protocols in it are already widely deployed, mobile devices must inter-operate with those protocols. Although most of the incompatiblities with mobiles have been solved, the protocols that deliver data reliably, and that account for the majority of Internet traffic, perform very poorly. A change in location causes a disruption in traffic, and disruption is dealt with by algorithms tailored only for stationary hosts. The Transmission Control Protocol (TCP) is the predominant transport-layer protocol in the Internet. In this thesis, we look at the performance of TCP in mobile environments. We provide a complete explanation for poor performance; we conduct a large number of experiments, simulations, and analyses that prove and quantify poor performance;and we propose simple and scalable solutions that address the limitations.

The combination of infrastructure-to-vehicle and vehicle-to-vehicle communications, namely the multi-hop Vehicular Communications Network (VCN) , appears as a promising solution for the ubiquitous access to IP services in vehicular environments. In this thesis, we address the challenges of multi-hop VCN, and investigate the seamless provision of IP services over such network. Three different schemes are proposed and analyzed. First, we study the limitations of current standards for the provision of IP services, such as 802.11p/WAVE, and propose a framework that enables multi-hop communications and a robust IP mobility mechanism over WAVE. An accurate analytical model is developed to evaluate the throughput performance, and to determine the feasibility of the deployment of IP-based services in 802.11p/WAVE networks. Next, the IP mobility support is extended to asymmetric multi-hop VCN. The proposed IP mobility and routing mechanisms react to the asymmetric links, and also employ geographic location and road traffic information to enable predictive handovers. Moreover, since multi-hop communications suffer from security threats, it ensures that all mobility signalling is authenticated among the participant vehicles. Last, we extend our study to a heterogeneous multi-hop VCN, and propose a hybrid scheme that allows for the on-going IP sessions to be transferred along the heterogeneous communications system. The proposed global IP mobility scheme focuses on urban vehicular scenarios, and enables seamless communications for in-vehicle networks, commuters, and pedestrians. The overall performance of IP applications over multi-hop VCN are improved substantially by the proposed schemes. This is demonstrated by means of analytical evaluations, as well as extensive simulations that are carried out in realistic highway and urban vehicular scenarios. More importantly, we believe that our dissertation provides useful analytical tools, for evaluating the throughput and delay performance of IP applications in multi-hop vehicular environments. In addition, we provide a set of practical and efficient solutions for the seamless support of IP tra c along the heterogeneous and multi-hop vehicular network, which will help on achieving ubiquitous drive-thru Internet, and infotainment traffic access in both urban and highway scenarios.

The explosive growth of the Internet during the last few years, has been propelled by the TCP/IP protocol suite and the best effort packet forwarding service. However, quality of service (QoS) is far from being a reality especially for multimedia services like video streaming and video conferencing. In the case of wireless and mobile networks, the problem becomes even worse due to the physics of the medium, resulting into further deterioration of the system performance. Goal of this dissertation is the systematic development of comprehensive models that jointly characterize the performance of transport protocols and media delivery in heterogeneous wireless networks. At the core of our novel methodology, is the use of analytical models for driving the design of media transport algorithms, so that the delivery of conversational and non-interactive multimedia data is enhanced in terms of throughput, delay, and jitter. More speciffically, we develop analytical models that characterize the throughput and goodput of the transmission control protocol (TCP) and the transmission friendly rate control (TFRC) protocol, when CBR and VBR multimedia workloads are considered. Subsequently, we enhance the transport protocol models with new parameters that capture the playback buffer performance and the expected video distortion at the receiver. In this way a complete end-to-end model for media streaming is obtained. This model is used as a basis for a new algorithm for rate-distortion optimized mode selection in video streaming appli- cations. As a next step, we extend the developed models for the aforementioned protocols, so that heterogeneous wireless networks can be accommodated. Subsequently, new algorithms are proposed in order to enhance the developed media streaming algorithms when heterogeneous wireless networks are also included. Finally, the aforementioned models and algorithms are extended for the case of concurrent multipath media transport over several hybrid wired/wireless links.

As multimedia processing and networking technologies, products and services evolve, the number of users communicating, collaborating and entertaining over the IP networks is growing rapidly. With the emergence of pervasive and ubiquitous multimedia services, this proliferation creates an abundant increase in the amount of the Internet backbone traffic. This brings the problem of efficient transmission of real-time and time-sensitive media content to the fore. Effective multimedia services demand appropriate application-specific and media-aware solutions, without which the full benefits of such services will not be realized. Poor approaches often lead to system performance degradations such as unacceptable presentation quality perceived by the users, possible network collapses due to the high-bandwidth nature of the multimedia applications, and poor performance observed by other data-oriented applications due to the unresponsiveness of multimedia flows. From a networking perspective, traditional approaches consider the application data as "sacred" and do not differentiate any part of it from the rest. While this keeps the data-delivery mechanisms, namely, the transport-layer protocols, as plain as possible, it also precludes these mechanisms from interpreting the media content and tailoring their actions according to the importance of the content. Given that this naive approach cannot satisfy the specific needs of each and every one of the today's emerging applications ranging from videotelephony to video-on-demand, from distance education to telemedicine, from remote surveillance to online video gaming, the study of Multimedia Transport Protocols (MMTP) is overdue. An MMTP solution basically integrates the multimedia content information into the responsible data-delivery mechanisms along with the requirements of the invoking application and network characteristics to deliver the highest level of service quality. In other words, an MMTP solution offers a unified environment where all cooperating protocol components interact with each other and make the best use of this collaboration to fulfill their respective duties. The focus of this thesis is on the design and evaluation of a set of end-to-end and system-level MMTP solutions for scalable, reliable, and high quality multimedia services in ever-changing, complex and heterogeneous computing and communication environments.

Interdomain traffic engineering (TE) controls the flow of traffic between autonomous systems (ASes) to achieve performance goals under various resource constraints. Interdomain TE can be categorized into ingress TE and egress TE, which aim to control the ingress and egress traffic flow in a network, respectively. Most interdomain TE techniques are based on BGP, which was not designed to support performance based routing. Hence even though some basic interdomain TE techniques are widely deployed, their overall effectiveness and impact on interdomain traffic are not well understood. Furthermore, systematic practices for deploying these techniques have yet to be developed. In this thesis, we explore these open issues for both ingress and egress TE. We first focus on the AS-Path prepending technique in interdomain ingress TE. We design a polynomial algorithm that takes network settings as input and produces the optimal prepending at each ingress link. We also develop methods to measure the inputs of the optimal algorithm by leveraging widely available looking glass severs and evaluate the errors of such measurement. We further propose an algorithm, based on this optimal algorithm, that is robust to input errors. We then focus on Intelligent Routing Control (IRC) systems often used at multihomed networks for egress interdomain TE. To address the possible traffic oscillation problem caused by multiple IRC systems, we design a class of randomized IRC algorithms. Through simulations, we show that the proposed algorithms can effectively mitigate oscillations. We also show that IRC systems using randomized path switching algorithms perform better than those switching path deterministically, when both types of IRC systems co-exist. To further understand the performance impact of IRC systems, we next focus on the performance of applications, such as TCP connections. We study the synergistic and antagonistic interactions between IRC and TCP connections, through a simple dual-feedback model. We first examine the impact of sudden RTT and available bandwidth changes in TCP connection. We then examine the effect of IRC measurement delays on closed loop traffic. We also show the conditions under which IRC is beneficial under various path impairment models.

We propose that the industry structure in Taiwan broadband market is a two-sided market. In this framework, the networks need to be completely interconnected in order to ensure unhindered (or smoothly) information flow. Based on a two-sided market model, we analyze the IP peering mechanism for Taiwan Internet market. We show that the IP peering access charges should be a very low constant amount to reflect the unique Taiwan broadband industry structure. Furthermore, in attracting more Internet content providers (ICP) and end users to provide more content services and Internet applications, the Internet service providers (ISP) should provide free broadband services to ICPs. Though these results are contradictory with the ¡§user-pays¡¨ principle, it ensures more profitable for ISPs and ICPs. Most importantly, the impacts on the whole social welfare are improved. Last, we examine a more efficacious framework for ensuring network neutrality is Efficient Component Pricing Rule (ECPR) in a vertically-integrated monopoly market, as in Taiwan Broadband industry.

Mission-critical system typically refers to a project or system for which the success is vital to the mission of the underlying organization. The failure or delayed completion of the tasks in mission-critical systems may cause severe financial loss, even human casualties. For example, failure of an accurate and timely forecast of Hurricane Rita in September 2005 caused enormous financial loss and several deaths. As such, real-time guarantee and reliability have always been two key foci of mission-critical system design. Many factors affect real-time guarantee and reliability. From the software design perspective, which is the focus of this paper, three aspects are most important. The first of these is how to design a single application to effectively support real-time requirement and improve reliability, the second is how to integrate different applications in a cluster environment to guarantee real-time requirement and improve reliability, and the third is how to effectively coordinate distributed applications to support real-time requirements and improve reliability. Following these three aspects, this dissertation proposes and implements three novel methodologies: real-time component based single node application development, real-time workflow-based cluster application integration, and real-time distributed admission control. For ease of understanding, we introduce these three methodologies and implementations in three real-world mission-critical application systems: single node mission-critical system, cluster environment mission-critical system, and wide-area network mission-critical system. We study full-scale design and implementation of these mission-critical systems, more specifically: 1) For the single node system, we introduce a real-time component based application model, a novel design methodology, and based on the model and methodology, we implement a real-time component based Enterprise JavaBean (EJB) System. Through component based design, efficient resource management and scheduling, we show that our model and design methodology can effectively improve system reliability and guarantee real-time requirement. 2) For the system in a cluster environment, we introduce a new application model, a real-time workflow-based application integration methodology, and based on the model and methodology, we implement a data center management system for the Southeastern Universities Research Association (SURA) project. We show that our methodology can greatly simplify the design of such a system and make it easier to meet deadline requirements, while improving system reliability through the reuse of fully tested legacy models. 3) For the system in a wide area network, we narrow our focus to a representative VoIP system and introduce a general distributed real-time VoIP system model, a novel system design methodology, and an implementation. We show that with our new model and architectural design mechanism, we can provide effective real-time requirement for Voice over Internet Protocol (VoIP).