Anycast in next generation Internet Protocol is a hot topic in the research of computer networks. It has promising potentials and also many challenges, such as architecture, routing, Quality-of-Service, anycast in ad hoc networks, application-layer anycast, etc. In this thesis, we tackle some important topics among them. The thesis at first presents an introduction about anycast, followed by the related work. Then, as our major contributions, a number of challenging issues are addressed in the following chapters. We tackled the anycast routing problem by proposing a requirement based probing algorithm at application layer for anycast routing. Compared with the existing periodical based probing routing algorithm, the proposed routing algorithm improves the performance in terms of delay. We addressed the reliable service problem by the design of a twin server model for the anycast servers, providing a transparent and reliable service for all anycast queries. We addressed the load balance problem of anycast servers by proposing new job deviation strategies, to provide a similar Quality-of-Service to all clients of anycast servers. We applied the mesh routing methodology in the anycast routing in ad hoc networking environment, which provides a reliable routing service and uses much less network resources. We combined the anycast protocol and the multicast protocol to provide a bidirectional service, and applied the service to Web-based database applications, achieving a better query efficiency and data synchronization. Finally, we proposed a new Internet based service, minicast, as the combination of the anycast and multicast protocols. Such a service has potential applications in information retrieval, parallel computing, cache queries, etc. We show that the minicast service consumes less network resources while providing the same services. The last chapter of the thesis presents the conclusions and discusses the future work.

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In an attempt to slow the exhaustion of the Internet Protocol (IP) address space, Class-less Inter-Domain Routing (CIDR) was proposed and adopted. However, the decision to utilize CIDR also increases the size of the routing table, since it allows an arbitrary partitioning of the routing space. We propose a scheme to reduce the size of routing table in the CIDR context. Our approach utilizes a well-known and highly efficient heuristic to perform 2-level logic minimization in order to compress the routing table. By considering the IP routing table as a set of completely specified logic functions, we demonstrate that our technique can achieve about 25% reduction in the size of IP routing tables, while ensuring that our approach can handle routing table updates in real-time. The resulting routing table can be used with existing routers without needing any change in architecture. However, by realizing the IP routing table as proposed in this thesis, the implementation requires less complex hardware than Ternary CAM (TCAM) which are traditionally used to implement IP routing tables. The proposed architecture also reduces lookup latency by about 46%, hardware area by 9% and power consumed by 15% in contrast to a TCAM based implementation.

Because Ministry of Transportation and Communication want to improve its service quality and increase its revenue, Unite Communication Technology Ltd. (UCT) bring up ¡§BOT of E-Commerce and Wireless Internet¡¨ to built up the service of wireless internet and e-commerce on the train and in the station. Under the project, this thesis address to operate the value-added service of VOIP business with wireless internet to boost the total return ratio. According to data collection and survey, the financial model was erected to evaluate the feasible of this new business model. To sum up the result, the NPV of the project is 50,227,238 in 19 years (the discount rate equal to 6.49), IRR reach to 23.93%, and payback period is 7.415 years. Although the payback period is long, in terms of sustained and stable profit in the future, the project is still valuable. Furthermore, UCT is a telecommunication company so that the project can use the bandwidth more efficiently to maximum the corporation revenue. Keywords¡GWireless Internet, Voice on Internet Protocol (VOIP), Financial Feasibility Analysis

Issues around Quality of Service (QoS) and security for Voice over IP (VoIP) have been extensively investigated separately, due to the great attention this technology currently attracts. The specific problem this work addresses centres upon the selection of optimal parameters for QoS and security for VoIP streams integrating both network impairments and user perception metrics into a novel empirically-driven approach. Specifically, the simulation model seeks the optimal parameters in terms of variable VoIP payloads, iterative block ciphers, codecs and authentication mechanisms to be used, so that optimum tradeoff between a set of conflicting factors is achieved. The model employs the widely used Transmission Rating Factor, R, as the methodology to predict and measure the perceived QoS based on current transmission and network impairments. The R factor is then used to map perceived QoS to the corresponding Mean Opinion Score value, which gives the average estimation of perceived voice quality (Quality of Experience). Furthermore, a genetic algorithm (GA) has been developed that uses the output from the simulation model as an input into an offline optimisation routine that simultaneously maximises the VoIP call volumes and the Level of Encryption (LoE) per call basis, without degrading the perceived quality of service under a specific threshold as dictated by the R factor. The solutions reflect the optimum combination of parameters for each codec used and due to the small size of the search space the actual speed of GA has been validated against an exhaustive search algorithm. The results extracted from this study demonstrate that under strict and pre-defined parameters the default payload size supported by the codecs is not the optimal selection in terms of call volume maximisation and perceived QoS when encryption is applied.

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Thesis(M.S.)--Auburn University, 2006. / Abstract. Vita. Includes bibliographic references (p.72-78).

Since the introduction of the IP Multimedia Subsystem (IMS) by the Third Generation Partnership Project (3GPP) in 2002, a lot of research has been conducted aimed at designing and implementing IMS capable clients and network elements. Though considerable work has been done in the development of IMS clients, there is no single, free and open source IMS client that provides researchers with all the required functionality needed to test the applications they are developing. For example, several open and closed source SIP/IMS clients are used within the Rhodes University Conver- gence Research Group (RUCRG) to test applications under development, as a result of the fact that the various SIP/IMS clients support different subsets of SIP/IMS features. The lack of a single client and the subsequent use of various clients comes with several problems. Researchers have to know how to deploy, configure, use and at times adapt the various clients to suit their needs. This can be very time consuming and, in fact, contradicts the IMS philosophy (the IMS was proposed to support rapid service creation). This thesis outlines the development of a Java-based, IMS compliant client called RUCRG IMS client, that uses the JAIN SIP Applet Phone (JSAP) as its foundation. JSAP, which originally offered only basic voice calling and instant messaging (IM) capabilities, was modified to be IMS compliant and support video calls, IM and presence using XML Configuration Access Protocol (XCAP).

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