M.Phil. (Electrical & Electronic Engineering) / Without any doubt, the entire range of voice and TV signals will migrate to the packet network. The universal addressable mode of Internet protocol (IP) and the interfacing framing structure of Ethernet are the main reasons behind the success of TCP/IP and Ethernet as a packet network and network access scheme mechanisms. Unfortunately, the success of the Internet has been the problem for real-time traffic such as voice, leading to more studies in the domain of Teletraffic Engineering; and the lack of a resource reservation mechanism in Ethernet, which constitutes a huge problem as switching system mechanism, have raised enough challenges for such a migration. In that context, ITU-T has released a series of Recommendation under the umbrella of H.323 to guarantee the required Quality of Service (QoS) for such services. Although the “utilisation” is not a good parameter in terms of traffic and QoS, we are here in proposing a multiplexing scheme with a queuing solution that takes into account the positive correlations of the packet arrival process experienced at the multiplexer input with the aim to optimize the utilisation of the buffer and bandwidth on the one hand; and the ITU-T H.323 Endpoints and Terminals configuration that can sustain such a multiplexing scheme on the other hand. We take into account the solution of the models from the M/M/1 up to G/G/1 queues based on Kolmogorov’s analysis as our solution to provide a better justification of our approach. This solution, the Diffusion approximation, is the limit of the Fluid process that has not been used enough as queuing solution in the domain of networking. Driven by the results of the Fluid method, and the resulting Gaussian distribution from the Diffusion approximation, the application of the asymptotic properties of the Maximum Likelihood Estimation (MLE) as the central limit theorem allowed capturing the fluctuations and therefore filtering out the positive correlations in the queue system. This has resulted in a queue system able to serve 1 erlang (100% of transmission link capacity) of traffic intensity without any extra delay and a queue length which is 60% of buffer utilization when compared to the ordinary Poisson queue length.

Approved for public release, distribution is unlimited / In this thesis, Voice over Internet Protocol (VoIP) technology will be explored and a recommendation of the operational benefit of VoIP will be provided. A network model will be used to demonstrate improvement of voice End-to-End delay by implementing quality of service (QoS) controls. An overview of VoIP requirements will be covered and recommended standards will be reviewed. A clear definition of a Battle Group will be presented and an overview of current analog RF voice technology will be explained. A comparison of RF voice technology and VoIP will modeled using OPNET Modeler 9.0. / Lieutenant, United States Navy

Thesis (M. Tech. Engineering: Electrical--Vaal University of Technology. / This dissertation discusses the design and implementation of a voice over internet protocol system for the multi-user enterprise. It is limited to small to medium enterprises of which the Vaal University of Technology is an example. Voice communications over existing Internet protocol networks are governed by standards, and to develop such a system it is necessary to have a thorough understanding of these standards. Two such standards namely the International Telecommunications Unions H.323 and the Internet Engineering Task Force's SIP were evaluated and compared to each other in terms of their complexity, extensibility and scalability as well as the services they offer. Based on these criteria it was decided to implement a SIP system. A SIP network consists of application software that act as clients and servers, as well as hardware components such as a proxy and redirect and registrar or location servers that allow users of this network to call each other on the data network. Gateways enable users of the network to call regular public switched telephone network numbers. A test network was set up in the laboratory that contained all the hardware and software components. This was done to understand the installation and configuration options of the different software components and to determine the suitability and interoperability of the software components. This network was then migrated to the network of the Vaal University of Technology which allowed selected users to test and use it. Bandwidth use is a major point of contention, and calculations and measurements showed that the codec being used during the voice call is the determining factor. This SIP system is being used on a daily basis and the users report excellent audio quality between soft phones and soft phones, soft phones and normal telephones and even cellular phones.

Leung Cheung-chi. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2001. / Includes bibliographical references (leaves 76-78). / Abstracts in English and Chinese. / Chapter 1 --- INTRODUCTION --- p.1 / Chapter 1.1 --- Background --- p.1 / Chapter 1.2 --- Objective --- p.2 / Chapter 1.3 --- Overview of the Thesis --- p.3 / Chapter 2 --- APPLICATION OF VOIP IN CALL CENTER --- p.6 / Chapter 2.1 --- An Intelligent IP-based Call Center Model --- p.6 / Chapter 2.1.1 --- Major Components --- p.7 / Chapter a) --- VoIP Gateways --- p.7 / Chapter b) --- Automatic Call Distributor (ACD) --- p.8 / Chapter c) --- Operators --- p.8 / Chapter d) --- Monitoring Tool --- p.9 / Chapter 2.1.2 --- Major Functions --- p.9 / Chapter 2.2 --- Experimental Study of an IP-to-IP Call Center - VoIP Application in Education --- p.10 / Chapter 2.2.1 --- Architecture --- p.11 / Chapter 2.2.2 --- Voice Connection Server --- p.12 / Chapter 2.2.3 --- Call Establishment --- p.14 / Chapter 2.2.4 --- A Preliminary Implementation --- p.14 / Chapter 3 --- THE ACD AND ITS SOFTWARE STRUCTURE --- p.17 / Chapter 3.1 --- Three-Layer Software Structure --- p.17 / Chapter 3.1.1 --- Network Infrastructure Layer --- p.18 / Chapter 3.1.2 --- Call Management Layer --- p.18 / Chapter 3.1.3 --- Application Layer --- p.19 / Chapter 3.1.4 --- Interoperation Between Layers --- p.19 / Chapter 3.2 --- Advantages of Adopting this Software Structure --- p.20 / Chapter 3.3 --- Functional Overview of the ACD --- p.21 / Chapter 3.3.1 --- Call Establishment --- p.21 / Chapter 3.3.2 --- Call Waiting --- p.23 / Chapter 3.3.3 --- Call Forwarding --- p.25 / Chapter 3.3.4 --- Routing Mechanism in the ACD --- p.26 / Chapter a) --- "Queues, Operator Groups and Operators" --- p.26 / Chapter b) --- Priority Based Call Routing --- p.28 / Chapter c) --- Routing of New Incoming Calls --- p.29 / Chapter d) --- Assigning Calls in Waiting Queues to Operators --- p.32 / Chapter 4 --- IMPLEMENTATION OF THE ACD --- p.34 / Chapter 4.1 --- Requirements in implementing the ACD --- p.34 / Chapter 4.1.1 --- Asynchronous Method Call --- p.34 / Chapter 4.1.2 --- Transaction Planning --- p.36 / Chapter 4.1.3 --- Failure Handling --- p.37 / Chapter 4.2 --- Available Technologies --- p.38 / Chapter 4.2.1 --- Enterprise JavaBean (EJB) --- p.38 / Chapter a) --- Entity Bean --- p.40 / Chapter b) --- Session Bean --- p.40 / Chapter c) --- Usage of Session Beans and Entity Beans --- p.41 / Chapter 4.2.2 --- COM+ --- p.42 / Chapter 4.2.3 --- EJB vs COM+ --- p.43 / Chapter 4.3 --- Implementation --- p.47 / Chapter 4.3.1 --- Mapping the EJB model to the Implementation of the ACD --- p.47 / Chapter 4.3.2 --- Design of Entity Beans --- p.49 / Chapter 4.3.3 --- Design of Session Beans --- p.51 / Chapter 4.3.4 --- Asynchronous Method Call --- p.53 / Chapter 4.3.5 --- Transaction Planning --- p.55 / Chapter 4.3.6 --- Failure Handling --- p.57 / Chapter a) --- Failure Handling for VoIP gateways --- p.58 / Chapter b) --- Failure Handling in the ACD --- p.60 / Chapter 5 --- AN EXPERIMENT --- p.64 / Chapter 5.1 --- Experiment on the Call Center Prototype --- p.64 / Chapter 5.1.1 --- Setup of the Experiment --- p.64 / Chapter 5.1.2 --- Experimental Results --- p.66 / Chapter a) --- Startup Time for Different Components --- p.66 / Chapter b) --- Possessing Time for Different Requests --- p.67 / Chapter 5.2 --- Observations --- p.69 / Chapter 5.2.1 --- Observations on Experimental Results --- p.69 / Chapter 5.2.2 --- Advantages and Disadvantages of Using EJB --- p.70 / Chapter 6 --- CONCLUSIONS --- p.72 / BIBLIOGRAPHY --- p.76

Thesis (Ph. D.)--School of Electrical and Computer Engineering, Georgia Institute of Technology, 2004. Directed by Mark Clements. / Vita. Includes bibliographical references (leaves 97-100).

M. Tech. Electrical Engineering. / A virtual Mobile Voice over IP (MVoIP) service can be implemented by a Mobile VoIP Operator (MVoIPO) in conjunction with a Mobile Network Operator (MNO). MVoIPOs do not operate their own mobile network infrastructure. Instead, they use the MNO's packet-based cellular network. However, the coexistence between the MVoIPO and the MNO raises two related problems: first, how to handle interconnection settlements, and second, how to (inter)connect the two operators to make such settlements. This dissertation uses a game-theoretic modelling approach to show that it is mutually beneficial economically if the MNO allows the MVoIPO to operate on its network. Further, a Service Level Agreement (SLA)-based Transparent Settlement Agreement (TSA) model is proposed to solve the first problem. The TSA model algorithm calculates the MVoIPO's throughput distribution at the edge of a UMTS Core Network (CN). This facilitates the determination of levels of conformance to the pre-set throughput thresholds and, subsequently, the issuing of compensation to the MVoIPO by the MNO after generating an economically acceptable volume of traffic. Further, possible network architecture to solve the second problem is suggested, by combining the TSA model algorithm, the UMTS CN, the IP Multimedia Subsystem (IMS), and the Online Charging System (OCS)

M. Tech. Electrical Engineering. / Proposes a voice and video service that uses context-awareness and Semantic Web technologies to restrict network access to non-privileged users during crisis situations. The laboratory tests show how the service takes over call adission control from the SIP server, rejects non-privileged calls and drops non-privileged ongoing call sessions. OPNet simulations further show how to proposed service improves network performance based on performance parameters such as end-to-end delay time and throughput.

Thesis (PhD.) -- Bond University, 2007. / "A dissertation submitted in fulfillment of the requirements for the degree of Doctor of Philosophy"-- t.p. Bibliography: leaves 219-242. Also available via the World Wide Web.

Thesis (PhD) - Swinburne University of Technology, Industrial Research Institute Swinburne - 2007. / Submitted for the degree of Doctor of Philosophy at the Industrial Research Institute Swinburne (IRIS), Swinburne University of Technology - 2007. Typescript. "August 2007". Includes bibliographical references (p. 256-270).

Thesis (M.Eng.) - Carleton University, 2005. / Includes bibliographical references (p. 114-118). Also available in electronic format on the Internet.