published_or_final_version / abstract / Electrical and Electronic Engineering / Doctoral / Doctor of Philosophy

This paper discusses current network technologies and protocols and presents a simulation study of the most common networking protocol used today, TCP/IP. The TCP/IP protocol stack has many inherent problems that will be shown through this simulation study. Using the SimpleScalar Toolset, the significance of the data copying and checksumming performed in TCP/IP will be shown along with the architecture needed to support the processing of TCP/IP. Solutions for these TCP/IP pitfalls including a zero-copy protocol and a design for an intelligent network interface card will also be presented. / Graduation date: 2000

Thesis (Ph. D.)--University of Hong Kong, 2006. / Title proper from title frame. Also available in printed format.

Yuen Ka-nang. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2001. / Includes bibliographical references (leaves 133-135). / Abstracts in English and Chinese. / Abstract --- p.i / Acknowledgement --- p.iii / Chapter 1. --- INTRODUCTION --- p.1 / Chapter 1.1. --- Background --- p.1 / Chapter 1.2. --- Objectives --- p.1 / Chapter 1.3. --- Overview of Thesis --- p.2 / Chapter 2. --- NETWORK ASPECT OF THE VOIP TECHNOLOGY --- p.3 / Chapter 2.1. --- VoIP Overview --- p.3 / Chapter 2.2. --- Elements in VoIP --- p.3 / Chapter 2.2.1. --- Call Setup --- p.3 / Chapter 2.2.2. --- Media Capture/Playback --- p.4 / Chapter 2.2.3. --- Media Encoding/Decoding --- p.4 / Chapter 2.2.4. --- Media Transportation --- p.5 / Chapter 2.3. --- Performance Factors Affecting VoIP --- p.6 / Chapter 2.3.1. --- Network Bandwidth --- p.6 / Chapter 2.3.2. --- Latency --- p.6 / Chapter 2.3.3. --- Packet Loss --- p.7 / Chapter 2.3.4. --- Voice Quality --- p.7 / Chapter 2.3.5. --- Quality of Service (QoS) --- p.7 / Chapter 2.4. --- Different Requirements of Intranet VoIP and Internet VoIP --- p.8 / Chapter 2.4.1. --- Packet Loss/Delay/Jitter --- p.8 / Chapter 2.4.2. --- Interoperability --- p.9 / Chapter 2.4.3. --- Available Bandwidth --- p.9 / Chapter 2.4.4. --- Security Requirement --- p.10 / Chapter 2.5. --- Some Feasibility Investigations --- p.10 / Chapter 2.5.1. --- Bandwidth Calculation --- p.10 / Chapter 2.5.2. --- Simulation --- p.12 / Chapter 2.5.3. --- Conclusion --- p.17 / Chapter 2.5.4. --- Simulation Restrictions --- p.17 / Chapter 3. --- SOFTWARE ASPECT OF THE VOIP TECHNOLOGY --- p.19 / Chapter 3.1. --- VoIP Client in JMF --- p.19 / Chapter 3.1.1. --- Architecture --- p.20 / Chapter 3.1.2. --- Incoming Voice Stream Handling --- p.23 / Chapter 3.1.3. --- Outgoing Voice Stream Handling --- p.23 / Chapter 3.1.4. --- Relation between Incoming/Outgoing Voice Stream Handling --- p.23 / Chapter 3.1.5. --- Areas for Further Improvement --- p.25 / Chapter 3.2. --- Capture/Playback Enhanced VoIP Client --- p.26 / Chapter 3.2.1. --- Architecture --- p.27 / Chapter 3.2.2. --- Native Voice Playback Mechanism --- p.29 / Chapter 3.2.3. --- Native Voice Capturing Mechanism --- p.31 / Chapter 3.3. --- Win32 C++ VoIP Client --- p.31 / Chapter 3.3.1. --- Objectives --- p.32 / Chapter 3.3.2. --- Architecture --- p.33 / Chapter 3.3.3. --- Problems and Solutions in Implementation --- p.37 / Chapter 3.4. --- Win32 DirectSound C++ VoIP Client --- p.38 / Chapter 3.4.1. --- Architecture --- p.39 / Chapter 3.4.2. --- DirectSound Voice Playback Mechanism --- p.40 / Chapter 3.4.3. --- DirectSound Voice Capturing Mechanism --- p.44 / Chapter 3.5. --- Testing VoIP Clients --- p.45 / Chapter 3.5.1. --- Setup of Experiment --- p.45 / Chapter 3.5.2. --- Experiment Results --- p.47 / Chapter 3.5.3. --- Experiment Conclusion --- p.48 / Chapter 3.6. --- Real-time Voice Stream Mixing Server --- p.48 / Chapter 3.6.1. --- Structure Overview --- p.48 / Chapter 3.6.2. --- Experiment --- p.53 / Chapter 3.6.3. --- Conclusion --- p.54 / Chapter 4. --- EXPERIMENTAL STUDIES --- p.55 / Chapter 4.1. --- Pure IP-side VoIP-based Call Center ´ؤ VoIP in Education --- p.55 / Chapter 4.1.1. --- Architecture --- p.55 / Chapter 4.1.2. --- Client Structure --- p.56 / Chapter 4.1.3. --- Client Applet User Interface --- p.58 / Chapter 4.1.4. --- Observations --- p.63 / Chapter 4.2. --- A Simple PBX Experiment --- p.63 / Chapter 4.2.1. --- Structural Overview --- p.63 / Chapter 4.2.2. --- PSTN Gateway Server Program --- p.64 / Chapter 4.2.3. --- Problems and Solutions in Implementation --- p.66 / Chapter 4.2.4. --- Experiment 1 --- p.66 / Chapter 4.2.5. --- Experiment 2 --- p.68 / Chapter 5. --- A COMPREHENSIVE VOIP PROJECT 一 GRADUATE SECOND PHONE (GSP) --- p.72 / Chapter 5.1. --- Overview --- p.72 / Chapter 5.1.1. --- Background --- p.72 / Chapter 5.1.2. --- Architecture --- p.76 / Chapter 5.1.3. --- Technologies Used --- p.78 / Chapter 5.1.4. --- Major Functions --- p.80 / Chapter 5.2. --- Client --- p.84 / Chapter 5.2.1. --- Structure Overview --- p.85 / Chapter 5.2.2. --- Connection Procedure --- p.89 / Chapter 5.2.3. --- User Interface --- p.91 / Chapter 5.2.4. --- Observations --- p.92 / Chapter 5.3. --- Gateway --- p.94 / Chapter 5.3.1. --- Structure Overview --- p.94 / Chapter 5.3.2. --- Connection Procedure --- p.97 / Chapter 5.3.3. --- Caller ID Simulator --- p.97 / Chapter 5.3.4. --- Observations --- p.98 / Chapter 5.4. --- Server --- p.101 / Chapter 5.4.1. --- Structure Overview --- p.101 / Chapter 5.5. --- Details of Major Functions --- p.103 / Chapter 5.5.1. --- Secure Local Voice Message Box --- p.104 / Chapter 5.5.2. --- Call Distribution --- p.106 / Chapter 5.5.3. --- Call Forward --- p.112 / Chapter 5.5.4. --- Call Transfer --- p.115 / Chapter 5.6. --- Experiments --- p.116 / Chapter 5.6.1. --- Secure Local Voice Message Box --- p.117 / Chapter 5.6.2. --- Call Distribution --- p.118 / Chapter 5.6.3. --- Call Forward --- p.121 / Chapter 5.6.4. --- Call Transfer --- p.122 / Chapter 5.6.5. --- Dial Out --- p.124 / Chapter 5.7. --- Observations --- p.125 / Chapter 5.8. --- Outlook --- p.126 / Chapter 5.9. --- Alternatives --- p.127 / Chapter 5.9.1. --- Netmeeting --- p.127 / Chapter 5.9.2. --- OpenH323 --- p.128 / Chapter 6. --- CONCLUSIONS --- p.129 / Bibliography --- p.133

Thesis (M.S.)--Worcester Polytechnic Institute. / Keywords: Loss; TFRC; AIMD; TCP. Includes bibliographical references (p. 147-155).

Thesis (Ph.D.)--Auburn University, 2005. / Abstract. Vita. Includes bibliographic references (ℓ. 98-99)

Thesis (M.App.Sc.) - Carleton University, 2003. / Includes bibliographical references (p. 90-92). Also available in electronic format on the Internet.

Thesis (Ph. D.)--University of Texas at Austin, 2004. / Supervisors: Brian L. Evans and Gustavo de Veciana. Vita. Includes bibliographical references.

The Transmission Control Protocol (TCP) has been designed for reliable data transport over the Internet. The performance of TCP is strongly influenced by its congestion control algorithms that limit the amount of traffic a sender can transmit based on end-to-end available capacity estimations. These algorithms proved successful in environments where applications rate requirements can be easily anticipated, as is the case for traditional bulk data transfer or interactive applications. However, an important new class of Internet applications has emerged that exhibit significant variations of transmission rate over time. Variable-rate traffic poses a new challenge for congestion control, especially for applications that need to share the limited capacity of a bottleneck over a long delay Internet path (e.g., paths that include satellite links). This thesis first analyses TCP performance of bursty applications that do not send data continuously, but generate data in bursts separated by periods in which little or no data is sent. Simulation analysis shows that standard TCP methods do not provide efficient support for bursty applications that produce variable-rate traffic, especially over long delay paths. Although alternative forms of congestion control like TCP-Friendly Rate Control and the Datagram Congestion Control Protocol have been proposed, they did not achieve widespread deployment. Therefore many current applications that rely upon User Datagram Protocol are not congestion controlled. The use of non-standard or proprietary methods decreases the effectiveness of Internet congestion control and poses a threat to the Internet stability. Solutions are therefore needed to allow bursty applications to use TCP. Chapter three evaluates Congestion Window Validation (CWV), an IETF experimental specification that was proposed to improve support for bursty applications over TCP. It concluded that CWV is too conservative to support many bursty applications and does not provide an incentive to encourage use by application designers. Instead, application designers often avoid generating variable-rate traffic by padding idle periods, which has been shown to waste network resources. CWV is therefore shown to not provide an acceptable solution for variable-rate traffic. In response to this shortfall, a new modification to TCP, TCP-JAGO, is proposed. This allows variable-rate traffic to restart quickly after an inactive (i.e., idle) period and to effectively utilise available network resources while sending at a lower rate than the available rate (i.e., during an application-limited period). The analysis in Chapter five shows that JAGO provides faster convergence to a steady-state rate and improves throughput by more efficiently utilising the network. TCP-JAGO is also shown to provide an appropriate response when congestion is experienced after restart. Variable-rate TCP traffic can also be impacted by the Initial Window algorithm at the start or during the restart of a session. Chapter six considers this problem, where TCP has no prior indication of the network state. A recent proposal for a larger initial window is analysed. Issues and advantages of using a large IW over a range of scenarios are discussed. The thesis concludes by presenting recommendations to improve TCP support for bursty applications. This also provides an incentive for application designers to choose TCP for variable-rate traffic.

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.