The principle of parsimonious modelling of Internet traffic states that a minimal number of descriptors should be used for its characterization. Until early 1990s, the conventional Markovian models for voice traffic had been considered suitable and parsimonious for data traffic as well. Later with the discovery of strong correlations and increased burstiness in Internet traffic, various self-similar count models have been proposed. But, in fact, such models are strictly mono-fractal and applicable at coarse time scales, whereas Internet traffic modelling is about modelling traffic at fine and coarse time scales; modelling traffic which can be mono and multi-fractal; modelling traffic at interarrival time and count levels; modelling traffic at access and core tiers; and modelling all the three structural components of Internet traffic, that is, packets, flows and sessions. The philosophy of this thesis can be described as: “the renewal of renewal theory in Internet traffic modelling”. Renewal theory has a great potential in modelling statistical characteristics of Internet traffic belonging to individual users, access and core networks. In this thesis, we develop an Internet traffic modelling framework based on fractal renewal processes, that is, renewal processes with underlying distribution of interarrival times being heavy-tailed. The proposed renewal framework covers packets, flows and sessions as structural components of Internet traffic and is applicable for modelling the traffic at fine and coarse time scales. The properties of superposition of renewal processes can be used to model traffic in higher tiers of the Internet hierarchy. As the framework is based on renewal processes, therefore, Internet traffic can be modelled at both interarrival times and count levels.

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.

Approved for public release; distribution is unlimited / Virtual Private Networks (VPNs) are designed to use the Transmission Control Protocol (TCP) or User Datagram Protocol (UDP) to establish secure communication tunnels over public Internet. Multipath TCP (MPTCP) extends TCP to allow data to be delivered over multiple network paths simultaneously. This thesis first builds a testbed and investigates the potential of using MPTCP tunnels to increase the goodput of VPN communications and support seamless mobility. Based on the empirical results and an analysis of the MPTCP design in Linux kernels, we further introduce a full-multipath kernel, implementing a basic Multipath UDP (MPUDP) protocol into an existing Linux MPTCP kernel.We demonstrate the MPUDP protocol provides performance improvements over single path UDP tunnels and in some cases MPTCP tunnels. The MPUDP kernel should be further developed to include more efficient scheduling algorithms and path managers to allow better performance and mobility benefits seen with MPTCP. / Outstanding Thesis / Lieutenant, United States Navy

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.

Com o advento da internet, a comunicação entre sistemas remotos foi amplamente facilitada, tornando viável a aplicação de sistemas supervisórios através dela, principalmente via protocolo TCP/IP. Embora esses sistemas já sejam empregados na indústria, principalmente utilizando como driver os controladores lógicos programáveis (CLP\'s), a implementação de qualquer sistema mais simples de supervisionamento teria um custo elevado, devido tanto ao preço do hardware quanto do software envolvido. Além disso, os programas supervisórios existentes fornecem um número limitado de recursos de visualização, que se restringem principalmente ao software. Assim, o que se propõe nesse trabalho é um sistema supervisório de baixo custo, via TCP/IP, aplicado em máquinas elétricas, mas que pode ser generalizado na supervisão de outros tipos de sistemas, onde o monitoramento remoto é necessário. Além da exibição convencional dos parâmetros supervisionados através de gráficos e labels, o sistema permite também que esses sejam exibidos através de painéis eletrônicos, fornecendo assim, uma nova interface visual ao usuário com o intuito de facilitar sua supervisão. / The advent of internet simplified the comunication between remote systems, allowing the application of supervisory systems through the internet, mainly using the TCP/IP protocol. Although these systems have already been employed in industry, using the Programmer Logic Controller (PLC\'s) as driver, the implementation of any simple supervisory system could have an expensive cost, due to the prices of hardware and software employed. Moreover, the existing supervisory systems provide limited visual resources, restricted to the used software. So, this work presents a low cost supervisory system, via TCP/IP, applied to electric machines but it can be generalized in the supervision of differents types of systems, where the remote monitoring is necessary. Besides the conventional displaying of the monitored parameters through graphics and labels, the developed system allows the exhibition of the supervisioned parameters using eletronic panels, providing an alternative visual interface to the user so that their supervision becomes easier.

by Chung Ling Chi. / Thesis (M.Phil.)--Chinese University of Hong Kong, 2001. / Includes bibliographical references (leaves 74-76). / Abstracts in English and Chinese. / Chapter Chapter 1 --- Introduction --- p.1 / Chapter 1.1 --- Overview --- p.1 / Chapter 1.2 --- Motivation and Proposed Solution --- p.2 / Chapter 1.3 --- Organization of the Thesis --- p.4 / Chapter Chapter 2 --- Background --- p.5 / Chapter 2.1 --- Basics of Transmission Control Protocol --- p.5 / Chapter 2.1.1 --- Slow Start and Congestion Avoidance --- p.5 / Chapter 2.1.2 --- Fast Retransmit and Fast Recovery --- p.7 / Chapter 2.2 --- SACK TCP Mechanism --- p.8 / Chapter 2.2.1 --- SACK-permitted Option during Three-way Handshake --- p.8 / Chapter 2.2.2 --- SACK blocks in SACK Option --- p.9 / Chapter 2.2.3 --- Interpreting the SACK Option using Scoreboard --- p.10 / Chapter 2.2.4 --- Retransmission Strategy --- p.11 / Chapter 2.3 --- TCP Veno Mechanism --- p.13 / Chapter 2.3.1 --- Refined Additive Increase --- p.13 / Chapter 2.3.2 --- Refined Multiplicative Decrease --- p.14 / Chapter Chapter 3 --- SACK TCPVeno --- p.16 / Chapter 3.1 --- Distinguishing between Types of Packet Loss --- p.17 / Chapter 3.2 --- Refined Multiplicative Decrease --- p.21 / Chapter 3.2.1 --- Algorithm --- p.21 / Chapter 3.2.2 --- Recovery in Consecutive packet Losses --- p.22 / Chapter 3.2.3 --- Recovering Multiple Packet Losses within a Single Window --- p.26 / Chapter 3.3 --- Refined Additive Increase --- p.37 / Chapter 3.3.1 --- Algorithm --- p.37 / Chapter 3.3.2 --- Advantages --- p.40 / Chapter 3.4 --- Other Issues --- p.43 / Chapter 3.4.1 --- Two Side Modifications --- p.43 / Chapter Chapter 4 --- Experiments --- p.44 / Chapter 4.1 --- The Network Scenario --- p.44 / Chapter 4.1.1 --- Dummynet --- p.45 / Chapter 4.2 --- Experiment Results --- p.47 / Chapter 4.2.1 --- Single Connection --- p.47 / Chapter 4.2.1.1 --- Congestion Window Evolution --- p.47 / Chapter 4.2.1.2 --- Sending Rate and Throughput Evolution --- p.49 / Chapter 4.2.1.2.1 --- Impact of Packet Loss Rate Due to Lossy Link --- p.49 / Chapter 4.2.1.2.2 --- Impact of Buffering --- p.52 / Chapter 4.2.1.2.3 --- Impact of Propagation Delay --- p.57 / Chapter 4.2.2 --- Multiple Connections --- p.62 / Chapter 4.2.2.1 --- Fairness --- p.62 / Chapter 4.2.2.2 --- Compatibility --- p.67 / Chapter Chapter 5 --- Conclusion --- p.72 / Bibliography --- p.74

El objetivo de la presente tesis es implementar un laboratorio virtual para simular los mecanismos de transición vigentes en redes IPv4/IPv6, aplicándolos en el enrutamiento dinámico, y para facilitar su implementación en una red de área local con usuarios Windows. La metodología usada en este documento consiste en, por un lado, establecer un enlace a un repositorio remoto con el uso de la herramienta Apache Subversion. En donde, bajo la supervisión del Asesor de la tesis, se realiza el control de versiones de los escenarios virtuales y del documento de la tesis. Así mismo, a través de reuniones presenciales y virtuales con la herramienta de comunicación Skype, se verifica el avance del presente trabajo. Por otro lado, en cuanto al desarrollo de la tesis, en primer lugar, se diseña la topología de los escenarios para su posterior implementación a través de la lógica del lenguaje de programación de la herramienta de virtualización. En segundo lugar, se realiza la ejecución de pruebas de conectividad y eficiencia de los escenarios virtuales. Y, en tercer lugar, a partir del resultado de las pruebas obtenidas en los escenarios virtuales, se selecciona los mecanismos adecuados para lograr la migración de una red con clientes Windows a IPv6. Esto, sin afectar los servicios y la operatividad de la red. En el Capítulo I, se describe la necesidad de adoptar el protocolo IPv6. Se exponen las ventajas del protocolo IPv6 sobre IPv4. Además, se analiza los mecanismos de traducción, NAT64+DNS64, y de túneles, 6to4 y 6RD; se describe el protocolo ICMPv6, el papel que cumple en el direccionamiento dinámico IPv6 y en el sistema de nombres de dominio (DNS) en IPv6. Así mismo, se muestra la situación actual de la Red Académica Peruana RAAP y de los operadores de telecomunicaciones en la adopción de IPv6. En relación a la tendencia actual de los mecanismos de transición en las redes móviles, se describe el estado y uso de las tecnologías, como Dual- Stack y 464XLAT. Por otro lado, el primer aporte de la tesis se muestra en el Capítulo II. La implementación de los escenarios virtuales utilizando una herramienta desarrollada por la Universidad Politécnica de Madrid llamada VNUML. En donde, el único requerimiento que permite ejecutar los escenarios completos es adecuar un solo equipo anfitrión con capacidades de procesamiento y almacenamiento necesarios para la óptima ejecución de cada máquina virtual. En ese sentido, a través del diseño e implementación de cada escenario virtual que corresponde a los mecanismos de traducción NAT64 y DNS64, y los túneles 6to4 y 6RD, se demuestra la capacidad de la herramienta VNUML en el diseño, ejecución y comprobación de los aspectos teóricos señalados en el primer capítulo de esta tesis. El siguiente aporte de esta tesis se muestra en el Capítulo III. La migración de una red privada de usuarios Windows a IPv6. En primer lugar, se hace un despliegue dinámico de direcciones IPv6 a través de DHCPv6 + SLAAC. En segundo lugar, el acceso a los servicios e Internet IPv4 se realiza a través de NAT64 stateful y DNS64. Y, en tercer lugar, el acceso a Internet IPv6 se hace posible con el mecanismo NAT66. De esta manera, se demuestra que es factible desplegar IPv6 de forma transparente para los usuarios finales. / Tesis

Se plantea un problema de cambio de infraestructura y rediseño de red de comunicaciones en una empresa dedicada al servicio de transferencia electrónica de fondos interbancarios a través de una red de cajeros automáticos y al procesamiento y administración de tarjetas de débito y crédito. Tanto en el frente de los cajeros automáticos o terminales transaccionales, como en el frente de los computadores que autorizan las transacciones, se han ido experimentando y exigiendo cambios, los cuales no solo han involucrado nuevas tecnologías y aplicaciones, sino también la apertura a protocolos de comunicación como el TCP/IP con nuevos servicios y posibilidades, en un servicio financiero, que por mantener altos niveles de seguridad, mantenía protocolos de comunicación “heredados” como el X.25 y el SNA. En el presente documento se revisa la situación inicial de la red, los servicios y las necesidades del negocio, y la evolución de las redes de cajeros. Con el fin de conseguir los objetivos, se plantean propuestas de solución para dar soporte a las aplicaciones con protocolos “heredados” en una red IP, se revisan las alternativas técnico-económicas de enlaces de comunicación, las propuestas para la renovación de la infraestructura de comunicaciones y seguridad, y finalmente una serie de recomendaciones para la implantación y la migración a la red IP. Con un adecuado planeamiento e implantación de políticas de seguridad adecuadas, en una red privada, publica o compartida con otra institución, es posible conseguir una red de cajeros automáticos TCP/IP eficiente, segura, con alta disponibilidad, y capaz de brindar mayores servicios. / Tesis

近年來，使用者通過移動數據網路，如3G和LTE，連接到互聯網的數目急劇增加。眾所周知無線網路和移動數據網路展現的特點和有線網路有很大的不同。儘管如此，大多數移動應用程式的基本構建塊，即傳輸控制協議（TCP），在很大程度上仍是根植於有線網路。本論文通過廣泛的開展多個移動數據網路，包括3G，HSPA，最新的LTE網路的測試和實驗，探討TCP在現代移動數據網路的性能。儘管移動數據網路頻寬的迅速增加，我們的測量結果均顯示，現有的TCP實現在實踐中表現不佳，未能利用高速移動數據網路豐富的頻寬。這項工作解決TCP的性能限制，採用一種新的方法透明協議優化，通過在中間網路設備即時優化TCP，顯著提高TCP的吞吐量。具體來說，這項工作發展（一）一個新穎的機會傳輸算法克服TCP的流量控制的瓶頸;（二）一個傳輸速率控制演算法來解決TCP的拥塞控制的瓶頸;（三）一個新穎的投機重傳演算法，以提高TCP在重傳中的吞吐量;（四）用隨機模型來量化TCP吞吐量性能對移動網路資源利用率的影響;（五）一個新的隊列長度測量算法，為擁塞控制和網路監測打開一條新的途徑。另外，擬議的協議優化技術已全面實施，變成一個移動加速器裝置已經成功在三個不同的3G/LTE生產移動數據網路領域試用，實驗顯示TCP的吞吐量從48%增加至163%。在發明一種新的傳輸協議，或修改現有的TCP實施相比，所提出的方法不要求在用戶端/伺服器的主機現有的TCP實施任何修改，不需要重新配置伺服器或用戶端，並因此可以容易在現今的3G和4G移動網路部署，提高所有現有網路上運行在TCP之上的應用程式的吞吐量性能。 / The number of Internet users which are connected via mobile networks such as 3G and LTE has increased dramatically in recent years. It is well-known that wireless networks in general, and mobile data networks in particular, exhibit characteristics that are very different from their wired counterparts. Nevertheless, the fundamental building block of most Internet applications, namely the Transmission Control Protocol (TCP), is still largely rooted in wired networks. This dissertation investigate the performance of TCP over modern mobile data networks through extensive measurements and experiments carried out in multiple production data networks, ranging from 3G, HSPA, to the latest LTE networks. Despite the rapid increases in mobile network bandwidth, our measurements consistently reveal that existing TCP implementations perform sub-optimally in practice, failing to utilize the abundant bandwidth available in high-speed mobile networks. This work tackles the performance limitations of TCP using a novel approach - transparent protocol optimization, to significantly improve TCP’s throughput performance using on-the-fly protocol optimization carried out by an intermediate network device in-between the TCP end-hosts. Specifically, this work develops (i) a novel opportunistic transmission algorithm to overcome the TCP’s flow control bottleneck; (ii) a transmission rate control algorithm to tackle TCP’s congestion control bottleneck; (iii) a new opportunistic retransmission algorithm to improve TCP’s performance during packet loss recovery; (iv) a stochastic model to quantify the impact of TCP throughput performance on mobile network capacity; and (v) a new queue length estimation algorithm which opens a new avenue for congestion control and network monitoring. In addition, the proposed protocol optimization techniques have been fully implemented into a mobile accelerator device which has been successfully field trialed in three different production 3G/LTE mobile networks, consistently increasing TCP’s throughput by 48% to 163%. In contrast to inventing a new transport protocol or modifying an existing TCP implementation, the proposed approach does not require any modification to the existing TCP implementation at the client/server hosts, does not require any reconfiguration of the server or client, and hence can be deployed readily in today’s 3G and 4G mobile networks, raising the throughput performance of all existing network applications running atop TCP. / Detailed summary in vernacular field only. / Liu, Ke. / Thesis (Ph.D.)--Chinese University of Hong Kong, 2013. / Includes bibliographical references (leaves 166-174). / Electronic reproduction. Hong Kong : Chinese University of Hong Kong, [2012] System requirements: Adobe Acrobat Reader. Available via World Wide Web. / Abstracts also in Chinese. / Abstract --- p.2 / Acknowledgement --- p.6 / Chapter 1 --- p.1 / Introduction --- p.1 / Chapter 1.1 --- Motivation --- p.1 / Chapter 1.2 --- Contributions --- p.3 / Chapter 1.3 --- Structure of the Thesis --- p.6 / Chapter 2 --- p.9 / Flow and Congestion Control --- p.9 / Chapter 2.1 --- TCP Performance Bottlenecks --- p.9 / Chapter 2.2 --- Background and related works --- p.16 / Chapter 2.3 --- Transparent Protocol Optimization --- p.20 / Chapter 2.3.1 --- Opportunistic Transmission --- p.20 / Chapter 2.3.2 --- Transmission Rate Control --- p.22 / Chapter 2.3.3 --- Lost Packet Recovery --- p.27 / Chapter 2.4 --- Modeling and Analysis --- p.28 / Chapter 2.4.1 --- Background and Assumptions --- p.28 / Chapter 2.4.2 --- Queue Length at the Radio Interface --- p.31 / Chapter 2.4.3 --- Queue Length Bounds --- p.38 / Chapter 2.4.4 --- Guaranteeing Full Bandwidth Utilization --- p.45 / Chapter 2.4.5 --- Link Buffer Size Requirement --- p.47 / Chapter 2.5 --- Performance Evaluation --- p.53 / Chapter 2.5.1 --- Parameter Tuning --- p.53 / Chapter 2.5.2 --- Bandwidth Efficiency --- p.56 / Chapter 3 --- p.62 / Packet Loss Recovery --- p.62 / Chapter 3.1 --- Introduction --- p.62 / Chapter 3.2 --- TCP Loss Recovery Revisited --- p.64 / Chapter 3.2.1 --- Standard TCP Loss Recovery Algorithm --- p.64 / Chapter 3.2.2 --- Loss Recovery Algorithm in Linux --- p.66 / Chapter 3.2.3 --- Loss Recovery Algorithm in A-TCP --- p.67 / Chapter 3.3 --- Efficiency of TCP Loss Recovery Algorithms --- p.68 / Chapter 3.3.1 --- Standard TCP Loss Recovery Algorithm --- p.70 / Chapter 3.3.2 --- TCP Loss Recovery in Linux --- p.72 / Chapter 3.3.3 --- Loss Recovery Algorithm Used in A-TCP --- p.72 / Chapter 3.3.4 --- Discussions --- p.73 / Chapter 3.4 --- Opportunistic Retransmission --- p.74 / Chapter 3.4.1 --- Applications and Performance Analysis --- p.76 / Chapter 3.4.2 --- Bandwidth Utilization During Loss Recovery --- p.78 / Chapter 3.5 --- Experimental Results --- p.81 / Chapter 3.5.1 --- Model Validation --- p.85 / Chapter 3.5.2 --- Impact of Loss Recovery Phase on TCP Throughput --- p.85 / Chapter 3.5.3 --- A-TCP with Opportunistic Retransmission --- p.86 / Chapter 3.6 --- Summary --- p.87 / Chapter 4 --- p.89 / Impact on Mobile Network Capacity --- p.89 / Chapter 4.1 --- Introduction --- p.89 / Chapter 4.2 --- Background and Related Work --- p.91 / Chapter 4.2.1 --- TCP Performance over Mobile Data Networks --- p.91 / Chapter 4.2.2 --- Modeling of Mobile Data Networks --- p.92 / Chapter 4.3 --- System Model --- p.94 / Chapter 4.3.1 --- Mobile Cell Bandwidth Allocation --- p.95 / Chapter 4.3.2 --- Markov Chain Model --- p.96 / Chapter 4.3.3 --- Performance Metric for Mobile Internet --- p.98 / Chapter 4.3.4 --- Protocol-limited Capacity Loss --- p.100 / Chapter 4.3.5 --- Channel-limited Capacity Loss --- p.101 / Chapter 4.4 --- Performance Evaluation --- p.102 / Chapter 4.4.1 --- Service Response Time --- p.103 / Chapter 4.4.2 --- Network Capacity Loss --- p.105 / Chapter 5 --- p.114 / Mobile Link Queue Length Estimation --- p.114 / Chapter 5.1 --- Introduction --- p.115 / Chapter 5.2 --- Sum-of-Delay (SoD) algorithm Revisited --- p.117 / Chapter 5.2.1 --- Queue Length and Link Buffer Size Estimation --- p.117 / Chapter 5.2.2 --- A Bound on Estimation Error --- p.120 / Chapter 5.2.3 --- Impact of Uplink Delay Variations --- p.122 / Chapter 5.3 --- Uplink Delay Variation Compensation --- p.127 / Chapter 5.3.1 --- Exploiting the TCP Timestamp Option --- p.127 / Chapter 5.3.2 --- TCP Timestamp Granularity --- p.130 / Chapter 5.4 --- Performance Evaluation --- p.131 / Chapter 5.4.1 --- Link buffer size estimation under uplink delay variations --- p.132 / Chapter 5.4.2 --- Queue length estimation under uplink delay variations --- p.136 / Chapter 5.5 --- Summary --- p.136 / Chapter 6 --- p.139 / Summary and Future Works --- p.139 / Chapter 6.1 --- Transparent Protocol Optimization --- p.139 / Chapter 6.2 --- Cross-Layer Modeling and Optimization of Mobile Networks --- p.141 / Chapter Appendix A. --- Derivation of Equations (2.24) and (2.25) --- p.143 / Chapter Appendix B. --- Proof of Theorem 2.1 --- p.145 / Chapter Appendix C. --- for Proof of Theorem 2.2 --- p.147 / Chapter Appendix D. --- for Proof of Theorem 2.3 --- p.150 / Chapter Appendix E. --- for Proof of Theorem 2.4 --- p.151 / Chapter Appendix F. --- for Proof of Theorem 2.5 --- p.152 / Chapter Appendix G. --- for Proof of Theorem 2.6 --- p.153 / Chapter Appendix H. --- for Proof of Theorem 2.7 --- p.156 / Chapter Appendix I. --- for Proof of Theorem 2.8 --- p.157 / Chapter Appendix J. --- for Proof of Theorem 3.2 --- p.161 / Chapter Appendix K. --- for Theorem 3.4 --- p.163 / Chapter Appendix H. --- for Theorem 3.5 --- p.164 / Bibliography --- p.166

by Fu Chengpeng. / Thesis (Ph.D.)--Chinese University of Hong Kong, 2001. / Includes bibliographical references (p. 102-119). / Electronic reproduction. Hong Kong : Chinese University of Hong Kong, [2012] System requirements: Adobe Acrobat Reader. Available via World Wide Web. / Mode of access: World Wide Web. / Abstracts in English and Chinese.