Dissertations / Theses on the topic 'IP (Computer network protocol) Data transmission systems'

An increasing number of network applications use reliable transport protocols. Applications with constant data transmission recover from loss without major performance disruption, however, applications that send data sporadically, in small packets, also called thin-streams, experience frequently high latencies due to 'Bufferbloat', that reduce the application performance. Active Queue Management mechanisms were proposed to dynamically manage the queues in routers by dropping packets early and reduce these, hence reducing latency. While their deployment to the internet remains an open issue, the proper investigation into how their functioning mechanism impacts latency is the main focus of this work and research questions have been devised to investigate the AQM impact on latency. A range of AQM mechanisms has been evaluated by the research, exploring performance of the methods for latency sensitive network applications. This has explored new single queue AQM mechanisms such as Controlled Delay (CODEL) and Proportional Integral Enhanced (PIE) and Adaptive RED (ARED). The evaluation has shown great improvements in queuing latency when AQM are used over a range of network scenarios. Scheduling AQM algorithms such as FlowQueue CODEL (FQ-CODEL) isolates traffic and minimises the impact of Bufferbloat on flows. The core components of FQ-CODEL, still widely misunderstood at the time of its inception, have been explained in depth by this study and their contribution to reducing latency have been evaluated. The results show significant reductions in queuing latency for thin streams using FQ-CODEL. When TCP is used for thin streams, high application latencies can arise when there are retransmissions, for example after dropping packets by an AQM mechanism. This delay is a result of TCP's loss-based congestion control mechanism that controls sender transmission rate following packet loss. ECN, a marking sender-side improvement to TCP reduces applicationlayer latency without disrupting the overall network performance. The thesis evaluated the benefit of using ECN using a wide range of experiments. The findings show that FQ-CODEL with ECN provides a substantial reduction of application latency compared to a drop-based AQM. Moreover, this study recommends the combination of FQ-CODEL with other mechanisms, to reduce application latency. Mechanisms such as ABE, have been shown to increase aggregate throughput and reduce application latency for thin-stream applications.

Orientadores: Maurício Ferreira Magalhães, Christian Rodolfo Esteve Rothenberg
Dissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Elétrica e de Computação
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Resumo: Os roteadores atuais implementam uma arquitetura verticalmente integrada composta de uma camada de software e um hardware proprietários. Este modelo resulta em soluções de alto custo e inviabiliza a experimentação de novas idéias. Em contrapartida, existem alternativas de alta flexibilidade baseadas em software e, consequentemente, de baixo custo. Entretanto, essas soluções apresentam baixo desempenho. Motivado pela disponibilidade de uma API aberta para programação do plano de encaminhamento (ex. OpenFlow), esta dissertação apresenta uma proposta de arquitetura de roteamento IP com separação de planos. Trata-se de uma abordagem que procura combinar o alto desempenho de hardwares de prateleira (commodities) com a flexibilidade de uma pilha de roteamento executada remotamente em computadores de uso geral. O grande desafio é garantir confiabilidade, escalabilidade e desempenho à rede, a partir de um controle remoto e centralizado sobre uma arquitetura que permita maior flexibilidade no mapeamento entre os elementos de controle e encaminhamento. O resultado corresponde a uma nova proposta de roteamento IP com perspectivas promissoras do ponto de vista do custo e da flexibilidade. Com o objetivo de avaliar a arquitetura proposta foi desenvolvido um protótipo com base em uma versão simplificada do modelo. Os resultados da avaliação apresentados nesta dissertação comprovam a viabilidade da arquitetura
Abstract: Today's networking gear follows the model of computer mainframes, where closed source software runs on proprietary hardware. This approach results in expensive solutions and prevents equipment owners to put new ideas into practice. In contrast, recent alternatives of highly flexible softwarebased routers promise low cost and programmability at the expense of low performance. Motivated by the availability of an open API to control packet forwarding engines (i.e., OpenFlow), we propose a commodity IP routing architecture that combines the line-rate performance of commercial hardware with the flexibility of open source routing stacks (remotely) running on general-purpose computers. The challenge is to ensure reliability, scalability and performance to a network running a remote and centralized control plane architecture that allows a flexible mapping between the control and forwarding elements. The outcome is a novel point in the design space of cost-effective IP routing solutions with far-reaching implications. The initial experimental evaluation of our prototype implementation validates the feasibility of the design
Mestrado
Engenharia de Computação
Mestre em Engenharia Elétrica

The best MPhil thesis in the Faculties of Dentistry, Engineering, Medicine and Science (University of Hong Kong), Li Ka Shing Prize,2010-11.
published_or_final_version
Electrical and Electronic Engineering
Master
Master of Philosophy

In the past it has been convenient to adopt existing long haul network (LHN) protocols for use in local area networks (LANs). However, due to the different operating parameters that exist between these two types of networks, it is not possible for a LHN protocol to fully exploit the characteristics of a LAN. Thus, the need arises for a protocol designed specifically for use in a LAN environment. LNTP is one such transport level protocol. It was designed for exclusive use in LANs, and thus does not incorporate those features which are not relevant to a LAN environment. The result of this is a simpler and more efficient protocol. As well, LNTP employs a novel deferred flow control strategy which minimizes the time that a transmitting process will be blocked. This thesis examines the implementation of LNTP in the 4.2 BSD UNIX operating system. Various measurements are taken, and LNTP's performance is compared to that of TCP/IP, a LHN protocol which is often used in LAN environments. Several formulas are developed to determine the optimum values for various LNTP parameters, and these theoretical results are compared to the experimentally observed values. We conclude that LNTP does indeed outperform TCP/IP. However, due to the overhead of the non-LNTP specific protocol layers, this improvement is not as great as it might be. Nonetheless, LNTP proves itself to be a viable replacement for TCP/IP.
Science, Faculty of
Computer Science, Department of
Graduate

Thesis (M. Phil.)--Hong Kong University of Science and Technology, 2002.
Includes bibliographical references (leaves 58-61). Also available in electronic version. Access restricted to campus users.

The Session Initiation Protocol (SIP) has in recent years become a popular protocol for the exchange of text, voice and video over IP networks. This thesis proposes the use of a class of structured peer to peer protocols - commonly known as Distributed Hash Tables (DHTs) - to provide a SIP overlay with services such as end-point location management and message relay, in the absence of traditional, centralised resources such as SIP proxies and registrars. A peer-to-peer layer named OverCord, which allows the interaction with any specific DHT protocol via the use of appropriate plug-ins, was designed, implemented and tested. This layer was then incorporated into a SIP user agent distributed by NIST (National Institute of Standards and Technology, USA). The modified user agent is capable of reliably establishing text, audio and video communication with similarly modified agents (peers) as well as conventional, centralized SIP overlays.

Bibliography: leaves 128-135. Investigates the behaviour of TCP bulk file transfer application sessions in a broadband access environment. Introduces some concepts for evaluating network behaviour: a path instability parameter for analyzing different TCP connections; a minimum RTT delay and a minimum typical path for estimating path characteristics between a client and application servers.

Digital audio networks allow multiple channels of audio to be streamed between devices. This eliminates the need for many different cables to route audio between devices. An added advantage of digital audio networks is the ability to configure and control the networked devices from a common control point. Common control of networked devices enables a sound engineer to establish and destroy audio stream connections between networked devices that are distances apart. On a digital audio network, an audio transport technology enables the exchange of data streams. Typically, an audio transport technology is capable of transporting both control messages and audio data streams. There exist a number of audio transport technologies. Some of these technologies implement data transport by exchanging OSI/ISO layer 2 data frames, while others transport data within OSI/ISO layer 3 packets. There are some approaches to achieving interoperability between devices that utilize different audio transport technologies. A digital audio device typically implements an audio control protocol, which enables it process configuration and control messages from a remote controller. An audio control protocol also defines the structure of the messages that are exchanged between compliant devices. There are currently a wide range of audio control protocols. Some audio control protocols utilize layer 3 audio transport technology, while others utilize layer 2 audio transport technology. An audio device can only communicate with other devices that implement the same control protocol, irrespective of a common transport technology that connects the devices. The existence of different audio control protocols among devices on a network results in a situation where the devices are unable to communicate with each other. Furthermore, a single control application is unable to establish or destroy audio stream connections between the networked devices, since they implement different control protocols. When an audio engineer is designing an audio network installation, this interoperability challenge restricts the choice of devices that can be included. Even when audio transport interoperability has been achieved, common control of the devices remains a challenge. This research investigates protocol command translation as a means to enable interoperability between networked audio devices that implement different audio control protocols. It proposes the use of a command translator that is capable of receiving messages conforming to one protocol from any of the networked devices, translating the received message to conform to a different control protocol, then transmitting the translated message to the intended target which understands the translated protocol message. In so doing, the command translator enables common control of the networked devices, since a control application is able to configure and control devices that conform to different protocols by utilizing the command translator to perform appropriate protocol translation.

Emerging wireless networks are characterized by increased heterogeneity in wireless access technologies as well as increased peer-to-peer communication among wireless hosts. The heterogeneity among wireless access interfaces mainly exists because of the fact that different wireless technologies deliver different performance trade-offs. Further, more and more infrastructure-less wireless networks such as ad-hoc networks are emerging to address several application scenarios including military and disaster recovery. These infrastructure-less wireless networks are characterized by the peer-to-peer communication model. In this thesis, we propose transport protocols that tackle the challenges that arise due to the above-mentioned properties of state-of-the-art wireless data networks. The main contributions of this work are as follows: 1. We determine the ideal nature and granularity of transport adaptation for efficient operation in heterogeneous wireless data networks by performing comprehensive experimental analysis. We then design and implement a runtime adaptive transport framework, *TP, which accommodates the capabilities of the ideal transport adaptation solution. 2. We prove that conversational transport protocols are not efficient under peer-to-peer wireless data networks. We then design and implement NCTP which is a non-conversational transport protocol.

Thesis (Ph.D.) -- University of Glasgow, 2001.
Includes bibliographical references (p.i-xv). Print version also available. Mode of access : World Wide Web. System requirments : Adobe Acrobat reader reuired to view PDF document.

The algorithms which control network routing are specific to the network because the algorithms are designed to take advantage of that network's topology. The "goodness" of a network includes such criteria as a simple routing algorithm and a simple routing algorithm would increase the use of the shuffle-exchange network.

Il lavoro è stato suddiviso in tre macro-aree. Una prima riguardante un'analisi teorica di come funzionano le intrusioni, di quali software vengono utilizzati per compierle, e di come proteggersi (usando i dispositivi che in termine generico si possono riconoscere come i firewall). Una seconda macro-area che analizza un'intrusione avvenuta dall'esterno verso dei server sensibili di una rete LAN. Questa analisi viene condotta sui file catturati dalle due interfacce di rete configurate in modalità promiscua su una sonda presente nella LAN. Le interfacce sono due per potersi interfacciare a due segmenti di LAN aventi due maschere di sotto-rete differenti. L'attacco viene analizzato mediante vari software. Si può infatti definire una terza parte del lavoro, la parte dove vengono analizzati i file catturati dalle due interfacce con i software che prima si occupano di analizzare i dati di contenuto completo, come Wireshark, poi dei software che si occupano di analizzare i dati di sessione che sono stati trattati con Argus, e infine i dati di tipo statistico che sono stati trattati con Ntop. Il penultimo capitolo, quello prima delle conclusioni, invece tratta l'installazione di Nagios, e la sua configurazione per il monitoraggio attraverso plugin dello spazio di disco rimanente su una macchina agent remota, e sui servizi MySql e DNS. Ovviamente Nagios può essere configurato per monitorare ogni tipo di servizio offerto sulla rete.

Active queue management (AQM) techniques for congestion control in Internet Protocol (IP) networks have been designed using both heuristic and analytical methods. But so far, there has been found no AQM scheme designed in the realm of stochastic optimization. Of the many options available in this arena, the gradient-based stochastic approximation method using Infintesimal Perturbation Analysis (IPA) gradient estimators within the Stochastic Fluid Model (SFM) framework is very promising. The research outlined in this thesis provides the theoretical basis and foundational layer for the development of IPA-based AQM schemes. Algorithms for computing the IPA gradient estimators for loss volume and queue workload were derived for the following cases: a single-stage queue with instantaneous, additive loss-feedback, a single-stage queue with instantaneous, additive loss-feedback and an unresponsive competing flow, a single-stage queue with delayed, additive loss-feedback, and a multi-stage tandem network of $m$ queues with instantaneous, additive loss-feedback. For all cases, the IPA gradient estimators were derived with the control parameter, $ heta$, being the buffer-limits of the queue(s). For the single-stage case and the multi-stage case with instantaneous, additive loss-feedback, the IPA gradient estimators for when the control parameter, $ heta$, is the loss-feedback constant, were also derived. Sensitivity analyses and optimizations were performed with control parameter, $ heta$, being the buffer-limits of the queue(s), as well as the loss-feedback constant.

The growth of Asynchronous Transfer Mode (ATM) was considered to be the ideal carrier of the high bandwidth applications like video on demand and multimedia e-learning. ATM emerged commercially in the beginning of the 1990's. It was designed to provide a different quality of service at a speed up 100 Gbps for both real time and non real time application. The turn of the 90's saw a variety of technologies being developed. This project analyzes these technologies, compares them to the Asynchronous Transfer Mode and assesses the future of ATM.

O controle de congestionamento padrão do TCP apresenta vários problemas; ele não consegue distinguir se o pacote foi perdido por falha no enlace ou por descarte de pacotes devido a um congestionamento de rede (se a falha foi no enlace não há necessidade de ativar os mecanismos de controle de congestionamento); e o correto ajuste de sua taxa de transmissão requer informação de perdas de pacotes. Neste trabalho é apresentado o new generalized window advertising (NGWA), que é um novo esquema de controle de congestionamento para o TCP. O NGWA traz informações da banda disponível da infraestrutura de rede para os pontos finais da conexão TCP. Seu desempenho foi comparado com TCP New Reno, RED e o TCP padrão via simulações com o software NS-3, considerando topologias de rede largamente citadas na literatura. O NGWA foi, também, implementado e testado no Linux (versão 2.6.34). O novo método demonstrou ser superior aos comparados, apresentando uma operação mais estável, melhor justiça e menor taxa de perda de pacotes, considerando o elenco de testes realizados.
The TCP congestion control mechanism in standard implementations presents several problems; he cannot distinguish if the packet was lost by link failure or by congestion in the net (if the fault was in the link there is no need to active congestion control mechanisms); and the right adjust of your transmission rate requires information from packet loss. This work presents the new generalized window advertising (NGWA), which is a new congestion control scheme for TCP. The NGWA provides information considering the available bandwidth of the network infrastructure to the endpoints of the TCP connection. Results obtained by the NGWA approach were compared with those from TCP New Reno, RED, and standard TCP (using the network simulator NS-3), considering network topologies widely cited in the literature. A NGWA Linux implementation is also presented. The new method proved to be superior when compared with the traditional approaches, presenting a more stable operation, better fairness and lower packet loss, considering the set of tests carried out.

Orientador: Mauricio Ferreira Magalhães
Dissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Eletrica e de Computação
Made available in DSpace on 2018-08-04T17:32:51Z (GMT). No. of bitstreams: 1 Sazima_Ricardo_M.pdf: 1107563 bytes, checksum: a1179beaec699c0dcd87b45284e4a99c (MD5) Previous issue date: 2004
Resumo: Dados a banda de transmissão atualmente disponível, o protocolo (IP) utilizado para transmitir a maior parte de tráfego e a quantidade de tráfego e seus requisitos de aos, a Engenharia de Tráfego (TE, de Traffic Engineering) tomou-se um componente cada vez mais importante nas redes de comunicações. O padrão do IETF para enncaminhamento roteamento entitulado Multi-Protocol Label Switching (MPLS) preenche lacunas importantes neste cenário e é peça chave das metodologias de TE mais sofisticadas. Nesta dissertação é apresentado um modelo genérico de TE e uma proposta para uma Plataforma para Experimentos com MPLS (PLEX MPLS) que permite que o usuário defina, execute, e analise configurações de Engenharia de Tráfego com MPLS em uma rede IP. Os principais objetivos da PLEX MPLS são: . Estudo da tecnologia MPLS e suporte a outros trabalhos em andamento no contexto do grupo de estudos em MPLS do DCA . Experimentos com engenharia de tráfego com MPLS: validação das tecnologias e conceitos relacionados . Experimentos didáticos em disciplinas de laboratório de redes. Os principais conceitos de TE e MPLS são apresentados como referência, bem como uma descrição do NIST Switch, a plataforma MPLS escolhida para este trabalho. A análise, projeto e implementação da PLEX MPLS são apresentados, pois formam parte significativa do trabalho desenvolvido. A PLEX não só utiliza, como também estende as funcionalidades oferecidas pelo NIST Switch a fim de oferecer um esquema de TE mais completo e eficiente. Na fase de análise, vemos quais os principais requisitos para a implementação desta plataforma, seguindo uma metodologia de Engenharia de Software. Na fase de projeto, vemos as soluções propostas para os problemas identificados na fase de análise e temos uma especificação dos componentes a serem implementados. Na fase seguinte, discutimos a implementação das principais características dos componentes da PLEX, justificando as decisões tomadas. Para validar a implementação da PLEX de acordo com sua proposta, foram realizados alguns experimentos em uma rede de testes com tráfego real. A execução destes experimentos é descrita e seus resultados analisados. Os resultados obtidos assinalam claramente a importância e utilidade de esquemas de TE baseada em MPLS. Uma interessante metodologia para TE, compilada a partir de várias propostas, é apresentada. Finalmente, apontam-se caminhos a seguir em um trabalho futuro de refinamento da PLEX
Abstract: Given the bandwidth currently available, the protocol (IP) used to transmit most Internet traffic, the quantity of traffic produced and its QoS requirements, Traffic Engineering (TE) has become an increasingly important component of communications networks. IETF's standard for forwarding/routing, which is entitled Multi-Protocol Label Switching (MPLS), presents important solutions in this scenario playing a major role in more sophisticated TE methodologies. This work presents a generic methodology for TE and a proposal for a Platform for Experiments with MPLS (PLEX MPLS) which allows the user to define, execute and analyze Traffic Engineering configurations with MPLS in an IP network. The main goals of PLEX MPLS are: . Study of the M PLS technology and support of other ongoing works with M PLS in the DCA . Traffic Engineering experiments with MPLS: validation of the related concepts and technologies . Support of didactic experiments in academic disciplines. The main concepts of MPLS and TE are presented as reference, as well as a brief description of NIST Switch, the MPLS software chosen for the PLEX MPLS implementation. The analysis, project and implementation of PLEX MPLS are presented, since are significant part of the developed work. PLEX not only uses, but also extends NIST Switch functionalities to offer a more complete and efficient TE scheme. In the analysis phase the main requirements for the PLEX implementation are specified, following a well-known Software Engineering methodology. The solutions found for the problems identified in the analysis phase are presented in the project phase alongside with a specification of the components that will be implemented. In the next phase, the implementation of PLEX is discussed focusing on the most important characteristics of PLEX components and justifying the implementation ecisions. In order to validate PLEX implementation and its proposal, some experiments were made in a test network with reallive traffic. These experiments are described and its results analyzed. The results obtained clearly indicate the importance and utility of TE schemes based on MPLS. Also an interesting TE methodology compiled from several proposals is presented. Finally, possible improvements and future work on PLEX MPLS are indicated.
Mestrado
Engenharia de Computação
Mestre em Engenharia Elétrica

The Transmission Control Protocol (TCP) is the most widely used transport protocol in the Internet. TCP is a reliable transport protocol that is tuned to perform well in wired networks where packet losses are mainly due to congestion. Wireless channels are characterized by losses due to transmission errors and handoffs. TCP interprets these losses as congestion and invokes congestion control mechanisms resulting in degradation of performance. TCP is usually layered over the Internet protocol (lP) at the network layer. JP is not reliable and does not provide for any Quality of Service (QoS). The Internet Engineering Task Force (IETF) has provided two techniques for providing QoS in the Internet. These include Integrated Services (lntServ) and Differentiated Services (DiffServ). IntServ provides flow based quality of service and thus it is not scalable on connections with large flows. DiffServ has grown in popularity since it is scalable. A packet in a DiffServ domain is classified into a class of service according to its contract profile and treated differently by its class. To provide end-to-end QoS there is a strong interaction between the transport protocol and the network protocol. In this dissertation we consider the performance of the TCP over a wireless channel. We study whether the current TCP protocols can deliver the desired quality of service faced with the challenges they have on wireless channel. The dissertation discusses the methods of providing for QoS in the Internet. We derive an analytical model for TCP protocol. It is extended to cater for the wireless channel and then further differentiated services. The model is shown to be accurate when compared to simulation. We then conclude by deducing to what degree you can provide the desired QoS with TCP on a wireless channel.
Thesis (M.Sc.Eng.)-University of Natal, Durban, 2001.

Using the approach of Coloured Petri nets (CPNs) and automata theory, this thesis shows how to formalise the service provided by the Transmission Control Protocol (TCP) and verify TCP Connection Management, an essential part of TCP. Most of the previous work on modelling and analysing TCP Connection Management is based on early versions of TCP, which are different from the current TCP specification. Also the scope is mainly confined to the connection establishment procedure, while the release procedure is either simplified or omitted from investigation. This thesis extends prior work by verifying a detailed model of TCP Connection Management. In defining the TCP service, the set of service primitives and their sequencing constraints are specified at each service access point.
thesis (PhDComputerSystemsEng)--University of South Australia, 2004.

This dissertation investigates the use of a hardware mechanism called Eager Data Transfer (EDT) for achieving the reduction of communication latency for user-level network protocol. To reach the goal, the dissertation addresses the following research issues. First, the development of a communication system performance evaluation tool called Linux/SimOS is presented. Linux/SimOS provides a full system profiling capability to allow measurement at various level including hardware, operating system, and application. Second, the performance analysis of network protocols is presented. For the assessment of overhead related to network protocol operation, Linux/SimOS was used to perform the detailed latency measurements for TCP/IP, UDP/IP, and M-VIA network protocols. Finally, EDT is proposed for reducing communication latency. Since the data transfer time constitutes a significant portion of overall communication latency, the reduction of data transfer time leads to low communication latency. EDT is based on cache coherence interface hardware for reducing data transfer overhead during network protocol operation. Our simulation result shows that EDT is very effective in attaining low communication latency compared to the DMA-based approaches.
Graduation date: 2005

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

OFS provisions bandwidth in the granularity of one wavelength. With such a coarse granularity, most applications including video download, HDTV, 3D movie, and 3D TV etc. will have very short flow sizes, in the order of seconds or even sub-second, which brings challenges to the utilization efficiency of bandwidth capacity. In this thesis we study the performance of OFS for short flows. The constraint of network resources is investigated. The effect of destination and path blocking is studied. A distributed scheduling protocol called Multi-Destination Control Protocol (MDCP) is proposed to deal with such constraint. Both single wavelength and multi-wavelength configurations are studied and characterized. Simulation results demonstrate that MDCPcan improve the OFS network throughput significantly and can be as much as eighty to one hundred percent for a single-wavelength OFS network. Even for an OFS network with four wavelengths, the throughput improvement can still approach 40%.
The Internet traffic has been growing tremendously. China Telecom predicts that the compound annual growth rate of IP traffic for the next decade is at 56% - 80% and the backbone capacity will grow by another two orders of magnitudes. Furthermore, the power consumption incurred by the next generation of huge electronic IP packet switching routers in the backbone will exceed gigawatts. In view of the grave enviromnental concerns, there is a great need for a more efficient way of transporting and switching the bits. This thesis investigates a new all-optical networking technology called optical flow switching (OFS). OFS bypasses electronic routers, and provides end-to-end transparent connections, thus taking full advantage of the enormous transmission capacity of optical networks and enjoying the extremely low error rate of transparent data transmission. The most important point about OFS is that it reduces the electrical power consumption by off-loading the huge electronic routers, which could be a major constraint for future Internet growth. Unlike many other exotic all-optical switching technologies, OFS is immediately deployable using the current optical technologies, Therefore OFS is very attractive for the next generation optical networks.
Qian, Zhengfeng.
Adviser: Kwok-wai Cheung.
Source: Dissertation Abstracts International, Volume: 73-04, Section: B, page: .
Thesis (Ph.D.)--Chinese University of Hong Kong, 2011.
Includes bibliographical references (leaves 113-118).
Electronic reproduction. Hong Kong : Chinese University of Hong Kong, [2012] System requirements: Adobe Acrobat Reader. Available via World Wide Web.
Electronic reproduction. [Ann Arbor, MI] : ProQuest Information and Learning, [201-] System requirements: Adobe Acrobat Reader. Available via World Wide Web.
Abstract also in Chinese.

by Chan Ho Chow.
Thesis (M.Phil.)--Chinese University of Hong Kong, 2001.
Includes bibliographical references (leaves 65-67).
Abstracts in English and Chinese.
Chapter Chapter 1 --- Introduction --- p.1
Chapter Chapter 2 --- Background --- p.5
Chapter 2.1. --- Packet Loss in the Internet --- p.5
Chapter 2.2. --- Shared Internet --- p.7
Chapter 2.3. --- Video Streaming --- p.10
Chapter 2.4. --- Real-time video transmission criteria --- p.11
Chapter 2.5. --- Existing Video Streaming Protocol --- p.13
Chapter Chapter 3 --- Transmission Control Protocol (TCP) --- p.16
Chapter Chapter 4 --- Variable-rate Streaming TCP (VarS-TCP) --- p.22
Chapter 4.1. --- General Idea --- p.22
Chapter 4.2. --- Assumptions --- p.25
Chapter 4.3. --- VarS-TCP Algorithm --- p.26
Chapter 4.3.1. --- Connection Initialization --- p.26
Chapter 4.3.2. --- Normal Data Transfer --- p.27
Chapter 4.4. --- Skipping packets in TCP --- p.32
Chapter 4.4.1. --- Types of skipped packet --- p.32
Chapter 4.4.2. --- Acknowledging skipped packets --- p.34
Chapter 4.4.3. --- Maintaining Normal Data Flow --- p.35
Chapter 4.4.4. --- Congestion Control --- p.37
Chapter 4.4.5. --- Packets skipped by receiver --- p.41
Chapter 4.5. --- Rebuffering --- p.44
Chapter Chapter 5 --- Simulation Result --- p.45
Chapter 5.1. --- Accumulating Data --- p.46
Chapter 5.2. --- Delay Constraints --- p.48
Chapter 5.3. --- Adapting network situation --- p.50
Chapter 5.4. --- Sharing bandwidth with TCP --- p.52
Chapter 5.5. --- Random Traffic --- p.58
Chapter 5.6. --- Effect of packet skip threshold --- p.59
Chapter 5.7. --- Effect of round-trip-time --- p.61
Chapter Chapter 6 --- Conclusion and Future Works --- p.63
Bibliography --- p.65

Wan, Wing San.
"September 2010."
Thesis (M.Phil.)--Chinese University of Hong Kong, 2010.
Includes bibliographical references (p. 53-55).
Abstracts in English and Chinese.
Acknowledgements --- p.ii
Abstract --- p.iii
摘要 --- p.iv
Contents --- p.v
Chapter Chapter 1 --- INTRODUCTION --- p.1
Chapter Chapter 2 --- BACKGROUND AND RELATED WORK --- p.4
Chapter 2.1 --- Sender-receiver-based approaches --- p.4
Chapter 2.2 --- Sender-based approaches --- p.5
Chapter 2.3 --- Receiver-based approaches --- p.6
Chapter Chapter 3 --- TCP FLOW CONTROL REVISITED --- p.8
Chapter Chapter 4 --- OPPORTUNISTIC TRANSMISSION --- p.12
Chapter 4.1 --- Link bandwidth estimation --- p.16
Chapter 4.2 --- Reception rate estimation --- p.18
Chapter 4.3 --- Transmission scheduling --- p.19
Chapter 4.4 --- Performance --- p.21
Chapter Chapter 5 --- Local Retransmission --- p.23
Chapter 5.1 --- The blackout period --- p.24
Chapter 5.2 --- Proactive retransmission --- p.28
Chapter 5.3 --- Performance --- p.30
Chapter Chapter 6 --- Loss Event Suppression --- p.31
Chapter 6.1 --- RTT modulation --- p.32
Chapter 6.2 --- Performance --- p.35
Chapter Chapter 7 --- Fairness --- p.37
Chapter 7.1 --- Packet forwarding --- p.37
Chapter 7.2 --- Non-uniform bandwidth allocation --- p.41
Chapter Chapter 8 --- EXPERIMENTS --- p.43
Chapter 8.1 --- Experiment setup --- p.43
Chapter 8.2 --- Packet loss --- p.44
Chapter 8.3 --- Unaccelerated TCP throughput --- p.45
Chapter 8.4 --- Accelerated TCP throughput --- p.46
Chapter 8.5 --- Fairness --- p.47
Chapter 8.6 --- Mobile handset performance --- p.47
Chapter Chapter 9 --- FUTURE WORK --- p.49
Chapter 9.1 --- Dynamic AWnd control --- p.49
Chapter 9.2 --- Split-TCP --- p.50
Chapter 9.3 --- Dynamic resource allocation --- p.50
Chapter 9.4 --- Sender-based acceleration --- p.51
Chapter Chapter 10 --- CONCLUSION --- p.52
BIBLIOGRAPHY --- p.53

Siu Chun.
Thesis (M.Phil.)--Chinese University of Hong Kong, 2001.
Includes bibliographical references (leaves 66-68).
Abstracts in English and Chinese.
Chapter 1 --- Introduction --- p.1
Chapter 1.1 --- Thesis Overview --- p.3
Chapter 2 --- Background and Related work --- p.4
Chapter 2.1 --- Emergence of IP-over-ATM --- p.4
Chapter 2.2 --- ATM architecture --- p.5
Chapter 2.3 --- Scheduling issues in output-queued switch --- p.6
Chapter 2.4 --- Scheduling issues in input-queued switch --- p.18
Chapter 3 --- The Deadline-ordered Burst-based Parallel Scheduling Strategy --- p.23
Chapter 3.1 --- Introduction --- p.23
Chapter 3.2 --- Switch and queueing model --- p.24
Chapter 3.2.1 --- Switch model --- p.24
Chapter 3.2.2 --- Queueing model --- p.25
Chapter 3.3 --- The DBPS Strategy --- p.26
Chapter 3.3.1 --- Motivation --- p.26
Chapter 3.3.2 --- Strategy --- p.31
Chapter 3.4 --- The Deadline-ordered Burst-based Parallel Iterative Matching --- p.33
Chapter 3.4.1 --- Algorithm --- p.34
Chapter 3.4.2 --- An example of DBPIM --- p.35
Chapter 3.5 --- Simulation results --- p.33
Chapter 3.6 --- Discussions --- p.46
Chapter 3.7 --- Future work --- p.47
Chapter 4 --- The Quasi-static DBPIM Algorithm --- p.50
Chapter 4.1 --- Introduction --- p.50
Chapter 4.2 --- Quasi-static path scheduling principle --- p.51
Chapter 4.3 --- Quasi-static DBPIM algorithm --- p.56
Chapter 4.4 --- An example of Quasi-static DBPIM --- p.59
Chapter 5 --- Conclusion --- p.63
Bibliography --- p.65

The emergence of portable terminals in work and living environments is accelerating the progression of wireless networks. A wireless ad hoc network is a new network concept where users establish peer-to-peer communication among themselves independently, in their small area. Since the wireless medium is a shared resource, it becomes an important design issue to efficiently allocate bandwidth among users. MAC (Medium Access Control) layer arbitrates the channel access to the wireless medium and is also responsible for bandwidth allocation to different users, therefore a large amount of research has been conducted on various MAC protocols for ad hoc wireless networks. This dissertation begins with a survey of existing wireless MAC protocols. The survey includes protocols designed for different network generations and topologies, classifying them based on architecture and mode of operation. Next, we concentrate on the MAC protocols proposed for distributed wireless networks. We propose a new MAC protocol based on a token-passing strategy; which not only incorporates the advantages of the guaranteed access scheme into the distributed type of wireless networks, but also the data rate and delay level QoS guarantees. Data rate QoS provides fairness into sharing of the channel, while delay level QoS introduces a flexible prioritized access to channels by adjusting transmission permission to the current network traffic activities. A simulation model for the protocol is developed and delay and throughput performance results are presented. To examine the efficiency and performance of the proposed MAC scheme in an ad hoc wireless environment, it is incorporated into the Bluetooth structured network. The model is then simulated in the Bluetooth environment and performance results are presented. Furthermore, an analytical model is proposed and an approximate delay analysis conducted for the proposed MAC scheme. Analytical results are derived and compared with results obtained from computer simulations. The dissertation concludes with suggestions for improvements and future work.
Thesis (M.Sc.-Engineering)-University of Natal, 2003.

This research is motivated by the recent developments in the Internet Engineering Task Force (IETF) to support seamless integration of moving networks deployed in vehicles to the global Internet. The effort, known as Network Mobility (NEMO), paves the way to support high-speed Internet access in mass transit systems, e.g. trains; buses; ferries; and planes; through the use of on-board mobile routers embedded in the vehicle. One of the critical research challenges of this vision is to achieve high-speed and reliable back-haul connectivity between the mobile router and the rest of the Internet. The problem is particularly challenging due to the fact that a mobile router must rely on wireless links with limited bandwidth and unpredictable quality variations as the vehicle moves around. In this thesis, the multi-homing concept is applied to approach the problem. With multi-homing, mobile router has more than one connection to the Internet. This is achieved by connecting the mobile router to a diverse array of wireless access technologies (e.g., GPRS, CDMA, 802.11, and 802.16) and/or a multiplicity of wireless service providers. While the aggregation helps addressing the bandwidth problem, quality variation problem can be mitigated by employing advanced traffic engineering techniques that dynamically control inbound and outbound traffic over multiple connections. More specifically, the thesis investigates traffic engineering solutions for mobile networks that can effectively address the performance objectives, e.g. maximizing profit for mobile network operator; guaranteeing quality of service for the users; and maintaining fair access to the back-haul bandwidth. Traffic engineering solutions with three different levels of control have been investigated. First, it is shown, using detailed computer simulation of popular applications and networking protocols(e.g., File Transfer Protocol and Transmission Control Protocol), that packet-level traffic engineering which makes decisions of which Internet connection to use for each and every packet, leads to poor system throughput. The main problem with packet-based traffic engineering stems from the fact that in mobile environment where link bandwidths and delay can vary significantly, packets using different connections may experience different delays causing unexpected arrivals at destinations. Second, a maximum utility flow-level traffic engineering has been proposed that aims to maximize a utility function that accounts for bandwidth utilization on the one hand, and fairness on the other. The proposed solution is compared against previously proposed flow-level traffic engineering schemes and shown to have better performance in terms of throughput and fairness. The third traffic engineering proposal addresses the issue of maximizing operator?s profit when different Internet connections have different charging rates, and guaranteeing per user bandwidth through admission control. Finally, a new signaling protocol is designed to allow the mobile router to control its inbound traffic.