Dissertations / Theses on the topic 'Multicasting (Computer networks) TCP/IP (Computer network protocol) Telecommunication'

Thesis (MScEng) -- Stellenbosch University, 2004.
ENGLISH ABSTRACT: Interest in TCP/IP as a communication protocol for use in space communication has increased substantially with the growth of the world wide web (WWW). TCP/IP is a relevant communication protocol for space based communication systems that need to access the broader terrestrial communication network. Low Earth Orbit(LEO) satellites have very short delay times between themselves and the ground, and correspondingly very short connection times. Staying in contact with a LEO satellite continuously through a space-based network requires large constellations of satellites and complex routing strategies. Connectivity with the world wide web using a widely accepted protocol such as TCP/IP is desirable because it would make communication with the satellite over a terrestrial station possible, were it to route communication onto the WWW. This thesis looks at the expected TCP/IP performance over satellite network links, identifies problem areas for current TCP/IP technologies, and makes suggestions for optimizing TCP/IP over such links. The thesis also introduces a new performance benchmark, the equivalence level, allowing for the simplified description of TCP throughput behaviour over a broad range of network parameters. The performance of the Linux kernel release 2.4.18 TCP/IP stack is also evaluated.
AFRIKAANSE OPSOMMING: Die belangstelling in TCP/IP as ’n kommunikasie protokol vir gebruik in die ruimte het kenmerklik toegeneem met die groei van die wereld wye web (WWW). TCP/IP is ’n relevante protokol vir kommunikasie stelsels in die ruimte, veral met die doel om toegang tot land gebaseerde kommunikasie netwerke te kry. Lae wentelbaan sateliete het baie kort vertragingstye tussen die aarde en die sateliet, en gevolglik baie kort verbindingstye. Groot sateliet konstelasies en komplekse verbintenis strategie word benodig om ’n lae wentelbaan sateliet deurentyd in kontak te hou met ’n ruimtegebaseerde netwerk. Verbinding met die wereld wye web deur die gebruik van ’n wyd aanvaarde protokol, soos TCP/IP, is wenslik, want dit sal kommunikasie met die sateliet oor ’n aardgebaseerde stasie moontlik maak, sou dit kommunikasie oor die wereld wye web stuur. Hierdie tesis kyk na die verwagte werking van TCP/IP oor sateliet netwerk konneksies, identifiseer probleme met huidiglike TCP/IP tegnologie, en maak voorstellings vir die optimale funtionering van TCP/IP oor sulke konneksies. Hierdie tesis stel ook ’n nuwe werkverrigtings maatstaf, die gelykheidsvlak, wat die vereenvoudige beskrywing van TCP/IP data tempo gedrag oor ’n groot variasie van netwerk parameters toelaat. Die werking van die Linux Kernel 2.4.18 TCP/IP stapel word ook geevalueer.

Thesis (M. Phil.)--Hong Kong University of Science and Technology, 2003.
Includes bibliographical references (leaves 68-71). Also available in electronic version. Access restricted to campus users.

Despite the persistent change and growth that characterizes the Internet, the Transmission Control Protocol (TCP) still dominates at the transport layer, carrying more than 90\% of the global traffic. Despite its astonishing success, it has been observed that TCP can cause poor end-to-end performance, especially for large transfers and in network paths with high bandwidth-delay product. In this thesis, we focus on mechanisms that can address key problems in TCP performance, without any modification in the protocol itself. This evolutionary approach is important in practice, as the deployment of clean-slate transport protocols in the Internet has been proved to be extremely difficult. Specifically, we identify a number of TCP-related problems that can cause poor end-to-end performance. These problems include poorly dimensioned socket buffer sizes at the end-hosts, suboptimal buffer sizing at routers and switches, and congestion unresponsive TCP traffic aggregates. We propose solutions that can address these issues, without any modification to TCP.

In network paths with significant available bandwidth, increasing the TCP window till observing loss can result in much lower throughput than the path's available bandwidth. We show that changes in TCP are {em not required} to utilize all the available bandwidth, and propose the application-layer SOcket Buffer Auto-Sizing (SOBAS) mechanism to achieve this goal. SOBAS relies on run-time estimation of the round trip time (RTT) and receive rate, and limits its socket buffer size when the receive rate approaches the path's available bandwidth. In a congested network, SOBAS does not limit its socket buffer size. Our experiment results show that SOBAS improves TCP throughput in uncongested network without hurting TCP performance in congested networks.

Improper router buffer sizing can also result in poor TCP throughput. Previous research in router buffer sizing focused on network performance metrics such as link utilization or loss rate. Instead, we focus on the impact of buffer sizing on end-to-end TCP performance. We find that the router buffer size that optimizes TCP throughput is largely determined by the link's output to input capacity ratio. If that ratio is larger than one, the loss rate drops exponentially with the buffer size and the optimal buffer size is close to zero. Otherwise, if the output to input capacity ratio is lower than one, the loss rate follows a power-law reduction with the buffer size and significant buffering is needed. The amount of buffering required in this case depends on whether most flows end in the slow-start phase or in the congestion avoidance phase.

TCP throughput also depends on whether the cross-traffic reduces its send rate upon congestion. We define this cross-traffic property as {em congestion responsiveness}. Since the majority of Internet traffic uses TCP, which reduces its send rate upon congestion, an aggregate of many TCP flows is believed to be congestion responsive. Here, we show that the congestion responsiveness of aggregate traffic also depends on the flow arrival process. If the flow arrival process follows an open-loop model, then even if the traffic consists exclusively of TCP transfers, the aggregate traffic can still be unresponsive to congestion. TCP flows that arrive in the network in a closed-loop manner are always congestion responsive, on the other hand. We also propose a scheme to estimate the fraction of traffic that follows the closed-loop model in a given link, and give practical guidelines to increase that fraction with simple application-layer modifications.

In recent times, the imminent lack of public IPv4 addresses has attracted the attention of both the research community and industry. The cellular industry has decided to combat this problem by using IPv6 for all new terminals. However, the success of 3G network deployment will depend on the services offered to end users. Currently, almost all services reside in the IPv4 address space, making them inaccessible to users in IPv6 networks. Thus, an intermediate translation mechanism is required. Previous studies on network address translation methods have shown that Realm Base Kluge Address Heuristic-IP, REBEKAH-IP supports all types of services that can be offered to IPv6 hosts from the public IPv4 based Internet, and provides excellent scalability. However, the method suffers from an ambiguity problem which may lead to call blocking. This thesis presents an improvement to REBEKAH-IP scheme in which the side effect is removed, creating a robust and fully scalable system. The improvement can be divided into two major tasks including a full investigation on the scalability of addressing and improvements to the REBEKAH-IP scheme that allow it to support important features such as ICMP and IP mobility. To address the first task a method called REBEKAH-IP with Port Extension (RPX) is introduced. RPX is extended from the original REBEKAH-IP scheme to incorporate centralised management of both IP address and port numbers. This method overcomes the ambiguity problem, and improves scalability. We propose a priority queue algorithm to further increase scalability. Finally, we present extensive simulation results on the practical scalability of RPX with different traffic compositions, to provide a guideline of the expected scalability in large-scale networks. The second task concerns enabling IP based communication. Firstly, we propose an ICMP translation mechanism which allows the RPX server to support important end-toend control functions. Secondly, we extend the RPX scheme with a mobility support scheme based on Mobile IP. In addition, we have augmented Mobile IP with a new tunneling mechanism called IP-in-FQDN tunneling. The mechanism allows for unique mapping despite the sharing of IP addresses while maintaining the scalability of RPX. We examine the viability of our design through our experimental implementation.

Thesis (MSc)--Stellenbosch University, 2003.
ENGLISH ABSTRACT: The Internet has experienced tremendous growth in the last three decades and has emerged as a platform to carryall forms of communications including voice, video and data. Along with this growth came the urgency for quality of service (QoS) controls in IP networks as different types of traffics have different service requirements. Although the IP protocol is able to scale to very large networks, it does not provide sufficient functionality for traffic engineering in order to enable QoS control. Multi-protocol label switching (MPLS) is a new routing technology that enhances IP with some QoS concepts from ATM and uses relatively simple packet forwarding mechanisms. MPLS has the ability to perform traffic engineering and QoS control by routing traffic flowson virtual connections called label switched paths (LSPs) which are assigned capacity. A large portion of the traffic carried on the Internet consists of data traffic in the form of TCP traffic. This thesis investigates several TCP models to find the ones most suitable to represent TCP traffic in MPLS networks. The models consist of three types. The first type models a single TCP source and the second type models a fixed number of TCP sources. The third type models an infinite number of TCP sources. The models were evaluated by comparing their throughput predictions and results obtained from simulation experiments that were done with the widely-used simulator ns. We also present a simple derivation of the 1/,;e law for the TCP congestion window size where e is the packet loss probability.
AFRIKAANSE OPSOMMING:In die afgelope drie dekades het die Internet beduidende groei ervaar, soveel so dat dit ontluik het as 'n medium om alle tipes van moderne kommunikasies te hanteer insluitend telefoon, video en data. Hierdie groei het gepaard gegaan met die behoefte na diensvlak (QoS) meganismes in IP netwerke aangesien verskillende tipe kommunikasies verskillende diens vereistes het. Alhoewel die IP protokol skalleerbaar is tot baie groot netwerke, voorsien dit nie voldoende funksionaliteit om QoS beheer toe te pas nie. "Multi-protocol label switching" (MPLS) is 'n nuwe roeterings tegnologie wat IP aanvul met QoS konsepte van ATM en dit maak gebruik van relatief eenvoudige pakkie versendings-meganismes. MPLS het die vermoe om netwerk-verkeer reeling en QoS beheer toe te pas deur verkeers-strome te roeteer op virtuele roetes genaamd "label switched paths" (LSPs) aan wie kapasiteit toegeken is. 'n Beduidende gedeelte van Internet-verkeer bestaan uit TCP-verkeer. Hierdie tesis ondersoek verskillende modelle van TCP om die te vind wat die mees geskik is om TCP verkeer in MPLS netwerke te verteenwoordig. Drie tipes modelle is ondersoek. Die eerste tipe moduleer 'n enkele TCP verkeersbron en die tweede tipe moduleer 'n vasgestelde aantal TCP verkeersbronne. Die derde tipe moduleer 'n oneindige aantal verkeersbronne. Die modelle is geevalueer deur hul voorspellings van die tempo van data transmissie te vergelyk met resultate van simulasies. Die simulasies is gedoen met die veelgebruikte simulator ns. Hierdie tesis bevat ook 'n eenvoudige afleiding vir die 1/,;e wet vir die TCP oorlading venster grootte met e die verlies waarskeinlikheid van 'n netwerk pakkie.

The Internet Engineering Task Force (IETF) has recently released network mobility standards that allow deployment of TCP/IP networks onboard a vehicle and maintain permanent network connectivity to the Internet via a vehicular mobile router. This recent development opens up new opportunities for providing efficient mobile computing for users on the move, especially for commuters traveling on public transports. Moreover, central and coordinated management of mobility in a single router, rather than by each user device individually, has numerous advantages. In this architecture, however, it becomes challenging to guarantee network performance due to the mobility of the network and inherently vulnerable nature of wireless links. In this thesis, a detailed performance study of onboard networks is conducted. It has been shown that disruptions in the mobile router connectivity can significantly degrade network throughput. Moreover, factors such as the limited wireless bandwidth of the access link, variations in the bandwidth due to technology switching, and the communication diversity of onboard users all contribute to the problem of unfair sharing of wireless bandwidth. By leveraging the fact that all onboard communications go through the mobile router, performance enhancing solutions are proposed that can be deployed in the mobile router to transparently address the throughput and fairness problems. In this architecture, when the route is known in advance and repetitive (e.g. for public transport or a regularly commuting private vehicle), a certain degree of prediction of impending link disruptions is possible. An anticipatory state freezing mechanism is proposed that relies on the prediction of link disruptions to freeze and unfreeze the state machine of TCP, the widely used transport protocol in the Internet. Simulation study shows that TCP throughput has a non-linear relationship with the prediction accuracy. As prediction accuracy increases, throughput problem diminishes quickly. An adaptive mobile router based fairness control mechanism is proposed to address the unfair sharing of wireless bandwidth in highly dynamic scenarios. The fairness is controlled by dynamically estimating the round-trip-times of all onboard TCP connections and transparently adjusting the protocol control parameters at the router. The thesis also discusses implementation issues for the proposed solutions.

In this thesis we study the key requirements and solutions for the feasibility and application of Ethernet-TCP/IP technology to the networks we termed Highly-Variable Real-Time Networks (HVRN). This particular class of networks poses exceptionally demanding requirements because their physical and logical topologies are both temporally and spatially variable. We devised and introduced specific mechanisms for applying Ethernet-TCP/IP to HVRNs with particular emphasis on effective and reliable modular connectivity. Using a railroad train as a reference, this work analyzes the unique requirements of HVRNs and focuses on the backbone architecture for such a system under Ethernet and TCP/IP.
Doctorat en sciences appliquées
info:eu-repo/semantics/nonPublished

Thesis (M.Sc.)--York University, 2005. Graduate Programme in Computer Science and Engineering.
Typescript. Includes bibliographical references (leaves 139-145). Also available on the Internet. MODE OF ACCESS via web browser by entering the following URL: http://gateway.proquest.com/openurl?url_ver=Z39.88-2004&res_dat=xri:pqdiss &rft_val_fmt=info:ofi/fmt:kev:mtx:dissertation&rft_dat=xri:pqdiss:MR11871

Ma Ke.
Thesis (M.Phil.)--Chinese University of Hong Kong, 2003.
Includes bibliographical references (leaves 64-68).
Abstracts in English and Chinese.
Chapter 1 --- Introduction --- p.1
Chapter 1.1 --- Background --- p.1
Chapter 1.2 --- Motivation --- p.2
Chapter 1.3 --- Contribution --- p.3
Chapter 1.4 --- Organization --- p.4
Chapter 2 --- Literature Review --- p.5
Chapter 2.1 --- Overview --- p.5
Chapter 2.2 --- Multi-Path Routing --- p.6
Chapter 2.2.1 --- OSPF-ECMP --- p.7
Chapter 2.2.2 --- LFI --- p.7
Chapter 2.2.3 --- QSMP and QDMP --- p.9
Chapter 2.2.4 --- WDP --- p.10
Chapter 2.2.5 --- DMPR --- p.11
Chapter 2.2.6 --- Cidon's Analysis --- p.13
Chapter 3 --- LSLF and SLSLF Conditions --- p.15
Chapter 3.1 --- Problem Formulation --- p.15
Chapter 3.2 --- LFI Conditions --- p.16
Chapter 3.3 --- LSLF Conditions --- p.17
Chapter 3.4 --- SLSLF Conditions --- p.20
Chapter 4 --- Performance of LSLF and SLSLF --- p.24
Chapter 4.1 --- Overview --- p.24
Chapter 4.2 --- Numerical Results --- p.26
Chapter 5 --- Analysis of Multi-path Routing --- p.42
Chapter 5.1 --- Assumptions --- p.43
Chapter 5.2 --- M/M/C/C Queueing System --- p.44
Chapter 5.3 --- Performance Analysis --- p.48
Chapter 5.3.1 --- "Case 1 Only QoS flows between (s, d) exist" --- p.48
Chapter 5.3.2 --- Case 2 QoS flows between other SD pairs also exist --- p.50
Chapter 5.3.3 --- Case 3 A QoS flow can try m times before it is dropped --- p.53
Chapter 5.4 --- Numerical Results --- p.56
Chapter 6 --- Conclusion --- p.62

Yeung, Fei-Fei.
Thesis (M.Phil.)--Chinese University of Hong Kong, 2003.
Includes bibliographical references (leaves 144-146).
Abstracts in English and Chinese.
Introduction --- p.1
Chapter Part I: --- A New Socket API for Enhancing Server Efficiency --- p.5
Chapter Chapter 1 --- Introduction --- p.6
Chapter 1.1 --- Brief Background --- p.6
Chapter 1.2 --- Deficiencies of Nagle's Algorithm and Goals and Objectives of this Research --- p.7
Chapter 1.2.1 --- Effectiveness of Nagle's Algorithm --- p.7
Chapter 1.2.2 --- Preventing Small Packets via Application Layer --- p.9
Chapter 1.2.3 --- Minimum Delay in TCP Buffer --- p.10
Chapter 1.2.4 --- Maximum Delay in TCP Buffer --- p.11
Chapter 1.2.5 --- New Socket API --- p.12
Chapter 1.3 --- Scope of Research and Summary of Contributions --- p.12
Chapter 1.4 --- Organization of Part 1 --- p.13
Chapter Chapter 2 --- Background --- p.14
Chapter 2.1 --- Review of Nagle's Algorithm --- p.14
Chapter 2.2 --- Additional Problems Inherent in Nagle's Algorithm --- p.17
Chapter 2.3 --- Previous Proposed Modifications on Nagle's Algorithm --- p.22
Chapter 2.3.1 --- The Minshall Modification --- p.22
Chapter 2.3.1.1 --- The Minshall Modification --- p.22
Chapter 2.3.1.2 --- The Minshall et al. Modification --- p.23
Chapter 2.3.2 --- The Borman Modification --- p.23
Chapter 2.3.3 --- The Jeffrey et al. Modification --- p.25
Chapter 2.3.3.1 --- The EOM and MORE Variants --- p.25
Chapter 2.3.3.2 --- The DLDET Variant --- p.26
Chapter 2.3.4 --- Comparison Between Our Proposal and Related Works --- p.26
Chapter Chapter 3 --- Min-Delay-Max-Delay TCP Buffering --- p.28
Chapter 3.1 --- Minimum Delay --- p.29
Chapter 3.1.1 --- Why Enabling Nagle's Algorithm Alone is Not a Solution? --- p.29
Chapter 3.1.2 --- Advantages of Min-Delay TCP-layer Buffering versus Application-layer Buffering --- p.30
Chapter 3.2 --- Maximum Delay --- p.32
Chapter 3.2.1 --- Why Enabling Nagle's Algorithm Alone is Not a Solution? --- p.32
Chapter 3.2.2 --- Advantages of Max-delay TCP Buffering versus Nagle's Algorithm --- p.33
Chapter 3.3 --- Interaction with Nagle's Algorithm --- p.34
Chapter 3.4 --- When to Apply Our Proposed Scheme? --- p.36
Chapter 3.5 --- New Socket Option Description --- p.38
Chapter 3.6 --- Implementation --- p.40
Chapter 3.6.1 --- Small Packet Transmission Decision Logic --- p.42
Chapter 3.6.2 --- Modified API --- p.44
Chapter Chapter 4 --- Experiments --- p.46
Chapter 4.1 --- The Effect of Kernel Buffering Mechanism on the Service Time --- p.47
Chapter 4.1.1 --- Aims and Methodology --- p.47
Chapter 4.1.2 --- Comparison of Transmission Time Required --- p.49
Chapter 4.2 --- Performance of Min-Delay-Max-Delay Scheme --- p.56
Chapter 4.2.1 --- Methodology --- p.56
Chapter 4.2.1.1 --- Network Setup --- p.56
Chapter 4.2.1.2 --- Traffic Model --- p.58
Chapter 4.2.1.3 --- Delay Measurement --- p.60
Chapter 4.2.2 --- Efficiency of Busy Server --- p.62
Chapter 4.2.2.1 --- Performance of Nagle's algorithm --- p.62
Chapter 4.2.2.2 --- Performance of Min-Delay TCP Buffering Scheme --- p.67
Chapter 4.2.3 --- Limiting Delay by Setting TCP´ؤMAXDELAY --- p.70
Chapter 4.3 --- Performance Sensitivity Discussion --- p.77
Chapter 4.3.1 --- Sensitivity to Data Size per Invocation of send() --- p.77
Chapter 4.3.2 --- Sensitivity to Minimum Delay --- p.83
Chapter 4.3.3 --- Sensitivity to Round Trip Time --- p.85
Chapter Chapter 5 --- Conclusion --- p.88
Chapter Part II: --- Two Analytical Models for a Refined TCP Algorithm (TCP Veno) for Wired/Wireless Networks --- p.91
Chapter Chapter 1 --- Introduction --- p.92
Chapter 1.1 --- Brief Background --- p.92
Chapter 1.2 --- Motivation and Two Analytical Models --- p.95
Chapter 1.3 --- Organization of Part II --- p.96
Chapter Chapter 2 --- Background --- p.97
Chapter 2.1 --- TCP Veno Algorithm --- p.97
Chapter 2.1.1 --- Packet Loss Type Identification --- p.97
Chapter 2.1.2 --- Refined AIMD Algorithm --- p.99
Chapter 2.1.2.1 --- Random Loss Management --- p.99
Chapter 2.1.2.2 --- Congestion Management --- p.100
Chapter 2.2 --- A Simple Model of TCP Reno --- p.101
Chapter 2.3 --- Stochastic Modeling of TCP Reno over Lossy Channels --- p.103
Chapter Chapter 3 --- Two Analytical Models --- p.104
Chapter 3.1 --- Simple Model --- p.104
Chapter 3.1.1 --- Random-loss Only Case --- p.105
Chapter 3.1.2 --- Congestion-loss Only Case --- p.108
Chapter 3.1.3 --- The General Case (Random + Congestion Loss) --- p.110
Chapter 3.2 --- Markov Model --- p.115
Chapter 3.2.1 --- Congestion Window Evolution --- p.115
Chapter 3.2.2 --- Average Throughput Formulating --- p.119
Chapter 3.2.2.1 --- Random-loss Only Case --- p.120
Chapter 3.2.2.2 --- Congestion-loss Only Case --- p.122
Chapter 3.2.2.3 --- The General Case (Random + Congestion Loss) --- p.123
Chapter Chapter 4 --- Comparison with Experimental Results and Discussions --- p.127
Chapter 4.1 --- Throughput versus Random Loss Probability --- p.127
Chapter 4.2 --- Throughput versus Normalized Buffer Size --- p.132
Chapter 4.3 --- Throughput versus Bandwidth in Asymmetric Networks --- p.135
Chapter 4.3 --- Summary --- p.136
Chapter Chapter 5 --- Sensitivity of TCP Veno Throughput to Various Parameters --- p.137
Chapter 5.1 --- Multiplicative Decrease Factor (α) --- p.137
Chapter 5.2 --- Number of Backlogs (β) and Fractional Increase Factor (γ) --- p.139
Chapter Chapter 6 --- Conclusions --- p.142
Bibliography --- p.144

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

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.

Indiana University-Purdue University Indianapolis (IUPUI)
Mobile IP though allows mobility features to a node it suffers from signaling Latencies which are mainly incurred due to the fact that the MN itself is involved in the handover process. To overcome this problem proxy mobile IPv6(PMIPv6) was defined where the mobility signaling is taken care of by a proxy server while keeping track of the MN's movement. PMIPv6 has considerably reduced the handover latency but the demand for real time applications over the network has increased tremendously due to recent explosion of the cloud era. My thesis focuses on increasing the L3 handoff signaling efficiency by reducing the latency. This is achieved by our idea to do both the AAA authentication as well as the LMA registration in PMIPv6 at the same time. The simulation results show that our proposed approach perform better than the current PMIPv6 L3 handover signaling reducing the latency as well as packet loss.