Dissertations / Theses on the topic 'Packet transport networks'

Mestrado em Engenharia Eletrónica e Telecomunicações
In this dissertation is performed a study of packet and circuit switch on optical transport networks, considering network nodes architecture specifications, functionalities and limitations. It is presented an optimization model for nodes dimensioning, considering network traffic requirements and associated costs, using integer linear programming (ILP) tools. To start, an analysis of the client signal is done, before it enters the optical domain, presenting the main protocols, procedures and transport containers for data. When in optical domain, various grooming configurations, considering client signals bit rate are analysed. Considering the node's architecture and transport modes, mathematical models capable to minimize the costs associated with the network nodes are developed for each considered architecture. Results for different simulation scenarios, continuously increasing network traffic are presented and discussed. This document is finalized with the main conclusions driven by techno-economic analysis.
Nesta dissertação é efetuado um estudo sobre comutação de pacotes e circuitos em redes de transporte óticas, considerando especificidades da arquitetura dos nós, funcionalidades e limitações. É apresentado um modelo de optimização para dimensionamento dos nós, considerando requisitos de tráfego da rede e custos associados, utilizando ferramentas de programação linear inteira (PLI). Analisando o sinal de cliente, antes de este entrar no domínio ótico, apresentando os principais protocolos, procedimentos e contentores para transporte de dados. No domínio ótico, diversas configurações de agregação são analisadas, considerando o bit rate dos sinais. Após análise da arquitetura dos nós e modos de transporte, desenvolve- -se um modelo matemático capaz de minimizar os custos associados aos nós da rede, para as arquiteturas consideradas. Os resultados dos diferentes cenários de simulação, com crescimento constante do tráfego são discutidos, finalizando o documento com as principais conclusões obtidas através de análise técnico-económica.

Deploying wireless networks in networked control systems (NCSs) has become more and more popular during the last few years. As a typical type of real-time control systems, an NCS is sensitive to long and nondeterministic time delay and packet losses. However, the nature of the wireless channel has the potential to degrade the performance of NCS networks in many aspects, particularly in time delay and packet losses. Transport layer protocols could play an important role in providing both reliable and fast transmission service to fulfill NCS’s real-time transmission requirements. Unfortunately, none of the existing transport protocols, including the Transport Control Protocol (TCP) and the User Datagram Protocol (UDP), was designed for real-time control applications. Moreover, periodic data and sporadic data are two types of real-time data traffic with different priorities in an NCS. Due to the lack of support for prioritized transmission service, the real-time performance for periodic and sporadic data in an NCS network is often degraded significantly, particularly under congested network conditions. To address these problems, a new transport layer protocol called Reliable Real-Time Transport Protocol (RRTTP) is proposed in this thesis. As a UDP-based protocol, RRTTP inherits UDP’s simplicity and fast transmission features. To improve the reliability, a retransmission and an acknowledgement mechanism are designed in RRTTP to compensate for packet losses. They are able to avoid unnecessary retransmission of the out-of-date packets in NCSs, and collisions are unlikely to happen, and small transmission delay can be achieved. Moreover, a prioritized transmission mechanism is also designed in RRTTP to improve the real-time performance of NCS networks under congested traffic conditions. Furthermore, the proposed RRTTP is implemented in the Network Simulator 2 for comprehensive simulations. The simulation results demonstrate that RRTTP outperforms TCP and UDP in terms of real-time transmissions in an NCS over wireless networks.

Wireless communication has seen a tremendous growth in the last decades. Continuing on this trend, wireless multi-hop networks  are nowadays used or planned for use in a multitude of contexts, spanning from Internet access at home to emergency situations. The Transmission Control Protocol (TCP) provides reliable and ordered delivery of a data and is used by major Internet applications such as web browsers, email clients and file transfer programs. TCP traffic is also the dominating traffic type on the Internet. However, TCP performs less than optimal in wireless multi-hop networks due to packet reordering, low link capacity, packet loss and variable delay. In this thesis, we develop novel proposals for enhancing the network and transport layer to improve TCP performance in wireless multi-hop networks. As initial studies, we experimentally evaluate the performance of different TCP variants, with and without mobile nodes. We further evaluate the impact of multi-path routing on TCP performance and propose packet aggregation combined with aggregation aware multi-path forwarding as a means to better utilize the available bandwidth. The last contribution is a novel extension to multi-path TCP to  enable single-homed hosts to fully utilize the network capacity.

Opponent changed. Prof. C. Lindeman from the University of Leipzig was substituted by Prof. Zhang.

PONTIFÍCIA UNIVERSIDADE CATÓLICA DO RIO DE JANEIRO
COORDENAÇÃO DE APERFEIÇOAMENTO DO PESSOAL DE ENSINO SUPERIOR
PROGRAMA DE EXCELENCIA ACADEMICA
Há, nos dias atuais, uma crescente demanda pelo transporte de video sobre IP, i.e., para distribuição de conteúdo pela Internet, por serviços de IPTV em definição padrão e em alta definição e, mesmo para uso interno nas redes de emissoras tradicionais de televisão, que transportam contribuições de elevada qualidade para seus programas. Em tais aplicações, o conteúdo dos programas é transportado usando MPEG-2 ou MPEG-4, sob a forma de MPEG-2 Transport Streams, encapsulados com protocolos tais como RTP, UDP e IP. As redes IP, que são modelizadas como Redes com Apagamento de Pacotes (PEC) não foram, no entanto, concebidas para o transporte de mídias em tempo real, esbarra portanto em problemas comuns como perdas de pacotes e jitter, gerando perturbações que se refletem na recepção do conteúdo. Os métodos tradicionais para superar estas dificuldades, como por exemplo, os que se baseiam em retransmissões usando protocolos ARQ (Automatic Repeat on Request), não são uma solução eficiente para proteger a transmissão de multimídia em tempo real. A proteção de multimídia transmitida em tempo real via IP recorre, neste caso, aos códigos para canal. Há códigos para canal recomendados em RFC s e Padrões, usados amplamente pelos fabricantes de equipamento. Os modernos Códigos Fontanais, possuem características atraentes para o transporte de conteúdos multimídia em tempo real. Neste trabalho, simulações são realizadas, onde o conteúdo encapsulado em Transport Stream, é protegido com Códigos Fontanais antes do encapsulamento para o envio através da rede. A título de comparação, o experimento é realizado também usando outros códigos para canal recomendados. Para realizar a comparação são usadas medições padronizadas do Transport Stream, medições objetivas como artefatos de blocagem e finalmente uma análise subjetiva do conteúdo recebido é usada. O trabalho conclui com a proposta de um Codificador de canal adaptável para Transport Stream.
There is a growing demand for the transport of video over IP today, i.e., for content distribution over the Internet, IPTV services in Standard and High Definition, or even inside traditional broadcasters networks, transporting broadcast quality contributions to the main program. In such applications, the source encoded MPEG-2 or -4 content is transported in the form of MPEG-2 Transport Streams, encapsulated over network protocols. However, IP networks, which can be modeled as Packet Erasure Networks (PEC), were not originally designed for the transport of real time media. There are problems, such as packet drops and jitter, which generate severe impairments in the content that is decoded at the reception. Traditional methods for overcoming these problems, as for example retransmissions performed by Automatic Repeat Request (ARQ) protocols, are not suitable for real-time multimedia protection. Channel coding is the solution of choice for protecting real-time multimedia over IP. There are channel coding schemes specified in open recommendations and Standards, widely adopted by equipment vendors today. Fountain Codes present very attractive characteristics for the transport of real-time multimedia. In the present work, simulations with a Fountain code, protecting Transport Stream contents prior to network encapsulation, are presented. The experiment if repeated with other channel coding techniques commonly employed today. In order to analyze the decoded contents and obtain comparative results, standardized Transport Stream measurements, objective Blocking Artifacts measurements and subjective analysis of the decoded samples are employed. This work is concluded with the proposal of a Transport Stream Adaptive channel encoder, that is explained in Appendix-B.

Due to the high availability of cheap hardware, wireless multi-hop networks and in particular Wireless Mesh Networks (WMNs) are becoming popular in more and more contexts. For instance, IEEE 802.11 based WMNs have already started to be deployed as means to provide Internet access to rural areas in the developing world. To lower the cost and increase the coverage in such deployments, the wired network is extended with a wireless backbone of fixed mesh routers. With advances in technology and reduction in price comes also the possibility for more powerful wireless nodes, having multiple radios that allow transmitting on different channels in parallel. To be a successful platform for providing general Internet access, wireless multi-hop networks must provide support for common Internet applications. As most of the applications in the Internet today use the Transmission Control Protocol (TCP), TCP performance is crucial. Unfortunately, the design of TCP’s congestion control that made it successful in today’s Internet makes it perform less than optimal in wireless multi-hop networks. This is due to, among others, TCP’s inability to distinguish wireless losses from congestion losses. The current trend for operating system designers is also to focus TCP development on high-speed fixed networks, rather than on wireless multi-hop networks. To enable wireless multi hop networks as a successful platform there is therefore a need to provide good performance using TCP variants commonly deployed in the Internet. In this thesis, we develop novel proposals for the network layer in wireless multi-hop networks to support TCP traffic more efficiently. As an initial study, we experimentally evaluate different TCP variants, with and without mobile nodes, in a MANET context. Our results show that TCP Vegas, which does not provoke packet loss to determine available bandwidth, reduces the stress on the network while still providing the same or slightly increased performance, compared to TCP Newreno. We further propose and evaluate packet aggregation combined with aggregation aware multi-path forwarding to better utilize the available bandwidth. IP layer packet aggregation, where small packets are combined to larger ones before sent to the link layer, has been shown to improve the performance in wireless multi-hop networks for UDP and small packet transfers. Only few studies have been made on the impact of packet aggregation on TCP traffic, despite the fact that TCP traffic constitutes the majority of the Internet traffic. We propose a novel aggregation algorithm that is specifically addressing TCP relevant issues like packet reordering, fairness and TCP timeouts. In a typical WMN scenario, the aggregation algorithm increases TCP performance by up to 70 % and decreases round trip time (RTT) by up to 40 %. A detailed evaluation of packet aggregation in a multi radio setting has shown that a naive combination of multi path routing and packet aggregation can cause valuable aggregation opportunities to be lost. Therefore, we propose a novel combined packet aggregation and aggregation aware forwarding strategy that can reduce delay, packet loss and increase TCP performance by around 30 %.

Dans cette thèse, nous étudions la solution LOCARN : ''Low Opex & Capex Architecture for Resilient Networks''. LOCARN est une architecture de réseaux paquet alternative conçue dans une optique de simplicité de sa structure et ses mécanismes tout en permettant par sa conception la résilience et l'auto-adaptation des services de transport clients. Compte tenu de la complexification croissante des réseaux de transport opérateurs ces dernières décennies, nous prenons ces réseaux comme cas d'usage privilégié. Dans ce cadre, LOCARN permet une simplification considérable des composants et de leur gestion en comparaison des solutions actuelles des opérateurs – ce qui implique respectivement des réductions de CAPEX et d'OPEX. Dans le travail qui suit, nous présentons LOCARN techniquement et mettons en évidence ses intérêts pour les opérateurs par rapport aux autres technologies de transport. Puis, la question prioritaire étant la capacité de mise à l'échelle de LOCARN pour des réseaux de grandes dimensions, nous étudions cette question en détails ce qui nous permet d'établir que l'architecture est tout à fait capable de passer à l'échelle dans des réseaux de transport réalistes. En outre, pour améliorer les performances nous avons également spécifié et évalué deux améliorations de conception permettant à l'architecture de transporter d'un très grand nombre de services, les résultats obtenus sont très encourageants
In this thesis, we study the LOCARN solution “Low Opex & Capex Architecture for Resilient Networks". LOCARN is an alternative packet network architecture that has been conceived with a special attention to the simplicity of its structure and mechanisms while allowing by design the resiliency and the self-adaptation of clients transportation services. Considering the growing complexification of operators transport networks during the last decades, we consider these latter as the privileged use case. In such a context, LOCARN would allow a drastical simplification of devices and their operation compared to common operator solutions – this involves respectively reductions of CAPEX and OPEX. In this work, we first present LOCARN technically and we bring out its interests for operators beside other transport technologies. Then, since the primary issue of LOCARN is it scalability for large networks, we study this point in details which allow us to establish that the architecture is altogether capable to scale in realistics transport networks. Moreover, to increase the performances we specified two design improvements allowing the architecture to transport a huge amount of clients, the obtained results are very encouraging

A Networked Control System (NCS) is a feedback-driven control system wherein the control loops are closed through a real-time network. Control and feedback signals in an NCS are exchanged among the system’s components in the form of information packets via the network. Nowadays, wireless technologies such as IEEE802.11 are being introduced to modern NCSs as they offer better scalability, larger bandwidth and lower costs. However, this type of network is not designed for NCSs because it introduces a large amount of dropped data, and unpredictable and long transmission latencies due to the characteristics of wireless channels, which are not acceptable for real-time control systems. Real-time control is a class of time-critical application which requires lossless data transmission, small and deterministic delays and jitter. For a real-time control system, network-introduced problems may degrade the system’s performance significantly or even cause system instability. It is therefore important to develop solutions to satisfy real-time requirements in terms of delays, jitter and data losses, and guarantee high levels of performance for time-critical communications in Wireless Networked Control Systems (WNCSs). To improve or even guarantee real-time performance in wireless control systems, this thesis presents several network layout strategies and a new transport layer protocol. Firstly, real-time performances in regard to data transmission delays and reliability of IEEE 802.11b-based UDP/IP NCSs are evaluated through simulations. After analysis of the simulation results, some network layout strategies are presented to achieve relatively small and deterministic network-introduced latencies and reduce data loss rates. These are effective in providing better network performance without performance degradation of other services. After the investigation into the layout strategies, the thesis presents a new transport protocol which is more effcient than UDP and TCP for guaranteeing reliable and time-critical communications in WNCSs. From the networking perspective, introducing appropriate communication schemes, modifying existing network protocols and devising new protocols, have been the most effective and popular ways to improve or even guarantee real-time performance to a certain extent. Most previously proposed schemes and protocols were designed for real-time multimedia communication and they are not suitable for real-time control systems. Therefore, devising a new network protocol that is able to satisfy real-time requirements in WNCSs is the main objective of this research project. The Conditional Retransmission Enabled Transport Protocol (CRETP) is a new network protocol presented in this thesis. Retransmitting unacknowledged data packets is effective in compensating for data losses. However, every data packet in realtime control systems has a deadline and data is assumed invalid or even harmful when its deadline expires. CRETP performs data retransmission only in the case that data is still valid, which guarantees data timeliness and saves memory and network resources. A trade-off between delivery reliability, transmission latency and network resources can be achieved by the conditional retransmission mechanism. Evaluation of protocol performance was conducted through extensive simulations. Comparative studies between CRETP, UDP and TCP were also performed. These results showed that CRETP significantly: 1). improved reliability of communication, 2). guaranteed validity of received data, 3). reduced transmission latency to an acceptable value, and 4). made delays relatively deterministic and predictable. Furthermore, CRETP achieved the best overall performance in comparative studies which makes it the most suitable transport protocol among the three for real-time communications in a WNCS.

While developing networking for games and applications, developers have a list of network specific requirements to be met as well as decide how to meet them. It is not always easy to decide what protocol is best suited for a given network configuration, or what is the best size of a data packet. By performing a comparative analysis, it becomes possible to identify how protocols, packet size, and network configuration impact the one-way travel time and throughput of a given implementation. The result shows how the different implementations compared against each other and the analysis tries to determine why they perform as they do. This gives a good overview of the pros and cons of how TCP, TCP(N), UDP, and RakNet, behave and perform over LANs and WLANs with varying packet size.

Handover performance is critical to support real-time traffic applications in wireless network communications. The longer the handover delay is, the longer an Mobile Node (MN) is prevented from sending and receiving any data packet. In real-time network communication applications, such as VoIP and video-conference, a long handover delay is often unacceptable. In order to achieve better handover performance, Fast Proxy Mobile IPv6 (FPMIPv6) has been standardised as an improvement to the original Proxy Mobile IPv6 (PMIPv6) in the Internet Engineering Task Force (IETF). The FPMIPv6 adopts a link layer triggering mechanism to perform two modes of operation: predictive and reactive modes. Using the link layer triggering, the handover performance of the FPMIPv6 can be improved in the predictive mode. However, an unsuccessful predictive handover operation will lead to activation of a reactive handover. In the reactive mode, MNs still experience long handover delays and a large amount of packet loss, which significantly degrade the handover performance of the FPMIPv6. Addressing this problem, this thesis presents an Enhanced Triggering Mechanism (ETM) in the FPMIPv6 to form an enhanced FPMIPv6 (eFPMIPv6). The ETM reduces the most time consuming processes in the reactive handover: the failed Handover Initiate (HO-Initiate) delay and bidirectional tunnel establishment delay. Consequently, the overall handover performance of the FPMIPv6 is enhanced in the eFPMIPv6. To show the advantages of the proposed eFPMIPv6, a theoretical analysis is carried out to mathematically model the performance of PMIPv6, FPMIPv6 and eFPMIPv6. Extensive case studies are conducted to validate the effectiveness of the presented eFPMIPv6 mechanism. They are carried out under various scenarios with changes in network link delay, traffic load, number of hops and MN moving velocity. The case studies show that the proposed mechanism ETM reduces the reactive handover delay, and the presented eFPMIPv6 outperforms the PMIPv6 and FPMIPv6 in terms of the overall handover performance.

The thesis deals with the implementation of Ethernet-based network communication interface into FPGA chip. VHDL programming language is used for description of the hardware. The interface includes the implementation of link-layer Ethernet protocol and network protocols such as IPv4, ARP, ICMP and UDP. The final design allows bi-directional communication on the transport-layer level of TCP/IP model. The designed interface was implemented into Virtex5 FPGA chip on development board ML506 by Xilinx.

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.

碩士
國立交通大學
資訊學院資訊學程
104
It will be a critical issue for operators to manage and operate 4G / 5G mobile communications backhaul networks with packet transmission networks (PTN), especially in large-scale deployment of 4G / LTE micro base stations based on the needs of the dynamic bandwidth scheduling. Operators are eager to find a solution that not only simplifies network management, but also reduces operating costs. By using SDN centralized control plane to provide an open and standard way to configure and manage data plane of backhaul network equipment, it is a highly potential technology that can be applied to the backhaul networks and PTN. To support SDN, equipment vendors need to support protocols such as OpenFlow in its control plane and network management system. In addition, it needs to update the data plane to make it compatible with OpenFlow. SDN and PTN will be the main technologies in the next decade, if the two can be combined together. It will have higher prospective value to next-generation network equipment. This thesis is a combination of PTN and SDN. We introduce PTN solutions to SDN devices and apply the solutions to next generation of 4G / 5G mobile communication backhaul networks to achieve a unified network management interface for heterogeneous equipment.

Research Doctorate - Doctor of Philosophy (PhD)
Vehicular Ad hoc Network (VANET) is envisaged by the automotive and the ICT industries as the future technology to enhance the safety and reliability of traffic systems as well as to improve the comfort level of road users. A number of applications such as cooperative awareness, traffic management and user entertainment could be supported by the inter-vehicle ad hoc communication networks. To efficiently support these applications in an ad hoc network, reliable packet communication architecture is necessary. Using the VANET architecture, a multi-service ad hoc network could be developed by employing packet broadcast techniques. A multiservice ad hoc network imposes stringent packet transmission requirements for most traffic types, introducing number of challenges such as packet collisions due to hidden nodes, multi-path fading and the packet broadcast storm. In this research, several efficient multi-service packet transmission techniques have been developed and evaluated for future road traffic applications in VANETs.

Road transport mobility continues to be a challenge to the City of Johannesburg (CoJ)’s economy in general. Traffic signals, their remote monitoring and control systems are the current implemented Intelligent Transport Systems (ITS), but daily systematic faults and maintenance uncertainties on such systems decrease the effectiveness of traffic engineers’ intersections optimization techniques. Inefficient electrical power supply to such ITS is a challenge, with conditional power cuts and fluctuations, uncertainties on traffic control system faults. Another factor leading to the problem is the communication channel which is using traditional modems which are not reliable. Reporting through both customer complaints and such unreliable remote monitoring systems makes maintenance to be ineffective. In this dissertation, the factors leading to the faults and uncertainties are considered. The proposed solution considers the important concerns of ITS, such as electrical power source performance optimization technique, road traffic control systems compatibility and communications systems
Electrical and Mining Engineering
M. Tech. (Electrical Engineering)

In this thesis, we examine the effects of packet loss on the quality of the MPEG-2 video transported over an IP network. We performed experiments in a video-ondemand testbed network using three constant bit rate MPEG-2 video elementary streams of differing activity levels. We assessed the video quality using the objective peak signal to noise ratio (PSNR) metric and the subjective mean opinion score (MOS) metric. Our results confirmed that the objective PSNR video quality metric is poorly correlated with subjective quality assessments. We found that the activity level of the video influences the subjective MOS video quality assessments but not the objective PSNR quality measurements. The difference in the perceived video quality between the activity streams was found to be attributed to characteristics in the human visual system. In particular, a high temporal frequency decreases the sensitivity of the visual system. We found that slice loss is linearly correlated with packet loss while picture header loss is poorly correlated with packet loss. We also found that slice loss is the dominant factor contributing to the degradation in video quality rather than picture header loss. We found that low packet loss rates translate into much higher frame error rates due to propagation of errors. Finally, we investigated the effectiveness of forward error correction (FEC) using Reed-Solomon coding on the video quality. We found that FEC can increase a stream's packet loss rate tolerance significantly with a small increase in overhead.