Tesis sobre el tema "TCP"
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BONAM, VEERA VENKATA SIVARAMAKRISHNA. "Multipath TCP and Measuring end-to-end TCP Throughput : Multipath TCP Descriptions and Ways to Improve TCP Performance". Thesis, Blekinge Tekniska Högskola, Institutionen för datalogi och datorsystemteknik, 2018. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-17075.
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Internet applications make use of the services provided by a transport protocol, such as TCP (a reliable, in-order stream protocol). We use this term Transport Service to mean the end-to- end service provided to application by the transport layer. That service can only be provided correctly if information about the intended usage is supplied from the application. The application may determine this information at the design time, compile time, or run time, and it may include guidance on whether a feature is required, a preference by the application, or something in between. Multipath TCP (MPTCP) adds the capability of using multiple paths to a regular TCP session. Even though it is designed to be totally backward compatible to applications. The data transport differs compared to regular TCP, and there are several additional degrees of freedom that the particular application may want to exploit. Multipath TCP is particularly useful in the context of wireless networks using both Wi-Fi and a mobile network is a typical use case. In addition to the gains in throughput from inverse multiplexing, links may be added or dropped as the user moves in or out of coverage without disrupting the end-to-end TCP connection. The problem of link handover is thus solved by abstraction in the transport layer, without any special mechanisms at the network or link level. Handover functionality can then be implemented at the endpoints without requiring special functionality in the sub-networks according to the Internet's end-to-end principle. Multipath TCP can balance a single TCP connection across multiple interfaces and reach very high throughput.
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SANA, VINEESHA. "Multipath TCP and Measuring endto-end TCP Throughput : Measuring TCP Metrics and ways to improve TCP Throughput performance". Thesis, Blekinge Tekniska Högskola, Institutionen för datalogi och datorsystemteknik, 2018. http://urn.kb.se/resolve?urn=urn:nbn:se:bth-17098.
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Internet applications make use of the services provided by a transport protocol, such as TCP (a reliable, in-order stream protocol). We use the term Transport Service to mean the endtoend service provided to application by the transport layer. That service can only be provided correctly if information about the intended usage is supplied from the application. The application may determine this information at the design time, compile time, or run time, and it may include guidance on whether a feature is required, a preference by the application, or something in between. Multipath TCP (MPTCP) adds the capability of using multiple paths to a regular TCP session. Even though it is designed to be totally backward compatible to applications. The data transport differs compared to regular TCP, and there are several additional degrees of freedom that the particular application may want to exploit. Multipath TCP is particularly useful in the context of wireless networks using both Wi-Fi and a mobile network is a typical use case. In addition to the gains in throughput from inverse multiplexing, links may be added or dropped as the user moves in or out of coverage without disrupting the end-to-end TCP connection. The problem of link handover is thus solved by abstraction in the transport layer, without any special mechanisms at the network or link level. Handover functionality can then be implemented at the endpoints without requiring special functionality in the sub-networks according to the Internet's end-to-end principle. Multipath TCP can balance a single TCP connection across multiple interfaces and reach very high throughput.
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Smith, Mark Anthony Shawn 1968. "Formal verification of TCP and T/TCP". Thesis, Massachusetts Institute of Technology, 1997. http://hdl.handle.net/1721.1/42779.
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Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1997.Includes bibliographical references (p. 421-424).
by Mark Anthony Shawn Smith.
Ph.D.
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Jeacle, Karl. "TCP-XM". Thesis, University of Cambridge, 2006. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.613731.
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Núñez, Mori Oscar. "TCP HolyWood". reponame:Biblioteca Digital de Teses e Dissertações da UFRGS, 2005. http://hdl.handle.net/10183/5710.
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Apresentamos um novo Protocolo de Controle de Transporte fim-a-fim, implementado somente do lado do transmissor, chamado TCP HolyWood ou, abreviadamente, TCP-HW. Em um ambiente de rede cabeada simulada, TCP HolyWood supera em vazão media três dos mais importantes protocolos TCPs já elaborados. Estamos falando de TCP Reno, TCP Westwood, e TCP Vegas; e em variação de retardo media ao TCP Reno bem como ao TCP Vegas. Alem disso, de acordo com o índice de Jain, nossa proposta e tão imparcial quanto o padrão, TCP Reno.We introduce a new end-to-end, sender side Transport Control Protocol called TCP HolyWood or in short TCP-HW. In a simulated wired environment, TCP HolyWood outperforms in average throughput, three of the more important TCP protocols ever made, we are talking about TCP Reno, TCP Westwood, and TCP Vegas; and in average jitter to TCP Reno and TCP Vegas too. In addition, according to Jain’s index, our proposal is as fair as TCP Reno, the Standard.
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Shepard, Timothy Jason. "TCP packet trace analysis". Thesis, Massachusetts Institute of Technology, 1990. http://hdl.handle.net/1721.1/13577.
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Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1990.Includes bibliographical references (leaves 66-67).
by Timothy Jason Shepard.
M.S.
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Squarcialupi, Riccardo. "Il Protocollo MultiPath TCP". Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2021. http://amslaurea.unibo.it/24311/.
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Ad oggi, il Transmission Control Protocol (TCP) è il protocollo più popolare per trasmettere e consegnare informazioni su Internet in modo affidabile. Tuttavia, esso fa uso di una connessione a percorso singolo, non sfruttando il multihoming e i percorsi multipli che sono sempre più disponibili per i dispositivi endpoint, cioè dispositivi mobili e server in configurazioni ad alta resilienza.
Il MultiPath TCP (MPTCP) è stato sviluppato per affrontare queste limitazioni. Esso, infatti, mira a fare uso della diversità dei percorsi, al fine di offrire una migliore connettività di rete complessiva, aumentando così la resilienza ai guasti, eseguendo il bilanciamento del carico tra i percorsi disponibili (quando più di uno è disponibile). Consente, inoltre il supporto multihoming, senza la necessità di modificare i dispositivi già esistenti attualmente sparsi sulla rete.
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Workman, Russell. "TCP over military networks". Thesis, Swansea University, 2004. https://cronfa.swan.ac.uk/Record/cronfa42626.
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The major issues with existing military communications are the requirement to improve connectivity between systems. At present there are many bespoke systems used by co-operating nations that are unable to share information because of inter-connectivity problems. The solution requires the use of a common standard that the different nations are willing to implement and use. One network technology that is able to address these problems is the Internet protocol suite. Tactical communications rely heavily on radio links to provide connectivity across the network. Radio links are inherently less reliable than fixed. The aim of this thesis is to study degrading effects at the radio link layer and see how this affects the performance of TCP/IP. The study of the radio link effects concentrated on the distribution of errors (error patterns). The results from radios were compared against standard models and revealed a discrepancy between radios and the models. The military radio showed characteristics that were very different from any of the standard models hence a new burst error model was developed that could mimic these error patterns. The next stage of the study was to observe how these burst errors affect the data link layer protocols. A hypothesis was proposed that burst errors would improve the performance of unprotected packet but adversely affect FEC. This was proven to be true, but the effect is only really noticeable at the point where the throughput collapses. The results also showed that the worst case to study for unprotected packets was random errors not burst errors. The IP layer has no recovery mechanism, it is Transmission Control Protocol (TCP) that provides the reliability. TCP contains several mechanisms to handle lost packet detection, retransmission and congestion management. The performance of each of the algorithms was tested in the presence of errors to determine which combination produced the highest throughput. A number of network restrictions where also detailed, the round trip time delay should be less than 9 seconds and the residual bit error rate should be better than 1 in 10.;Various ways of separating congestioncontrol and corruption were considered to try and improve the performance of TCP in the presence of errors. These mechanisms were TCP Vegas, a modified version of Vegas and Packet Pair measurement. Both the Modified Vegas and Packet Pair Control produced throughputs that were better than standard TCP in the presence of errors. The Modified Vegas produced very good results under congestion, again better than standard TCP. The other approach that was considered for improving the performance of TCP was to change the links to reduce the errors that effect the performance rather than changing TCP to handle errors. The use of ARQ greatly improved the throughput. The use of ARQ increased the variance in the round trip time which has a major impact on the time delay sensitive algorithms such as Vegas and Modified Vegas. It is recommended that a combination of Modified Vegas and ARQ is not used. From the results it is clear that there could be interaction effects between different protocol layers. Burst bit error may or may not produce burst packet losses. Pulse errors can have a dramatic impact on the throughput of TCP. The time delay variance introduced by ARQ can greatly effect time sensitive congestion control mechanisms such as Vegas. Hence with any system it is necessary to consider all layers not just the performance of a single layer under some arbitrary condition.
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Yu, Inkwan. "Improving TCP slow start". [Gainesville, Fla.] : University of Florida, 2009. http://purl.fcla.edu/fcla/etd/UFE0041219.
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Hallencreutz, Redmond Michael. "TCP-X: An NS-2 environment for implementation and evaluation of TCP clones". Thesis, KTH, Reglerteknik, 2005. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-107517.
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The Transmission Control Protocol (TCP) is the dominating end-to-end protocol on the Internet today, carrying approximately 90% of the total traffic. TCP appear in numerous clones (e.g. Reno, Vegas, Sack etc.) all with different features and advantages but with maximal throughput as main objective (obeying fairness constraints). The performance of a protocol for varying network conditions and settings can effectively be evaluated using simulations. The free discrete network simulator NS-2 provides a dynamic simulation environment including several common TCP implementations. The software NS-2 is the result of an on-going effort of research and development and is hence not a finished product. The task in this masters’ thesis is to coordinate the (in NS-2) existing TCP implementations and develop, implement and thoroughly document a generic TCP protocol collecting existing protocols and where crucial algorithms easily can be manipulated for future experimental validation. The new implementation will be used for evaluating performance of existing and new TCP-versions.
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Hurtig, Per. "Fast retransmit inhibitions for TCP". Thesis, Karlstad University, Division for Information Technology, 2006. http://urn.kb.se/resolve?urn=urn:nbn:se:kau:diva-109.
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The Transmission Control Protocol (TCP) has been the dominant transport protocol in the Internet for many years. One of the reasons to this is that TCP employs congestion control mechanisms which prevent the Internet from being overloaded. Although TCP's congestion control has evolved during almost twenty years, the area is still an active research area since the environments where TCP are employed keep on changing. One of the congestion control mechanisms that TCP uses is fast retransmit, which allows for fast retransmission of data that has been lost in the network. Although this mechanism provides the most effective way of retransmitting lost data, it can not always be employed by TCP due to restrictions in the TCP specification.
The primary goal of this work was to investigate when fast retransmit inhibitions occur, and how much they affect the performance of a TCP flow. In order to achieve this goal a large series of practical experiments were conducted on a real TCP implementation.
The result showed that fast retransmit inhibitions existed, in the end of TCP flows, and that the increase in total transmission time could be as much as 301% when a loss were introduced at a fast retransmit inhibited position in the flow. Even though this increase was large for all of the experiments, ranging from 16-301%, the average performance loss, due to an arbitrary placed loss, was not that severe. Because fast retransmit was inhibited in fewer positions of a TCP flow than it was employed, the average increase of the transmission time due to these inhibitions was relatively small, ranging from 0,3-20,4%.
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Bansal, Dushyant. "Third-Party TCP Rate Control". Thesis, University of Waterloo, 2005. http://hdl.handle.net/10012/803.
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The Transmission Control Protocol (TCP) is the dominant transport protocol in today?s Internet. The original design of TCP left congestion control open to future designers. Short of implementing changes to the TCP stack on the end-nodes themselves, Internet Service Providers have employed several techniques to be able to operate their network equipment efficiently. These techniques amount to shaping traffic to reduce cost and improve overall customer satisfaction. The method that gives maximum control when performing traffic shaping is using an inline traffic shaper. An inline traffic shaper sits in the middle of any flow, allowing packets to pass through it and, with policy-limited freedom, inspects and modifies all packets as it pleases. However, a number of practical issues such as hardware reliability or ISP policy, may prevent such a solution from being employed. For example, an ISP that does not fully trust the quality of the traffic shaper would not want such a product to be placed in-line with its equipment, as it places a significant threat to its business. What is required in such cases is third-party rate control.
Formally defined, a third-party rate controller is one that can see all traffic and inject new traffic into the network, but cannot remove or modify existing network packets. Given these restrictions, we present and study a technique to control TCP flows, namely triple-ACK duplication. The triple-ACK algorithm allows significant capabilities to a third-party traffic shaper. We provide an analytical justification for why this technique works under ideal conditions and demonstrate via simulation the bandwidth reduction achieved. When judiciously applied, the triple-ACK duplication technique produces minimal badput, while producing significant reductions in bandwidth consumption under ideal conditions. Based on a brief study, we show that our algorithm is able to selectively throttle one flow while allowing another to gain in bandwidth.
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Alheid, Amani Abdullatif. "Multipath TCP over heterogeneous environments". Thesis, University of Bristol, 2017. http://ethos.bl.uk/OrderDetails.do?uin=uk.bl.ethos.720819.
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Urbina, Tovar Leonardo Andrés. "Applying network coding to TCP". Thesis, Massachusetts Institute of Technology, 2012. http://hdl.handle.net/1721.1/77005.
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Thesis (M. Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2012.Cataloged from PDF version of thesis.
Includes bibliographical references (p. 93-94).
In this thesis, I contribute to the design and implemention of a new TCP-like protocol, CTCP, that uses network coding to provide better network use of the network bandwidth in a wireless environment. CTCP provides the same guarantees as TCP whilst providing significant enhancements to previous TCP implementations, such as permitting multipath packet delivery. CTCP's flow and congestion control policies are based on those of TCP Reno and TCP Vegas, which allow for prompt recovery from packet erasures and cope with congested networks. Unlike previous attempts at using network coding with TCP, this implementation uses block coding schemes, which are better suited to delay sensitive applications. As a result, CTCP permits content streaming. Overall, the efficient integration of network coding into CTCP allows for improved robustness against erasures as well as efficient content delivery over multiple paths.
by Leonardo Andrés Urbina Tovar.
M.Eng.
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Kassa, Debassey Fesehaye. "Analytic models of TCP performance". Thesis, Stellenbosch : University of Stellenbosch, 2011. http://hdl.handle.net/10019.1/16608.
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Thesis (MSc)--University of Stellenbosch, 2005.ENGLISH ABSTRACT: The majority of tra c on the Internet uses the Transmission Control Protocol (TCP) as a transport layer protocol for the end-to-end control of information transfer. Measurement, simulation and analytical models are the techniques and tools that can be used to understand and investigate the Internet and its performance. Measurements can only be used to explore existing network scenario or otherwise become costly and in exible with the growth and complexity of the Internet. Simulation models do not scale with the growth of network capacities and the number of users. Computationally e cient analytical models are therefore important tools for investigating, designing, dimensioning and planning IP (Internet Protocol) networks. Existing analytical models of TCP performance are either too simple to capture the internal dynamics of TCP or are too complex to be used to analyze realistic network topologies with several bottleneck links. The literature shows that the xed point algorithm (FPA) is a very useful way of solving analytical models of Internet performance. This thesis presents fast and accurate analytical models of TCP performance with the FPA used to solve them. Apart from what is observed in experimental literature, no comprehensive proof of the convergence and uniqueness of the FPA is given. In this thesis we show how the FPA of analytical models of reliable Internet protocols such as TCP converges to a unique xed point. The thesis speci es the conditions necessary in order to use the FPA for solving analytical models of reliable Internet protocols. We also develop a general implementation algorithm of the FPA of analytical models of TCP performance for realistic and arbitrary network topologies involving heterogenous TCP connections crossing many bottleneck links. The models presented in this thesis give Internet performance metrics, assuming that only basic network parameters such as the network topology, the number of TCP connections, link capacity, distance between network nodes and router bu er sizes are known. To obtain the performance metrics, TCP and network sub{models are used. A closed network of :=G=1 queues is used to develop each TCP sub-model where each queue represents a state of a TCP connection. An M=M=1=K queue is used for each network sub{model which represents the output interface of an IP router with a bu er capacity of K ��������1 packets. The two sub-models are iteratively solved. We also give closed form expressions for important TCP performance values and distributions. We show how the geometric, bounded geometric and truncated geometric distributions can be used to model reliable protocols such as TCP. We give models of the congestion window cwnd size distribution by conditioning on the slow start threshold ssthresh distribution and vice-versa. We also present models of the probabilities of TCP timeout and triple duplicate ACK receptions. Numerical results based on comparisons against ns2 simulations show that our models are more accurate, simpler and computationally more e cient than another well known TCP model. Our models can therefore be used to rapidly analyze network topologies with several bottlenecks and obtain detailed performance metrics.
AFRIKAANSE OPSOMMING: Die meerderheid van die verkeer op die Internet gebruik die Transmission Control Protocol (TCP) as `n vervoer laag protokol vir die einde-tot-einde kontrole van inligting oordrag. Meting, simulasie en analitiese modelle is die tegnieke en gereedskap wat gebruik kan word om die Internet te ondersoek en verstaan. Meting kan slegs gebruik word om bestaande netwerke scenarios te verken. Meting is duur en onbuigsaam met die groei en samegesteldheid van die Internet. Simulasie modelle skaal nie met die groei van netwerk kapasiteit en gebruikers nie. Analitiese modelle wat berekening e ektief is is dus nodige gereedskap vir die ondersoek, ontwerp, afmeting en beplanning van IP (Internet Protocol) netwerke. Bestaande analitiese TCP modelle is of te eenvoudig om die interne dinamiek van die TCP saam te vat of hulle is te ingewikkeld om realistiese netwerk topologie met heelwat bottelnek skakels te analiseer. Literatuur toon dat die xed point algorithm (FPA) baie handig is vir die oplos van analitiese modelle van Internet verrigting. In hierdie tesis word vinnige en akkurate analitiese modelle van TCP verrigting opgelos deur FPA weergegee. Buiten wat deur eksperimentele literatuur aangedui word is daar geen omvattende bewyse van die konvergensie en uniekheid van die FPA nie. In hierdie tesis word aangedui hoe die FPA van analitiese modelle van betroubare Internet protokolle soos die TCP konvergeer na `n unieke vaste punt. Hierdie tesis spesi seer die voorwaardes benodig om die FPA te gebruik vir die oplos van analitiese modelle van realistiese Internet protokolle. `n Algemene uitvoer algoritme van die FPA van analitiese modelle van TCP vir realistiese en arbitr^ere netwerk topogra e insluitende heterogene TCP konneksies oor baie bottelnek skakels is ontwikkel. Die model in hierdie tesis gee Internet verrigting metodes met die aanname dat slegs basiese netwerk parameters soos netwerk topologie, die aantal TCP konneksies, die konneksie kapasiteit, afstand tussen netwerk nodusse en die roete bu er grotes bekend is. Om die verrigting metodes te verkry, word TCP en netwerk sub-modelle gebruik. `n Geslote netwerk van :=G=1 rye is gebruik om elke TCP sub-model, waar elke ry 'n toestand van `n TCP konneksie voorstel, te ontwikkel. `n M=M=1=K ry is gebruik vir elke netwerk sub-model wat die uitset koppelvlak van `n IP roetemaker met `n bu er kapasiteit van K ������� 1 pakkies voorstel. Die twee submodelle word iteratief opgelos. Geslote vorm uitdrukkings vir belangrike TCP verrigting waardes en verspreidings word gegee. Daar word getoon hoe geometriese, begrensde geometriese en geknotte geometriese verspreidings gebruik kan word om betroubare protokolle soos die TCP te modelleer. Modelle van die kongestie venster cwnd grootte verspreiding word gegee deur die kondisionering van die stadige aanvang drempel ssthresh verspreiding en andersom. Modelle van die voorspelling van TCP tyduit en trippel duplikaat ACK resepsie word weergegee. Numeriese resultate gebaseer op vergelykings met ns2 simulasies wys dat ons modelle meer akkuraat, eenvoudiger en berekeningsgewys meer e ektief is as ander wel bekende TCP modelle. Ons modelle kan dus gebruik word vir vinnig analise van netwerk topologie met verskeie bottelnekke en om gedetailleerde verrigting metodes te bekom.
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Torgerson, Leigh, Joseph Hutcherson y James McKelvey. "TCP PERFORMANCE ENHANCEMENT OVER IRIDIUM". International Foundation for Telemetering, 2007. http://hdl.handle.net/10150/604499.
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ITC/USA 2007 Conference Proceedings / The Forty-Third Annual International Telemetering Conference and Technical Exhibition / October 22-25, 2007 / Riviera Hotel & Convention Center, Las Vegas, NevadaIn support of iNET maturation, NASA-JPL has collaborated with NASA-Dryden to develop, test and demonstrate an over-the-horizon vehicle-to-ground networking capability, using Iridium as the vehicle-to-ground communications link for relaying critical vehicle telemetry. To ensure reliability concerns are met, the Space Communications Protocol Standards (SCPS) transport protocol was investigated for its performance characteristics in this environment. In particular, the SCPS-TP software performance was compared to that of the standard Transmission Control Protocol (TCP) over the Internet Protocol (IP). This paper will report on the results of this work.
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Sanhueza, Gutiérrez Andrés Edgardo. "Scalable video coding sobre TCP". Tesis, Universidad de Chile, 2015. http://repositorio.uchile.cl/handle/2250/136454.
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Ingeniero Civil EléctricoEn tiempos modernos la envergadura del contenido multimedia avanza más rápido que el desarrollo de las tecnologías necesarias para su correcta difusión a través de la red. Es por esto que se hacen necesarios nuevos protocolos que sirvan como puente entre ambas entidades para así obtener un máximo de provecho del contenido a pesar de que la tecnología para distribuirlos aún no sea la adecuada. Es así, que dentro de las últimas tecnologías de compresión de video se encuentra Scalable Video Coding (SVC), la cual tiene por objetivo codi car distintas calidades en un único bitstream capaz de mostrar cualquiera de las calidades embebidas en éste según se reciba o no toda la información. En el caso de una conexión del tipo streaming, en donde es necesaria una uidez y delidad en ambos extremos, la tecnología SVC tiene un potencial muy grande respecto de descartar un mínimo de información para privilegiar la uidez de la transmisión. El software utilizado para la creación y manipulación de estos bitstreams SVC es Joint Scalable Video Model (JSVM). En este contexto, se desarrolla el algoritmo de deadline en Matlab, que omite informaci ón del video SVC de acuerdo a qué tan crítico sea el escenario de transmisión. En este escenario se considera la percepción de uidez del usuario como medida clave, por lo cual se prioriza mantener siempre una tasa de 30 fps a costa de una pérdida de calidad mínima. El algoritmo, omite información de acuerdo a qué tan lejos se esté de este deadline de 30 fps, si se está muy lejos, se omite información poco relevante, y si se está muy cerca, información más importante. Los resultados se contrastan con TCP y se evalúan para distintos valores de RTTs, cumpliendo totalmente el objetivo para valores menores a 150 ms que resultan en diferencias de hasta 20 s a favor del algoritmo de deadline al término de la transmisión. Esta mejora en tiempo de arribo no descarta información esencial y sólo degrada ligeramente la calidad del video en pos de mantener la tasa de 30fps. Por el contrario, en escenarios muy adversos de 300 ms en RTT, las omisiones son de gran envergadura y comprometen frames completos, en conjunto con una degradación generalizada del video y la aparición de artefactos en éste. Por tanto la propuesta cumple los objetivos en ambientes no muy adversos. Para toda la simulación se uso un video en movimiento de 352x288 y 150 frames de largo.
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Freire, Sérgio Manuel Silva. "A TCP-layer name service". Master's thesis, Universidade de Aveiro, 2008. http://hdl.handle.net/10773/1941.
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Mestrado em Engenharia Electrónica e TelecomunicaçõesA Internet ´e hoje a maior rede mundial mas para al´em disso, ´e tamb´em e essencialmente um meio de disponibiliza¸c˜ao de acesso a conhecimento e a servi¸cos diversos. Tendo como base o protocolo de encaminhamento IP, ´e poss´ıvel endere¸car e comunicar com pessoas, servi¸cos, m´aquinas e dispositivos variados. Uma forma de comunica¸c˜ao usual assenta no protocolo TCP, que permite um di´alogo bidirecional entre servi¸cos locais e/ou remotos, com tolerˆancia e recupera¸c˜ao face a erros e perda de pacotes. No TCP, um servi¸co ´e identificado pelo n´umero do porto a que fica associado, o que tem algumas consequˆencias menos positivas. A mais ´obvia ´e o varrimento de portos (port scanning) para posteriores tentativas de ataque a vulnerabilidades nos servi¸cos identificados/associados a esses portos. Esta tese pretende extender o conceito de endere¸camento dum determinado servi¸co associando-o primordialmente a um nome, ou seja, dotar o TCP dum servi¸co pr´oprio de resolu¸c˜ao de nomes. A fase de estabelecimento da liga¸c˜ao TCP, baseada no three-way handshake, pode ser substancialmente evolu´ıda para suportar mecanismos de resolu¸c˜ao e de autentica¸c˜ao. A solu¸c˜ao encontrada tem a seguran¸ca sempre como um aspecto presente e essencial, por forma a combater diversos tipos de ataque. A resolu¸c˜ao de nomes sugerida pode ser integrada com mecanismos de autentica¸c˜ao/valida¸c˜ao atrav´es do uso de dom´ınios de interpreta¸c˜ao (DOI - domain of interpretation). Os DOIs possibilitam uma forma flex´ıvel de adicionar mecanismos de resolu¸c˜ao e autentica¸c˜ao mais ou menos complexos ao pr´oprio estabelecimento da liga¸c˜ao TCP. ABSTRACT: Internet is the largest network deployed worldwide but besides that it’s also and essentially a way of accessing and distributing knowledge and a way to to interact with services. By using the IP routing protocol it’s possible to address and communicate with other persons, services, hosts or network enabled devices. An usual way for establishing a dialogue between internet endpoints is based on the TCP protocol, permitting a bidirectional, reliable and fault-tolerant data exchange. In TCP a service is identified by an associated port number which by itself has some less positive consequences. The obvious one consists on guessing which services are available by find out the available port numbers (port scanning) so that attacks on service vulnerabilities can take place. The purpose of this thesis is to extend the current concept used for addressing TCP services by associating them with names, or simply to provide TCP an in-band name resolution. The connection establishment phase, three-way handshake, can be improved in order to support simple name resolution mechanisms or even complex authentication. Security aspects towards avoiding attacks was a major concern that is present in the foundations of the proposed architecture. The name resolution model can be integrated with several mechanisms for authentication/validation, implemented as logic defined within domains of interpretation (DOI). DOIs allow a flexible and extensible way for adding those mechanisms to the connection establishment procedures of TCP.
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Du, Chanming. "TCP extensions for wireless networks /". Leeds, 2001. http://www.leeds.ac.uk/library/counter2/compstmsc/20002001/du.pdf.
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Yuen, Kwan Hung. "TCP performance over satellite networks /". View Abstract or Full-Text, 2003. http://library.ust.hk/cgi/db/thesis.pl?COMP%202003%20YUEN.
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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.
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Palash, Mijanur R. "MULTIPATH TCP IN WIRELESS NETWORKS". OpenSIUC, 2018. https://opensiuc.lib.siu.edu/theses/2308.
Texto completoResumen
Multipath TCP (MPTCP) is a new modification of TCP protocol which enables a client to transfer data over multiple paths simultaneously under a single TCP connection, for improved throughput and fault resilience. However, MPTCP is susceptible to some major drawbacks when applied in a wireless network. We found several cases where, despite improving individual MPTCP clients throughput, MPTCP reduces the capacity of the overall wireless network due to the mac level fairness and contention-based access schemes. Additionally, even if the bandwidth improves, employing Multipath TCP (MPTCP) in wireless networks can be energy inecient due to additional energy consumption by multiple interfaces. This creates a dilemma between bandwidth improvement and energy efficiency. This thesis research aims to solve these important issues for MPTCP in the wireless environment. We analyzed the root cause of these drawbacks and identified instances where they can arise. Two novel schemes denoted MPWiFi and kMPTCP, are developed to solve the bandwidth degradation and energy efficiency issues respectively, while maintaining the promised benefitts of MPTCP. The MPWiFi assigns dierent priorities to the subflows and aggressively suppresses some of them based on some design logic. Similarly, kMPTCP adds an additional multipath subflow only if the bandwidth requirement can't be fulllled by single path and the new subflow meets the data rate and signal strength condition. Moreover, kMPTCP keeps additional subflows as long as the signal strength remains in good range and this subflow remain mandatory to provide the necessary bandwidth to the application. These two schemes have been implemented along with Linux Kernel MPTCP implementation. Extensive real-world deployment and NS3 simulation show that the proposed schemes can eectively alleviate the adverse impacts of the MPTCP based multipath access in Wireless networks.
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Afzal, Zeeshan. "Towards Secure Multipath TCP Communication". Licentiate thesis, Karlstads universitet, Institutionen för matematik och datavetenskap (from 2013), 2017. http://urn.kb.se/resolve?urn=urn:nbn:se:kau:diva-48172.
Texto completoResumen
The evolution in networking coupled with an increasing demand to improve user experience has led to different proposals to extend the standard TCP. Multipath TCP (MPTCP) is one such extension that has the potential to overcome few inherent limitations in the standard TCP. While MPTCP's design and deployment progresses, most of the focus has been on its compatibility. The security aspect is confined to making sure that the MPTCP protocol itself offers the same security level as the standard TCP. The topic of this thesis is to investigate the unexpected security implications raised by using MPTCP in the traditional networking environment. The Internet of today has security middle-boxes that perform traffic analysis to detect intrusions and attacks. Such middle-boxes make use of different assumptions about the traffic, e.g., traffic from a single connection always arrives along the same path. This along with many other assumptions may not be true anymore with the advent of MPTCP as traffic can be fragmented and sent over multiple paths simultaneously. We investigate how practical it is to evade a security middle-box by fragmenting and sending traffic across multiple paths using MPTCP. Realistic attack traffic is used to evaluate such attacks against Snort IDS to show that these attacks are feasible. We then go on to propose possible solutions to detect such attacks and implement them in an MPTCP proxy. The proxy aims to extend the MPTCP performance advantages to servers that only support standard TCP, while ensuring that intrusions can be detected as before. Finally, we investigate the potential MPTCP scenario where security middle-boxes only have access to some of the traffic. We propose and implement an algorithm to perform intrusion detection in such situations and achieve a nearly 90% detection accuracy. Another contribution of this work is a tool, that converts IDS rules into equivalent attack traffic to automate the evaluation of a middle-box.Multipath TCP (MPTCP) is an extension to standard TCP that is close to being standardized. The design of the protocol is progressing, but most of the focus has so far been on its compatibility. The security aspect is confined to making sure that the MPTCP protocol itself offers the same security level as standard TCP. The topic of this thesis is to investigate the unexpected security implications raised by using MPTCP in a traditional networking environment. Today, the security middleboxes make use of different assumptions that may not be true anymore with the advent of MPTCP.We investigate how practical it is to evade a security middlebox by fragmenting and sending traffic across multiple paths using MPTCP. Realistic attack traffic generated from a tool that is also presented in this thesis is used to show that these attacks are feasible. We then go on to propose possible solutions to detect such attacks and implement them in an MPTCP proxy. The proxy aims to extend secure MPTCP performance advantages. We also investigate the MPTCP scenario where security middleboxes can only observe some of the traffic. We propose and implement an algorithm to perform intrusion detection in such situations and achieve a high detection accuracy.
HITS
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Chinta, Madhav. "ILC-TCP an interlayer collaboration model for TCP performance improvement in mobile and wireless environments /". [Gainesville, Fla.] : University of Florida, 2002. http://purl.fcla.edu/fcla/etd/UFE1001126.
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Nguyen, Truc Anh N., Siddharth Gangadhar y Greeshma Umapathi. "Performance Evaluation of the AeroTP Protocol in Comparison to TCP NewReno, TCP Westwood, and SCPS-TP". International Foundation for Telemetering, 2012. http://hdl.handle.net/10150/581679.
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ITC/USA 2012 Conference Proceedings / The Forty-Eighth Annual International Telemetering Conference and Technical Exhibition / October 22-25, 2012 / Town and Country Resort & Convention Center, San Diego, CaliforniaDue to the unique characteristics of highly dynamic airborne telemetry environments, TCP when deployed in such networks suffers significant performance degradation. Given the limitations of TCP, the AeroTP opportunistic transport protocol with multiple reliability modes has been developed to specifically address the issues posed by telemetry networks. In our previous work, the different modes of AeroTP have been simulated and tested using the open source ns-3 network simulator. In this paper, we use ns-3 to evaluate the overall performance of AeroTP by comparing it with well-studied TCP variants: the widely-deployed TCP NewReno and TCP Westwood designed for wireless environments. Since space networks share many similar characteristics with telemetry environments, we also compare AeroTP with SCPS-TP.
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Möller, Niels. "Automatic control in TCP over wireless". Licentiate thesis, KTH, School of Electrical Engineering (EES), 2005. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-444.
Texto completoResumen
Over the last decade, both the Internet and mobile telephony has become parts of daily life, changing the ways we communicate and search for information. These two distinct tools are now slowly merging. The topic of this thesis is TCP over wireless, and the automatic control that is used both within the system, from the link-layer power control to the end-to-end congestion control. It consists of three main contributions.
The first contribution is a proposed split-connection scheme for downloads to a mobile terminal. A wireless mobile terminal requests a file or a web page from a proxy, which in turn requests the data from a server on the Internet. During the file transfer, the radio network controller (RNC) sends radio network feedback (RNF) messages to the proxy. These messages include information about bandwidth changes over the radio channel, and the current RNC queue length. A novel control mechanism in the proxy uses this information to adjust the sending rate. The stability and convergence speed of the proxy controller is analyzed theoretically. The performance of the proposed controller is compared to end-to-end TCP Reno, using ns-2 simulations of realistic use cases. It is shown that the proxy control is able to reduce the response time experienced by users, and increase the utilization of the radio channel. The changes are loalized to the RNC and the proxy; no changes are required to the TCP implementation in terminal or server.
The second contribution is the analysis of an uplink channel using power control and link-layer retransmissions. To be able to design the link-layer mechanisms in a systematic way, good models for the link-layer processes, and their interaction with TCP, are essential.The use of link-layer retransmissions transforms a link with constant delay and random losses into a link with random delay and almost no losses. As seen from the TCP end points, the difference between such a link and a wired one is no longer the loss rate, but the packet delay distribution. Models for the power control and link-layer retransmissions on the link are used to derive packet delay distribution, and its impact on TCP performance is investigated.
The final contribution considers ways to optimize the link-layer processes. The main result is that TCP performance, over a wireless link with random retransmission delays, can be improved by adding carefully chosen artificial delays to certain packets. The artificial delays are optimized off-line and applied on-line. The additional delay that is applied to a packet depends only on the retransmission delay experienced by that same packet, and this information is available locally at the link.
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Adolfsson, Klas. "TCP performance in an EGPRS system". Thesis, Linköping University, Department of Electrical Engineering, 2003. http://urn.kb.se/resolve?urn=urn:nbn:se:liu:diva-10306.
Texto completoResumen
The most widely used protocol for providing reliable service and congestion control in the Internet is the Transmission Control Protocol (TCP). When the Internet is moving towards more use in mobile applications it is getting more important to know how TCP works for this purpose.
One of the technologies used for mobile Internet is the Enhanced General Packet Radio Service (EGPRS) extension to the popular GSM system. This thesis presents a low-level analysis of TCP performance in an EGPRS system and an overview of existing TCP, GSM and EGPRS technologies.
The bottleneck in an EGPRS system is the wireless link – the connection between the mobile phone and the GSM base station. The data transfer over the wireless link is mainly managed by the complex RLC/MAC protocol.
In this thesis, simulations were made to identify some problems with running TCP and RLC/MAC together. The simulations were made using existing EGPRS testing software together with a new TCP module. The simulation software is also briefly described in the thesis.
Additionaly, some suggestions are given in order to enhance performance, both by changing the EGPRS system and by modifying the TCP algorithms and parameters.
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Wähner, Ralf. "TCP-feste Handgelenke : Entwicklung und Simulation". Universitätsbibliothek Chemnitz, 2010. http://nbn-resolving.de/urn:nbn:de:bsz:ch1-201000745.
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TCP-feste Handgelenke dienen der Orientierung eines Werkzeuges an Portalmaschinen. Zum Einsatz kommen sie unter anderem im Bereich "Thermisches Schneiden", wenn Fasenschneiden an Blechen oder die Anarbeitung von räumlichen Konturen an Stahlrohren oder Hohlprofilen gewünscht ist. Kinematisch grenzen sie sich durch die mechanische Entkopplung zwischen Positionierung und Orientierung des Werkzeuges von anderen Maschinenstrukturen ab.
Der folgende Pro/Engineer Anwendervortrag umfasst das Erstellen einer Animation mit Pro/E Mechanismen zur Darstellung der Funktionsweise eines TCP-festen Handgelenkes. Außerdem wird die Kalibrierung dieses Handgelenkes an Hand einer Animation gezeigt, welche mit Pro/E DAO (Design Animation Option) erstellt wurde.
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Möller, Niels. "Automatic control in TCP over wireless /". Stockholm : School of Electrical Engineering, Royal Institute of Technology, 2005. http://urn.kb.se/resolve?urn=urn:nbn:se:kth:diva-444.
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Allman, Mark. "Improving TCP performance over satellite channels". Ohio : Ohio University, 1997. http://www.ohiolink.edu/etd/view.cgi?ohiou1177615641.
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Bangalore, Vijayakumar Pooja. "Piggybacking of UDP and TCP packets". Thesis, Wichita State University, 2011. http://hdl.handle.net/10057/5161.
Texto completoResumen
In this research, the concept of piggybacking has been extended to concatenate
Transmission Control Protocol (TCP) and User Datagram Protocol (UDP) packets. In any given
network scenario, voice (UDP) is given priority over data (TCP) packet. During high traffic load,
a UDP packet is processed at a higher priority, thus accumulating the TCP packets and leading to
congestion and retransmission of TCP packets. Because it is challenging to integrate UDP and
TCP packets an efficient queuing model has been proposed to concatenate the packets without
causing a negative impact on UDP or voice traffic. In this paper, we propose the algorithm and
model to concatenate UDP and TCP packets into a single packet and obtain better throughput for
TCP traffic, efficient utilization of network bandwidth. The concatenation is performed based on
parameters such as priority, destination and MTU size. The concept of discrete-time Markov
chain has been leveraged to analyze the router utilization and efficiently handle the concatenated
packets. The PCATTCP tool has been used to study the performances of UDP and TCP packets
traversing a LAN between two end hosts. These values have been used for the simulation of the
problem. Discrete Event Simulation has also been performed to determine the optimal values and
associated system parameters.Thesis (M.S.)--Wichita State University, College of Engineering, Dept. of Electrical Engineering and Computer Science.
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Arpaci, Mutlu. "Congestion avoidance in TCP/IP networks". Diss., Georgia Institute of Technology, 2001. http://hdl.handle.net/1853/15356.
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Edwan, Talal A. "Improved algorithms for TCP congestion control". Thesis, Loughborough University, 2010. https://dspace.lboro.ac.uk/2134/7141.
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Reliable and efficient data transfer on the Internet is an important issue. Since late 70's the protocol responsible for that has been the de facto standard TCP, which has proven to be successful through out the years, its self-managed congestion control algorithms have retained the stability of the Internet for decades. However, the variety of existing new technologies such as high-speed networks (e.g. fibre optics) with high-speed long-delay set-up (e.g. cross-Atlantic links) and wireless technologies have posed lots of challenges to TCP congestion control algorithms. The congestion control research community proposed solutions to most of these challenges. This dissertation adds to the existing work by: firstly tackling the highspeed long-delay problem of TCP, we propose enhancements to one of the existing TCP variants (part of Linux kernel stack). We then propose our own variant: TCP-Gentle. Secondly, tackling the challenge of differentiating the wireless loss from congestive loss in a passive way and we propose a novel loss differentiation algorithm which quantifies the noise in packet inter arrival times and use this information together with the span (ratio of maximum to minimum packet inter arrival times) to adapt the multiplicative decrease factor according to a predefined logical formula. Finally, extending the well-known drift model of TCP to account for wireless loss and some hypothetical cases (e.g. variable multiplicative decrease), we have undertaken stability analysis for the new version of the model.
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Jayananthan, Aiyathurai. "TCP performance enhancement over wireless networks". Thesis, University of Canterbury. Electrical and Computer Engineering, 2007. http://hdl.handle.net/10092/1229.
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Transmission Control Protocol (TCP) is the dominant transport protocol in the Internet and supports many of the most popular Internet applications, such as the World Wide Web (WWW), file transfer and e-mail. TCP congestion control algorithms dynamically learn the network bandwidth and delay characteristics of a network and adapt its performance to changes in traffic so as to avoid network collapse. TCP is designed to perform well in traditional wireline networks with the assumptions that packet losses are mainly due to network congestion and random bit error rate (BER) is negligible. However, networks with wireless links suffer from significant packet losses due to random bit errors and handoffs. Hence TCP performs poorly in networks with wireless links because it treats any packet loss in the network to be a result of network congestion and slows down its transmission rate, or even cause the TCP sender to experience unnecessary timeouts, further reducing its performance. The development of advance wireless networks, such as WiFi, UMTS and WiMAX, make it necessary to find ways to improve TCP's efficiency and resource utilization, as well as improve the user's experience and reduce latency times. In order to find effective solutions to this effect, packet losses across wireless links should be distinguished from congestion related packet losses. In this thesis, we concentrate on two main strategies for enabling the TCP congestion control mechanism to determine the cause for a packet loss. One is a proxy-based mechanism that monitors the radio network interface and sends radio network feedback (RNF) to the TCP sender with the status of the wireless link. The other one is an end-to-end mechanism, in which the packet error pattern is used as the system metric to fine-tune the congestion control mechanism. It also presents an analytical model of TCP with enhanced recovery mechanism for wireless environments. In a proxy-based mechanism, TCP sender is explicitly informed of any effects caused by wireless links. However, the implementation technique is network dependent. We have proposed and developed three proxy-based schemes; the radio network feedback (RNF) scheme over an 802.11 WLAN network, the radio network controller (RNC) feedback over a UMTS network and a wireless enhancement proxy (WENP) over both the 802.11 WLAN and UMTS networks. The RNF scheme is introduced at the 802.11 WLAN base station that monitors the TCP packet flows over the wireless links, detects wireless packet losses and provides feedback to the TCP sender using one of the TCP header reserved control bits, called RNF flag. TCP Reno is modified to utilize the radio network feedback to distinguish the losses due to wireless effects form the congestion and fine-tuned to perform wireless enhanced fast retransmit and fast recovery mechanisms. The RNF scheme is implemented using the OPNET tool, and the simulation results show that the TCP performance is significantly improved. The RNC feedback mechanism, similar to the RNF scheme, is developed and implemented in a UMTS network. The GPRS Tunneling Protocol (GTP) layer of the UMTS Radio Network Control (RNC) protocol stack was modified to detect and notify the TCP sender of the wireless packet losses, which is the main difference between the RNF and RNC mechanisms. The simulation results shows that the RNC feedback mechanism significantly improves the TCP performance compared to that of standard TCP over UMTS. The wireless enhancement proxy (WENP) is developed to minimize spurious TCP timeouts over wireless networks and implemented in both 802.11 WLAN and UMTS networks. WENP extends the proposed RNF and RNC feedback mechanisms to detect both wireless packet losses and large delays across the wireless link, and to notify the TCP sender of these events with the aid of two reserved bits in the TCP header. TCP Reno is further modified to utilize the WENP feedback to distinguish both wireless packet losses from congestion losses and spurious timeouts from normal timeouts. It is also fine-tuned to perform both the wireless enhanced fast retransmit and fast recovery mechanism and the timeout mechanism. The simulation results demonstrate that the proposed scheme markedly improves the TCP performance compared to that of standard WLAN and UMTS implementations. An end-to-end early packet loss recovery (EPLR) mechanism that modifies the TCP Reno fast retransmit algorithm to detect packet losses early and to speed up the packet recovery process to reduce the number of TCP timeouts over networks with heavy packet losses, such as wireless networks is also presented. TCP Reno with EPLR scheme is implemented in a UMTS network and its performance is compared with that of TCP Reno and New Reno. Simulation results shows that Reno with EPLR improves the TCP performance and application response time significantly compared to that of both Reno and New Reno by reducing the TCP timeouts, which is the main cause of degradation of the TCP performance in a wireless environment. Finally, we develop an analytical TCP throughput model with enhanced TCP Reno fast retransmit algorithm to avoid timeouts. The model captures the TCP fast retransmit mechanism and expresses the steady state congestion window and throughput as a function of network utilization factor, round trip time (RTT) and loss rate. Another new feature added to the model is dynamic adjustment of the congestion window size depending on the packet drop rates. This speeds up the packet recovery process and reduces the number of TCP timeouts over networks with heavy packet losses. The proposed model is implemented over a UMTS network and its performance is compared with that of TCP Reno. Simulation results show that the proposed model reduces the TCP timeouts and improves the TCP performance compared to that of TCP Reno. It is also found that the model provides a very good match to the steady-state congestion window behavior.
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Zayes, Pedro A. (Pedro Angel) 1975. "Analyzing the behavior of TCP implementations". Thesis, Massachusetts Institute of Technology, 1998. http://hdl.handle.net/1721.1/47627.
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Thesis (S.B. and M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1998.Includes bibliographical references (p. 72-73).
by Pedro A. Zayes.
S.B.and M.Eng.
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Lecumberri, David (Lecumberri Iriarte) 1973. "TCP performance improvements over heterogeneous networks". Thesis, Massachusetts Institute of Technology, 2000. http://hdl.handle.net/1721.1/86444.
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Kotze, Johannes Marthinus Albertus. "TCP/IP facilitated flexible robotics controller". Thesis, Stellenbosch : University of Stellenbosch, 2007. http://hdl.handle.net/10019.1/2665.
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Thesis (MScEng (Industrial Engineering))--University of Stellenbosch, 2008.Robot control by TCP/IP communication is investigated in this thesis for existing robots used in tertiary education. The request for newer software for robotic computer control came from the University of Stellenbosch Industrial Engineering department where existing software dating back to 1988 is still in use. A thorough investigation into the research and technologies available is followed by a discussion on the proposed software to adhere to the requirement of compatibility with existing languages in use by the department and provide tools to assist in future research in robotic manipulators and control. The proposed software solution uses a client/server model running over an IP-based network providing online and offline programming with visual feedback by means of video streaming and 3D simulations, developed as separate modules combined into an effective tool for future research and development.
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Hu, Xuenan y Tingxian Zhou. "TCP EXTENSIONS FOR A SATELLITE CHANNEL". International Foundation for Telemetering, 2003. http://hdl.handle.net/10150/607480.
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International Telemetering Conference Proceedings / October 20-23, 2003 / Riviera Hotel and Convention Center, Las Vegas, NevadaThe usage of Internet is explosively growing. Satellite has become a choice solution breaking through the bandwidth bottleneck and the terrain limit. TCP, which is well suited to terrestrial networks, performs poorly on a satellite channel. The reduced efficiency and QoS(Quality of Service) mainly result from three characteristics of a satellite link: higher bit error rate, the high latency, asymmetry. For this issue, the paper presents connection-subsection network architecture, and brings forward S-TCP based on the architecture.
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Pereira, Felipe Massia. "Protocolos TCP/IP para sistemas embarcados". [s.n.], 2003. http://repositorio.unicamp.br/jspui/handle/REPOSIP/276429.
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Orientador : Celio Cardoso GuimarãesDissertação (mestrado) - Universidade Estadual de Campinas, Instituto de Matematica, Estatistica e Computação Cientifica
Made available in DSpace on 2018-08-03T14:41:09Z (GMT). No. of bitstreams: 1 Pereira_FelipeMassia_M.pdf: 1232996 bytes, checksum: 6534b5a29c260209081ce1c6168f6575 (MD5) Previous issue date: 2003
Mestrado
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Lien, Kai-Wen. "Enhancing TCP delivery over wireless networks". Thesis, University of Southampton, 2009. https://eprints.soton.ac.uk/66207/.
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Wireless communication has become a significant life style in the daily use. The wireless communication can be used to extend the service of wired communication. Based on the idea of simplicity, Transmission Control Protocol (TCP) has been used widely over wired network. When network applications take place using wireless link, TCP is still useful because partial wired connection might be necessary. Although wired and wireless communication do share something in common, they have distinct features. Therefore, TCP needs to be adjusted to fit in the wireless environment. This research aims to enhance wireless TCP performance. In order to study network protocols and behaviours, network simulators are often used for researchers to configure and monitor the network factors and system states. Unfortunately, most network simulators cannot demonstrate what the real network does. They are applications. In this research, a network simulator based on the real Linux TCP/IP stacks is proposed. This simulator is able to not only simulate the wired network, but also allow users to extend its structure for live wireless emulation. By means of simulator and emulator, researchers can understand and configure detail factors for further experiments. Eventually, a new wireless TCP enhancement is proposed. In this research, some contributions are delivered. Firstly, a network simulator based on Linux TCP stacks is implemented. Secondly, a wireless emulator and test environment are built, so the wireless factors can be configured and performance can be monitored. Thirdly, a wireless TCP mechanism, TCP NewZag, is proposed. Finally, several experiments to show the value of NewZag are reported in the thesis.
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Brennan, Stephen. "Exploring Alternative Routes Using Multipath TCP". Case Western Reserve University School of Graduate Studies / OhioLINK, 2017. http://rave.ohiolink.edu/etdc/view?acc_num=case1497013802717687.
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Zagorodnov, Dmitrii. "Engineering fault-tolerant TCP/IP services /". Diss., Connect to a 24 p. preview or request complete full text in PDF format. Access restricted to UC campuses, 2004. http://wwwlib.umi.com/cr/ucsd/fullcit?p3137231.
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Di, Fiore Rosa Divina. "Criteri di valutazione del protocollo TCP". Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2014. http://amslaurea.unibo.it/6696/.
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La diffusione di Internet negli ultimi anni e lo sviluppo sempre crescente nell'ambito delle reti di telecomunicazione, hanno portato oggi ad una vera e propria esplosione della diffusione di nuove tecnologie di trasmissione. Inizialmente il protocollo di trasporto dati, TCP, non era stato pensata per operare in scenari diversi da quello della rete fissa. Con l'introduzione di nuovi scenari, come quello wireless, wimax e satellitare, si è notato come le prestazioni peggiorino in questi ambienti. Proprio per questo, il protocollo TCP ha subito parecchie modifiche negli anni e sono state realizzate alternative atte a migliorare l'inefficienza del protocollo. Le modifiche, a cui il TCP è stato sottoposto, sono basate su alcuni criteri di valutazione e l'introduzione di meccanismi come il controllo del flusso o il controllo di congestione in modo da migliorare le performance in ambienti “ostili”. Molti ricercatori si sono concentrati nello studio e modifica di questi nuovi meccanismi cercando di adattare al meglio il TCP secondo diversi scenari di rete, trascurando così altri criteri un pò meno noti.
Dopo aver introdotto lo scenario con la descrizione del protocollo TCP, andremo a descrivere e illustrare questi “nuovi criteri” presentando alcuni recenti studi effettuati, in seguito andremo a presentare un nuova versione del protocollo chiamata Early Warning TCP e nell'ultimo capitolo andremo a presentare delle conclusioni degli studi presentati.
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Pazzaglia, Gessica. "Criteri di valutazione del protocollo TCP". Bachelor's thesis, Alma Mater Studiorum - Università di Bologna, 2014. http://amslaurea.unibo.it/7224/.
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Song, Lin. "Multipath approaches to avoiding TCP Incast". Diss., University of Iowa, 2017. https://ir.uiowa.edu/etd/6859.
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TCP was conceived to ensure reliable node-to-node communication in moderate-bandwidth, moderate-latency, WANs. As it is now a mature Internet standard, it is the default connection-oriented protocol in networks built from commodity components, including Internet data centers. Data centers, however, rely on high-bandwidth, low-latency networks for communication. Moreover, their communication patterns, especially those generated by distributed applications such as MapReduce, often take the form of synchronous multi-node to node bursts. Under the right conditions, the network switch buffer overflow losses induced by these bursts confuse TCP's feedback mechanisms to the point that TCP throughput collapses. This collapse, termed TCP Incast, results in gross underutilization of link capacities, significantly degrading application performance.
Conventional approaches to mitigating Incast have focused on single-path solutions, for instance, adjusting TCP's receive windows and timers, modifying the protocol itself, or adopting explicit congestion notifications. This thesis explores complementary multi-path approaches to avoiding Incast's onset. The principal idea is to use the regularity and high connectivity of typical data center networks, such as the increasingly popular fat-tree topology, to better distribute multi-node to node bursts across the available paths, thereby avoiding the switch buffer overflows that induce TCP Incast.
The thesis's main contributions are: (1) development of new oblivious, multi-path, routing schemes for fat-tree networks, (2) derivation of relations between the schemes and Incast's onset, and (3) investigation of a novel "front-back" approach to minimizing the packet reordering introduced by multipath routing. Formal analyses are focused on relating schemes' worst-case loading of certain network resources - expressed as oblivious performance ratios (OPRs) - to Incast's onset. Potential benefits are assessed through ns-3 simulations on fat-trees under a variety of communication patterns. Results indicate that over a variety of experimental conditions, the proposed schemes reduce the incidence of TCP Incast compared to standard routing schemes.
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Silva, Roseli Maria da. "Desenvolvimento de Serviços em Redes TCP/IP - Projeto de um servidor de modem para redes TCP/IP". Universidade de São Paulo, 1992. http://www.teses.usp.br/teses/disponiveis/55/55134/tde-23112018-164242/.
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A crescente utilização de sistemas computacionais distribuídos construídos sobre redes de computadores é apresentada. Este projeto considera o ambiente de rede de SUNs existente no Departamento de Ciências de Computação e Estatística do 1CMSC - USP. Este trabalho destaca a utilização de mecanismos fornecidos pelo sistema SunOS para o desenvolvimento de novos serviços de rede. O servidor de modem projetado tem por finalidade ilustrar o acréscimo de novos serviços no ambiente SunOS, permitir a interconexão da rede local com outras redes remotas (via linha telefônica) e a utilização dos serviços oferecidos por outros sistemas possibilitando o compartilhamento do modem e da linha telefônica pelos usuários. A idéia básica do servidor de modem é permitir que qualquer estação da rede tenha acesso a serviços centralizados no servidor. Também são analisados e descritos recursos para o estabelecimento de conexão com outros sistemas e protocolos de comunicação - com ênfase ao conjunto conhecido como TCP/IP. Da maneira que o projeto foi desenvolvido existe uma grande transparência para o usuário da localização do servidor - os serviços fornecidos por meio de servidores como \"inetd\" e \"portmapper\" são introduzidos como elementos para fornecer portabilidade, modularidade e transparência aos serviços.The growing use of distributed systems built over computer networks is presented. Having the design of a modern server as the focusing point , thios work is set to describe the mechanisms provided by SunOS operating systems for the development of networks services. Communication protocols are described with special attention to the suite of protocols known as TCP/IP. The inetd and portmapper services provided by SunOS are introduced as elements to provide portability, modularity and transparency to services. The design of a modern server is then presented as an example.
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GUPTA, RAHUL. "A TCP ANALYSIS: IMPACT OF RECEIVER PERCEIVED INFORMATION ON THE PERFORMANCE OF TCP OVER Ad Hoc NETWORKS". University of Cincinnati / OhioLINK, 2002. http://rave.ohiolink.edu/etdc/view?acc_num=ucin1037305279.
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Gupta, Rahul. "A TCP analysis impact of receiver perceived information on the performance of TCP over ad hoc networks /". Cincinnati, Ohio : University of Cincinnati, 2002. http://rave.ohiolink.edu/etdc/view?acc%5Fnum=ucin1037305279.
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Holl, David J. "A performance analysis of TCP and STP implementations and proposals for new QoS classes for TCP/IP". Link to electronic thesis, 2003. http://www.wpi.edu/Pubs/ETD/Available/etd-0501103-111419.
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Thesis (M.S.)--Worcester Polytechnic Institute.Keywords: TCP; RED; satellite; PEP; STP; performance enhancing proxy; segment caching; IP-ABR; Internet; bandwidth reservation; IP-VBR; congestion avoidance; bandwidth sharing. Includes bibliographical references (p. 98-99).
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Cho, Soohyun. "Congestion control schemes for single and parallel TCP flows in high bandwidth-delay product networks". Texas A&M University, 2003. http://hdl.handle.net/1969.1/3935.
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In this work, we focus on congestion control mechanisms in Transmission Control
Protocol (TCP) for emerging very-high bandwidth-delay product networks and suggest
several congestion control schemes for parallel and single-flow TCP. Recently, several
high-speed TCP proposals have been suggested to overcome the limited throughput
achievable by single-flow TCP by modifying its congestion control mechanisms.
In the meantime, users overcome the throughput limitations in high bandwidth-delay
product networks by using multiple parallel TCP flows, without modifying TCP itself.
However, the evident lack of fairness between the high-speed TCP proposals (or
parallel TCP) and existing standard TCP has increasingly become an issue.
In many scenarios where flows require high throughput, such as grid computing
or content distribution networks, often multiple connections go to the same or nearby
destinations and tend to share long portions of paths (and bottlenecks). In such cases
benefits can be gained by sharing congestion information. To take advantage of this
additional information, we first propose a collaborative congestion control scheme for
parallel TCP flows. Although the use of parallel TCP flows is an easy and effective
way for reliable high-speed data transfer, parallel TCP flows are inherently unfair
with respect to single TCP flows. In this thesis we propose, implement, and evaluate
a natural extension for aggregated aggressiveness control in parallel TCP flows.
To improve the effectiveness of single TCP flows over high bandwidth-delay product networks without causing fairness problems, we suggest a new TCP congestion
control scheme that effectively and fairly utilizes high bandwidth-delay product networks
by adaptively controlling the flowÂs aggressiveness according to network situations
using a competition detection mechanism. We argue that competition detection
is more appropriate than congestion detection or bandwidth estimation. We further
extend the adaptive aggressiveness control mechanism and the competition detection
mechanism from single flows to parallel flows. In this way we achieve adaptive aggregated
aggressiveness control. Our evaluations show that the resulting implementation
is effective and fair.
As a result, we show that single or parallel TCP flows in end-hosts can achieve
high performance over emerging high bandwidth-delay product networks without requiring
special support from networks or modifications to receivers.
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Sup, Luciano Mauro Arley. "Novas metodologias AQM e TCP visando a eficiência de fluxos de controle UDP e dados TCP/IP compartilhados". reponame:Repositório Institucional da UnB, 2017. http://repositorio.unb.br/handle/10482/31307.
Texto completoResumen
Tese (doutorado)—Universidade de Brasília, Faculdade de Tecnologia, Departamento de Engenharia Elétrica, 2017.Submitted by Raquel Almeida (raquel.df13@gmail.com) on 2018-02-21T17:02:16Z No. of bitstreams: 1 2017_LucianoMauroArleySup_PARCIAL.pdf: 4566541 bytes, checksum: ed7c646700fed8a4900c009de162ecfc (MD5)
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Coordenação de Aperfeiçoamento de Pessoal de Nível Superior (CAPES).
Na atualidade a Internet se tornou uma rede capaz de interconectar diversos tipos de usuários, dispositivos, casas, edifícios, plantas industriais, automóveis, hospitais, escolas, paradigma conhecido como Internet das Coisas (IoC). Convertendo-se assim em um meio que abriga fluxos de dados TCP/IP e fluxos UDP de sistemas de controle em rede denominados NCS (Networked Control Systems) utilizados em diversas aplicações, tais como, automação industrial, automação predial, telecirurgias, smart grid e smart city. Para estudar ambos os tipos de fluxos em uma abordagem concomitante, neste trabalho, primeiramente foi desenvolvida uma metodologia que admite modelar e simular topologias de redes de comunicação junto com seus protocolos e algoritmos usando o UPPAAL, uma ferramenta de software que admite modelar sistemas físicos e lógicos através de autômatos temporizados e também fazer simulações e verificações determinísticas e estatísticas do modelo. Então, foram modelados sistemas de controle através da Internet junto com outros fluxos TCP/IP genéricos, e foi observado e estudado como as características do sistema de comunicação afetam o desempenho do sistema de controle em diferentes cenários de simulações e com diferentes técnicas AQM (Active Queue Management) implementadas nos roteadores. Neste contexto, com o objetivo de corrigir esses efeitos, e visando à eficiência de ambos os tipos de fluxos (maior vazão para fluxos TCP/IP e menor ITAE (Integral Time-weighted Absolute Error) para a NCS) foi desenvolvida uma nova metodologia que combina ações AQM com ações TCP/IP, chamada ENCN (Explicit Non-Congestion Notification) cujo desempenho foi comparado com as técnicas AQM Drop Tail, RED, CoDel e PIE trabalhando em conjunto com TCP-Reno, e com os protocolos TCP-Jersey, DCTCP e E-DCTCP (todos modelados e simulados no UPPAAL). Resultados de simulações e verificações estatísticas mostraram que comparado com essas metodologias o ENCN melhorou tanto a vazão e a justiça dos fluxos TCP/IP quanto o ITAE da NCS (fluxo UDP). Porém, o ENCN utiliza alguns recursos tecnológicos (bits para notificar não congestionamento) que nem sempre estão disponíveis na prática. Para superar essa limitação do ENCN, foi desenvolvida uma metodologia chamada ANCE (Acknowledge-based Non- Congestion Estimation) a qual trabalha de forma análoga a ENCN, porém, utiliza o valor estimado do tamanho da fila ao invés de utilizar uma notificação explícita de não congestionamento advinda dos roteadores, o que torna essa técnica mais viável para implementação. Finalmente, para superar algumas limitações de ANCE, foi desenvolvido um protocolo de transporte de dados chamado TCP-Puerto-Londero que a diferença de ANCE faz um ajuste adaptativo da janela de transmissão de dados em função de um atraso relativo no caminho de ida, ao invés de utilizar o RTT. Resultados de simulações mostraram que apesar de não utilizar recursos para fazer notificações explícitas de não congestionamento, o TCPPuerto-Londero fornece um desempenho comparável ao ENCN, superando o TCP-Jersey em 12,03 % e o E-DCTCP em 4,21 % em termos de vazão para os fluxos TCP/IP, e reduzindo o ITAE da NCS em 36,84 %, 36,68 % e 4,16 % em relação ao TCP-Jersey, o E-DCTCP e o ENCN, respectivamente. Esse bom desempenho do TCP-Puerto-Londero também foi observado nas verificações estatísticas do modelo.
Actually the Internet has become a network capable of interconnecting several types of users, objects, houses, buildings, industrial plants, automobiles, hospitals, schools, this paradigm is known as Internet of Things (IoC). Thus, the Internet has becoming a communication medium that hosts TCP/IP data flows and networked control systems (NCS) UDP flows used in several applications, such as, industrial automation, building automation, remote surgery , smart grid and Smart city. To study both types of flows in a concomitant approach, in this work, a modeling methodology was developed that supports modeling and simulation of communication network topologies along with their protocols and others algorithms using the UPPAAL, a software tool that admit modeling of physical and logical systems through timed automata and make simulations and statistical verifications. Thus, control systems over the Internet were modeled along with other generic flows on which we can observe and study how the communication network features affect control performance behavior for different simulation scenarios and with different AQM (Active Queue Management) techniques implemented in the routers. Then, in order to improve the tradeoff between TCP-throughput and ITAE (Integral Time-weighted Absolute Error) for the NCS that share de same network topology, we've developed a new AQM technique called ENCN (Explicit Non-Congestion Notification) whose performance was compared with TCP-Jersey, DCTCP and, E-DCTCP ECN based transport layer protocols, and with Drop Tail, RED, CoDel, and PIE schemes implemented joint with TCP-Reno protocol (all them modeled in UPPAAL). Simulations and statistical verifications show that the ENCN provides better throughput and fairness for TCP/IP flows, and ITAE for NCS UDP flow compared to other methodologies. However, the ENCN uses some technological resources (bits to notify non-congestion) that are not always available. In order to overcome this ENCN limitation we developed a methodology called ANCE (Acknowledge-Based Non- Congestion Estimation), which work like ENCN but making inferences about queue length instead to using explicit non-congestion notification coming from the routers, which makes this technique more feasible for implementation. Finally, in order to overcome some ANCE limitations we developed a new transport layer protocol called TCP-Puerto-Londero, which rather than using the RTT, makes an adaptive adjustment of the congestion window as a function of a relative delay in the forward path. Simulation results showed that TCP-Puerto- Londero provides performance comparable to ENCN, overcoming TCP-Jersey by 12.03 % and the E-DCTCP in 4.21 % in terms of throughput for TCP/IP flows and reducing the ITAE of the NCS UDP flow by 36.84 %, 36.68 % and 4.16 % in relation to the TCP-Jersey, E-DCTCP, and ENCN, respectively. This good performance of TCP-Puerto-Londero was also observed in the statistical verifications.