Academic literature on the topic 'Computer networks Packet switching (Data transmission) TCP/IP (Computer network protocol)'
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Journal articles on the topic "Computer networks Packet switching (Data transmission) TCP/IP (Computer network protocol)"
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Prabadevi, B., and N. Jeyanthi. "Security Solution for ARP Cache Poisoning Attacks in Large Data Centre Networks." Cybernetics and Information Technologies 17, no. 4 (November 27, 2017): 69–86. http://dx.doi.org/10.1515/cait-2017-0042.
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AbstractThe bridge protocol (Address Resolution Protocol) ARP, integrating Ethernet (Layer 2) and IP protocol (Layer 3) plays a vital role in TCP/IP communication since ARP packet is the first packet generated during any TCP/IP communications and they are the first traffic from the host. In the large data center, as the size of the broadcast domain (i.e., number of hosts on the network) increases consequently the broadcast traffic from the communication protocols like ARP also increases. This paper addresses the problem faced by Layer 2 protocols like insecured communication, scalability issues and VM migration issues. The proposed system addresses these issues by introducing two new types of messaging with traditional ARP and also combat the ARP Cache poisoning attacks like host impersonation, MITM, Distributed DoS by making ARP stateful. The components of the proposed methodology first start the process by decoding the packets, updates the invalid entry made by the user with Timestamp feature and messages being introduced. The system has been implemented and compared with various existing solutions.
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Park, Byungjoo, Ankyu Hwang, and Haniph Latchman. "Design of Optimized Multimedia Data Streaming Management Using OMDSM over Mobile Networks." Mobile Information Systems 2017 (2017): 1–13. http://dx.doi.org/10.1155/2017/2867127.
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Mobility management is an essential challenge for supporting reliable multimedia data streaming over wireless and mobile networks in the Internet of Things (IoT) for location-based mobile marketing applications. The communications among mobile nodes for IoT need to have a seamless handover for delivering high quality multimedia services. The Internet Engineering Task Force (IETF) mobility management schemes are the proposals for handling the routing of IPv6 packets to mobile nodes that have moved away from their home network. However, the standard mobility management scheme cannot prevent packet losses due to longer handover latency. In this article, a new enhanced data streaming route optimization scheme is introduced that uses an optimized Transmission Control Protocol (TCP) realignment algorithm in order to prevent the packet disordering problem whenever the nodes in the IoT environment are communicating with each other. With the proposed scheme, data packets sequence realignment can be prevented, the packet traffic speed can be controlled, and the TCP performance can be improved. The experimental results show that managing the packet order in proposed new scheme remarkably increases the overall TCP performance over mobile networks within the IoT environment thus ensuring the high quality of service (QoS) for multimedia data streaming in location-based mobile marketing applications.
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Zheng, Yani, Gaurav Dhiman, Ashutosh Sharma, Amit Sharma, and Mohd Asif Shah. "An IoT-Based Water Level Detection System Enabling Fuzzy Logic Control and Optical Fiber Sensor." Security and Communication Networks 2021 (August 27, 2021): 1–11. http://dx.doi.org/10.1155/2021/4229013.
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The usage of wireless sensors has become widespread for the collection of data for various Internet of Things (IoT) products. Specific wireless sensors use optical fiber technology as transmission media and lightwave signals as carriers, showing the advantages of antielectromagnetic interference, high sensitivity, and strong reliability. Hence, their application in IoT systems becomes a research hotspot. In this article, multiple optical fiber sensors are constructed as an IoT detection system, and a Transmission Control Protocol (TCP)/Internet Protocol (IP) communication stack is used for the sensor module. Furthermore, design of gateway module, data server, and monitoring module is established in order to run the data server in the Windows system and communicate across the network segments. Furthermore, the optical fiber sensor is connected to the development board with WiFi, meanwhile considering the optical fiber wireless network’s congestion problem. The fuzzy logic concept is introduced from the perspective of cache occupancy, and a fiber sensor’s network congestion control algorithm is proposed. In the experiment, the IoT detection system with multiple optical fiber sensors is used for water level detection, and the sensor’s real-time data detected by the User Interface (UI) are consistent with the feedback results. The proposed method is also compared with the SenTCP algorithm and the CODA algorithm, and it was observed that the proposed network congestion control algorithm based on the fuzzy logic can improve network throughput and reduce the network data packet loss.
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Irawan, Addie, Mohammad Fadhil Abas, and Nurulfadzilah Hasan. "Robot Local Network Using TQS Protocol for Land-to-Underwater Communications." Journal of Telecommunications and Information Technology 1 (March 29, 2019): 23–30. http://dx.doi.org/10.26636/jtit.2019.125818.
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This paper presents a model and an analysis of the Tag QoS switching (TQS) protocol proposed for heterogeneous robots operating in different environments. Collaborative control is topic that is widely discussed in multirobot task allocation (MRTA) – an area which includes establishing network communication between each of the connected robots. Therefore, this research focuses on classifying, prioritizing and analyzing performance of the robot local network (RLN) model which comprises a point-to-point topology network between robot peers (nodes) in the air, on land, and under water. The proposed TQS protocol was inspired by multiprotocol label switching (MPLS), achieving a quality of service (QoS) where swapping and labeling operations involving the data packet header were applied. The OMNET++ discrete event simulator was used to analyze the percentage of losses, average access delay, and throughput of the transmitted data in different classes of service (CoS), in a line of transmission between underwater and land environments. The results show that inferior data transmission performance has the lowest priority with low bitrates and extremely high data packet loss rates when the network traffic was busy. On the other hand, simulation results for the highest CoS data forwarding show that its performance was not affected by different data transmission rates characterizing different mediums and environments.
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Atmadja, Martono Dwi. "Single Board Computer Applications as Multi-Server VoIP." International Journal for Research in Applied Science and Engineering Technology 9, no. VII (July 15, 2021): 1023–28. http://dx.doi.org/10.22214/ijraset.2021.36512.
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Telecommunication technology is developing along with information technology and several innovations in several audio and data transmission and reception techniques. Innovation and communication technology are hoped to be able to create efficiencies in regards to time, equipment, and cost. The Public Switched Telephone Network (PSTN) telephone technology has experienced integration towards communication using Internet Protocol (IP) networks, better known as Voice over Internet Protocol (VoIP). VoIP Technology transmits conversations digitally through IP-based networks, such as internet networks, Wide Area Networks (WAN), and Local Area Networks (LAN). However, the VoIP cannot fully replace PSTN due to several weaknesses, such as delay, jitter, packet loss, as well as security and echo. Telephones calls using VoIP technology are executed using terminals in the form of computer devices or existing analogue telephones. The benefit of VoIP is that it can be set in all ethernet and IP addresses. Prefixes can be applied for inter-server placements as inter-building telephone networks without the addition of inefficient new cables on single board computers with Elastix installed. Prefix and non-prefix analysis on servers from single board computers can be tested using QoS for bandwidth, jitter, and packet loss codec. The installation of 6 clients, or 3 simultaneous calls resulted in a packet loss value in the prefix Speex codex of 2.34%. The bandwidth in the prefix PCMU codec has an average value of 82.3Kbps, and a non-prefix value of 79.3Kbps, in accordance to the codec standards in the VoIP. The lowest jitter was found in the non-prefix PCMU codec with an average of 51.05ms, with the highest jitter for the prefix Speex codec being 314.65ms.
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Memon, Sheeba, Jiawei Huang, and Hussain Saajid. "Gentle Slow Start to Alleviate TCP Incast in Data Center Networks." Symmetry 11, no. 2 (January 27, 2019): 138. http://dx.doi.org/10.3390/sym11020138.
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Modern data center networks typically adopt symmetric topologies, such as leaf-spine and fat-tree. When a large number of transmission control protocol (TCP) flows in data center networks send data to the same receiver, the congestion collapse, called TCP Incast, frequently happens because of the huge packet losses and Time-Out. To address the TCP Incast issue, we firstly demonstrate that adjusting the increasing speed of the congestion window during the slow start phase is crucially important. Then we propose the Gentle Slow Start (GSS) algorithm, which adjusts the congestion window according to real-time congestion state in a gentle manner and smoothly switches from slow start to congestion avoidance phase. Furthermore, we present the implementation and design of Gentle Slow Start and also integrate it into the state-of-the-art data center transport protocols. The test results show that GSS effectively decreases the Incast probability and increases the network goodput by average 8x.
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Lee, Sungwon, Yeongjoon Bae, Muhammad Toaha Raza Khan, Junho Seo, and Dongkyun Kim. "Avoiding Spurious Retransmission over Flooding-Based Routing Protocol for Underwater Sensor Networks." Wireless Communications and Mobile Computing 2020 (August 25, 2020): 1–9. http://dx.doi.org/10.1155/2020/8839541.
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In underwater wireless sensor networks (UWSN), acoustic communication naturally introduces challenges such as long propagation delay and high packet loss. The flooding-based routing protocol can address these challenges with its multipath characteristics. As in flooding-based routing, due to multipath propagation mechanism, not only DATA but also ACK messages are transmitted through multiple routes however still some packet loss will degrade the performance. So, to provide high reliability of message delivery, an efficient retransmission mechanism is inevitable. Though, if the network uses conventional transport layer protocol such as TCP, it will suffer a spurious retransmission problem as TCP was originally not designed for the multipath environment. In this paper, we propose route discrimination for flooding-based routing to reduce spurious retransmission in UWSN to solve the limitation. The notion of ACK copies waiting time (ACWT) is utilized which is selectively updated based on the similarity of paths of transmission of ACK message copies. We also improved our previous solution that lacks flexibility to cope with dynamic link error characteristics. Through evaluation, we verified that our new scheme achieves the performance improvements of 14%~84% in terms of retransmission ratio compared to the previous research.
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Xu, Yong, Hong Ni, and Xiaoyong Zhu. "An Effective Transmission Scheme Based on Early Congestion Detection for Information-Centric Network." Electronics 10, no. 18 (September 9, 2021): 2205. http://dx.doi.org/10.3390/electronics10182205.
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As one of the candidates for future network architecture, Information-Centric Networking (ICN) has revolutionized the manner of content retrieval by transforming the communication mode from host-centric to information-centric. Unlike a traditional TCP/IP network, ICN uses a location-independent name to identify content and takes a receiver-driven model to retrieve the content. Moreover, ICN routers not only perform a forwarding function but also act as content providers due to pervasive in-network caching. The network traffic is more complicated and routers are more prone to congestion. These distinguished characteristics pose new challenges to ICN transmission control mechanism. In this paper, we propose an effective transmission scheme by combining the receiver-driven transport protocol and the router-driven congestion detection mechanism. We first outline the process of content retrieval and transmission in an IP-compatible ICN architecture and propose a practical receiver-driven transport protocol. Then, we present an early congestion detection mechanism applied on ICN routers based on an improved Active Queue Management (AQM) algorithm and design a receiver-driven congestion control algorithm. Finally, experiment results show that the proposed transmission scheme can maintain high bandwidth utilization and significantly reduce transmission delay and packet loss rate.
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Reddy, N. Ramanjaneya, Chenna Reddy Pakanati, and M. Padmavathamma. "An Enhanced Queue Management Scheme for Eradicating Congestion of TFRC over Wired Environment." International Journal of Electrical and Computer Engineering (IJECE) 7, no. 3 (June 1, 2017): 1347. http://dx.doi.org/10.11591/ijece.v7i3.pp1347-1354.
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<p class="Abstract">To accomplish increasing real time requirements, user applications have to send different kinds of data with different speeds over the internet. To effectuate the aims of the computer networks, several protocols have been added to TCP/IP protocol suite. Transport layer has to implement emerging techniques to transfer huge amount of data like multimedia streaming. To transmit multimedia applications, one of the suitable congestion control mechanisms in transport layer is TCP Friendly Rate Control Protocol (TFRC). It controls congestion based on its equation. To get more smoothed throughput, intermediate nodes (like Routers. etc.) have to use suitable procedures in all real time situations. To eradicate the level of congestion in the network, we introduce enhanced Holt-Winters equations to RED queue management algorithm and applied to TFRC. The simulation results have shown that this strategy reduces packet loss and increases throughput.</p>
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Mudassir, Mumajjed Ul, and M. Iram Baig. "MFVL HCCA: A Modified Fast-Vegas-LIA Hybrid Congestion Control Algorithm for MPTCP Traffic Flows in Multihomed Smart Gas IoT Networks." Electronics 10, no. 6 (March 18, 2021): 711. http://dx.doi.org/10.3390/electronics10060711.
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Multihomed smart gas meters are Internet of Things (IoT) devices that transmit information wirelessly to a cloud or remote database via multiple network paths. The information is utilized by the smart gas grid for accurate load forecasting and several other important tasks. With the rapid growth in such smart IoT networks and data rates, reliable transport layer protocols with efficient congestion control algorithms are required. The small Transmission Control Protocol/Internet Protocol (TCP/IP) stacks designed for IoT devices still lack efficient congestion control schemes. Multipath transmission control protocol (MPTCP) based congestion control algorithms are among the recent research topics. Many coupled and uncoupled congestion control algorithms have been proposed by researchers. The default congestion control algorithm for MPTCP is coupled congestion control by using the linked-increases algorithm (LIA). In battery powered smart meters, packet retransmissions consume extra power and low goodput results in poor system performance. In this study, we propose a modified Fast-Vegas-LIA hybrid congestion control algorithm (MFVL HCCA) for MPTCP by considering the requirements of a smart gas grid. Our novel algorithm operates in uncoupled congestion control mode as long as there is no shared bottleneck and switches to coupled congestion control mode otherwise. We have presented the details of our proposed model and compared the simulation results with the default coupled congestion control for MPTCP. Our proposed algorithm in uncoupled mode shows a decrease in packet loss up to 50% and increase in average goodput up to 30%.
Dissertations / Theses on the topic "Computer networks Packet switching (Data transmission) TCP/IP (Computer network protocol)"
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Low, Douglas Wai Kok. "Network processor memory hierarchy designs for IP packet classification /." Thesis, Connect to this title online; UW restricted, 2005. http://hdl.handle.net/1773/6973.
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Chan, Ming Kit. "Active queue management schemes using a capture-recapture model /." View Abstract or Full-Text, 2002. http://library.ust.hk/cgi/db/thesis.pl?COMP%202002%20CHAN.
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Thesis (M. Phil.)--Hong Kong University of Science and Technology, 2002.Includes bibliographical references (leaves 58-61). Also available in electronic version. Access restricted to campus users.
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Miller, Alan Henry David. "Best effort measurement based congestion control." Thesis, Connect to e-thesis, 2001. http://theses.gla.ac.uk/1015/.
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Thesis (Ph.D.) -- University of Glasgow, 2001.Includes bibliographical references (p.i-xv). Print version also available. Mode of access : World Wide Web. System requirments : Adobe Acrobat reader reuired to view PDF document.
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Talau, Marcos. "NGWA: esquema de controle de congestionamento para TCP baseado na banda disponível." Universidade Tecnológica Federal do Paraná, 2012. http://repositorio.utfpr.edu.br/jspui/handle/1/304.
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O controle de congestionamento padrão do TCP apresenta vários problemas; ele não consegue distinguir se o pacote foi perdido por falha no enlace ou por descarte de pacotes devido a um congestionamento de rede (se a falha foi no enlace não há necessidade de ativar os mecanismos de controle de congestionamento); e o correto ajuste de sua taxa de transmissão requer informação de perdas de pacotes. Neste trabalho é apresentado o new generalized window advertising (NGWA), que é um novo esquema de controle de congestionamento para o TCP. O NGWA traz informações da banda disponível da infraestrutura de rede para os pontos finais da conexão TCP. Seu desempenho foi comparado com TCP New Reno, RED e o TCP padrão via simulações com o software NS-3, considerando topologias de rede largamente citadas na literatura. O NGWA foi, também, implementado e testado no Linux (versão 2.6.34). O novo método demonstrou ser superior aos comparados, apresentando uma operação mais estável, melhor justiça e menor taxa de perda de pacotes, considerando o elenco de testes realizados.The TCP congestion control mechanism in standard implementations presents several problems; he cannot distinguish if the packet was lost by link failure or by congestion in the net (if the fault was in the link there is no need to active congestion control mechanisms); and the right adjust of your transmission rate requires information from packet loss. This work presents the new generalized window advertising (NGWA), which is a new congestion control scheme for TCP. The NGWA provides information considering the available bandwidth of the network infrastructure to the endpoints of the TCP connection. Results obtained by the NGWA approach were compared with those from TCP New Reno, RED, and standard TCP (using the network simulator NS-3), considering network topologies widely cited in the literature. A NGWA Linux implementation is also presented. The new method proved to be superior when compared with the traditional approaches, presenting a more stable operation, better fairness and lower packet loss, considering the set of tests carried out.
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Sazima, Ricardo. "PLEX MPLS : analise, projeto e implementação de uma plataforma para experimentos com MPLS com suporte a QoS." [s.n.], 2004. http://repositorio.unicamp.br/jspui/handle/REPOSIP/259128.
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Orientador: Mauricio Ferreira MagalhãesDissertação (mestrado) - Universidade Estadual de Campinas, Faculdade de Engenharia Eletrica e de Computação
Made available in DSpace on 2018-08-04T17:32:51Z (GMT). No. of bitstreams: 1 Sazima_Ricardo_M.pdf: 1107563 bytes, checksum: a1179beaec699c0dcd87b45284e4a99c (MD5) Previous issue date: 2004
Resumo: Dados a banda de transmissão atualmente disponível, o protocolo (IP) utilizado para transmitir a maior parte de tráfego e a quantidade de tráfego e seus requisitos de aos, a Engenharia de Tráfego (TE, de Traffic Engineering) tomou-se um componente cada vez mais importante nas redes de comunicações. O padrão do IETF para enncaminhamento roteamento entitulado Multi-Protocol Label Switching (MPLS) preenche lacunas importantes neste cenário e é peça chave das metodologias de TE mais sofisticadas. Nesta dissertação é apresentado um modelo genérico de TE e uma proposta para uma Plataforma para Experimentos com MPLS (PLEX MPLS) que permite que o usuário defina, execute, e analise configurações de Engenharia de Tráfego com MPLS em uma rede IP. Os principais objetivos da PLEX MPLS são: . Estudo da tecnologia MPLS e suporte a outros trabalhos em andamento no contexto do grupo de estudos em MPLS do DCA . Experimentos com engenharia de tráfego com MPLS: validação das tecnologias e conceitos relacionados . Experimentos didáticos em disciplinas de laboratório de redes. Os principais conceitos de TE e MPLS são apresentados como referência, bem como uma descrição do NIST Switch, a plataforma MPLS escolhida para este trabalho. A análise, projeto e implementação da PLEX MPLS são apresentados, pois formam parte significativa do trabalho desenvolvido. A PLEX não só utiliza, como também estende as funcionalidades oferecidas pelo NIST Switch a fim de oferecer um esquema de TE mais completo e eficiente. Na fase de análise, vemos quais os principais requisitos para a implementação desta plataforma, seguindo uma metodologia de Engenharia de Software. Na fase de projeto, vemos as soluções propostas para os problemas identificados na fase de análise e temos uma especificação dos componentes a serem implementados. Na fase seguinte, discutimos a implementação das principais características dos componentes da PLEX, justificando as decisões tomadas. Para validar a implementação da PLEX de acordo com sua proposta, foram realizados alguns experimentos em uma rede de testes com tráfego real. A execução destes experimentos é descrita e seus resultados analisados. Os resultados obtidos assinalam claramente a importância e utilidade de esquemas de TE baseada em MPLS. Uma interessante metodologia para TE, compilada a partir de várias propostas, é apresentada. Finalmente, apontam-se caminhos a seguir em um trabalho futuro de refinamento da PLEX
Abstract: Given the bandwidth currently available, the protocol (IP) used to transmit most Internet traffic, the quantity of traffic produced and its QoS requirements, Traffic Engineering (TE) has become an increasingly important component of communications networks. IETF's standard for forwarding/routing, which is entitled Multi-Protocol Label Switching (MPLS), presents important solutions in this scenario playing a major role in more sophisticated TE methodologies. This work presents a generic methodology for TE and a proposal for a Platform for Experiments with MPLS (PLEX MPLS) which allows the user to define, execute and analyze Traffic Engineering configurations with MPLS in an IP network. The main goals of PLEX MPLS are: . Study of the M PLS technology and support of other ongoing works with M PLS in the DCA . Traffic Engineering experiments with MPLS: validation of the related concepts and technologies . Support of didactic experiments in academic disciplines. The main concepts of MPLS and TE are presented as reference, as well as a brief description of NIST Switch, the MPLS software chosen for the PLEX MPLS implementation. The analysis, project and implementation of PLEX MPLS are presented, since are significant part of the developed work. PLEX not only uses, but also extends NIST Switch functionalities to offer a more complete and efficient TE scheme. In the analysis phase the main requirements for the PLEX implementation are specified, following a well-known Software Engineering methodology. The solutions found for the problems identified in the analysis phase are presented in the project phase alongside with a specification of the components that will be implemented. In the next phase, the implementation of PLEX is discussed focusing on the most important characteristics of PLEX components and justifying the implementation ecisions. In order to validate PLEX implementation and its proposal, some experiments were made in a test network with reallive traffic. These experiments are described and its results analyzed. The results obtained clearly indicate the importance and utility of TE schemes based on MPLS. Also an interesting TE methodology compiled from several proposals is presented. Finally, possible improvements and future work on PLEX MPLS are indicated.
Mestrado
Engenharia de Computação
Mestre em Engenharia Elétrica
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Krishnan, Anupama. "The Multipath Fault-Tolerant Protocol for Routing in Packet-Switched Communication Network." Thesis, University of North Texas, 2003. https://digital.library.unt.edu/ark:/67531/metadc4199/.
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In order to provide improved service quality to applications, networks need to address the need for reliability of data delivery. Reliability can be improved by incorporating fault tolerance into network routing, wherein a set of multiple routes are used for routing between a given source and destination. This thesis proposes a new fault-tolerant protocol, called the Multipath Fault Tolerant Protocol for Routing (MFTPR), to improve the reliability of network routing services. The protocol is based on a multipath discovery algorithm, the Quasi-Shortest Multipath (QSMP), and is designed to work in conjunction with the routing protocol employed by the network. MFTPR improves upon the QSMP algorithm by finding more routes than QSMP, and also provides for maintenance of these routes in the event of failure of network components. In order to evaluate the resilience of a pair of paths to failure, this thesis proposes metrics that evaluate the non-disjointness of a pair of paths and measure the probability of simultaneous failure of these paths. The performance of MFTPR to find alternate routes based on these metrics is analyzed through simulation.
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Suryadevara, Usha. "Buffer management in tone allocated multiple access protocol." [Gainesville, Fla.] : University of Florida, 2001. http://purl.fcla.edu/fcla/etd/UFE0000357.
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Thesis (M.S.)--University of Florida, 2001.Title from title page of source document. Document formatted into pages; contains vi, 78 p.; also contains graphics. Includes vita. Includes bibliographical references.
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Li, Xiaoming Biaz Saad. "Performance evaluation of biased queue management." Auburn, Ala., 2006. http://repo.lib.auburn.edu/2006%20Summer/Theses/LI_XIAOMING_37.pdf.
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Eddy, Wesley M. "Improving Transport Control Protocol Performance With Path Error Rate Information." Ohio University / OhioLINK, 2004. http://www.ohiolink.edu/etd/view.cgi?ohiou1087844627.
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Grigorescu, Eduard. "Reducing internet latency for thin-stream applications over reliable transport with active queue management." Thesis, University of Aberdeen, 2018. http://digitool.abdn.ac.uk:80/webclient/DeliveryManager?pid=236098.
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An increasing number of network applications use reliable transport protocols. Applications with constant data transmission recover from loss without major performance disruption, however, applications that send data sporadically, in small packets, also called thin-streams, experience frequently high latencies due to 'Bufferbloat', that reduce the application performance. Active Queue Management mechanisms were proposed to dynamically manage the queues in routers by dropping packets early and reduce these, hence reducing latency. While their deployment to the internet remains an open issue, the proper investigation into how their functioning mechanism impacts latency is the main focus of this work and research questions have been devised to investigate the AQM impact on latency. A range of AQM mechanisms has been evaluated by the research, exploring performance of the methods for latency sensitive network applications. This has explored new single queue AQM mechanisms such as Controlled Delay (CODEL) and Proportional Integral Enhanced (PIE) and Adaptive RED (ARED). The evaluation has shown great improvements in queuing latency when AQM are used over a range of network scenarios. Scheduling AQM algorithms such as FlowQueue CODEL (FQ-CODEL) isolates traffic and minimises the impact of Bufferbloat on flows. The core components of FQ-CODEL, still widely misunderstood at the time of its inception, have been explained in depth by this study and their contribution to reducing latency have been evaluated. The results show significant reductions in queuing latency for thin streams using FQ-CODEL. When TCP is used for thin streams, high application latencies can arise when there are retransmissions, for example after dropping packets by an AQM mechanism. This delay is a result of TCP's loss-based congestion control mechanism that controls sender transmission rate following packet loss. ECN, a marking sender-side improvement to TCP reduces applicationlayer latency without disrupting the overall network performance. The thesis evaluated the benefit of using ECN using a wide range of experiments. The findings show that FQ-CODEL with ECN provides a substantial reduction of application latency compared to a drop-based AQM. Moreover, this study recommends the combination of FQ-CODEL with other mechanisms, to reduce application latency. Mechanisms such as ABE, have been shown to increase aggregate throughput and reduce application latency for thin-stream applications.
Books on the topic "Computer networks Packet switching (Data transmission) TCP/IP (Computer network protocol)"
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1968-, Rovithakis George A., ed. End to end adaptive congestion control in TCP/IP networks. Boca Raton, FL: Taylor & Francis, 2012.
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Building switched networks: Multilayer switching, QoS, IP multicast, network policy, and service level agreements. Reading, Mass: Addison-Wesley, 1999.
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Emma, Minoli, ed. Delivering voice over IP networks. 2nd ed. New York: Wiley, 2002.
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Minoli, Daniel. Delivering Voice over IP Networks. New York: John Wiley & Sons, Ltd., 2002.
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The simple book: An introduction to management of TCP/IP-based internets. Englewood Cliffs, N.J: Prentice Hall, 1991.
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Rose, Marshall T. The Simple book: An introduction to management of TCP/IP based internets. Englewood Cliffs, N.J: Prentice Hall International, 1991.
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Comer, Douglas. Internetworking with TCP/IP. 2nd ed. Upper Saddle River, N.J: Prentice Hall, 1996.
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Comer, Douglas. Internetworking with TCP/IP. Englewood Cliffs, N.J: Prentice Hall, 1993.
Find full textBook chapters on the topic "Computer networks Packet switching (Data transmission) TCP/IP (Computer network protocol)"
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Clayton, John F. "Education, the Internet, and the World Wide Web." In Encyclopedia of Human Computer Interaction, 175–78. IGI Global, 2006. http://dx.doi.org/10.4018/978-1-59140-562-7.ch028.
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The development of the Internet has a relatively brief and well-documented history (Cerf, 2001; Griffiths, 2001; Leiner et al., 2000; Tyson, 2002). The initial concept was first mooted in the early 1960s. American computer specialists visualized the creation of a globally interconnected set of computers through which everyone quickly could access data and programs from any node, or place, in the world. In the early 1970s, a research project initiated by the United States Department of Defense investigated techniques and technologies to interlink packet networks of various kinds. This was called the Internetting project, and the system of connected networks that emerged from the project was known as the Internet. The initial networks created were purpose-built (i.e., they were intended for and largely restricted to closed specialist communities of research scholars). However, other scholars, other government departments, and the commercial sector realized the system of protocols developed during this research (Transmission Control Protocol [TCP] and Internet Protocol [IP], collectively known as the TCP/IP Protocol Suite) had the potential to revolutionize data and program sharing in all parts of the community. A flurry of activity, beginning with the National Science Foundation (NSF) network NSFNET in 1986, over the last two decades of the 20th century created the Internet as we know it today. In essence, the Internet is a collection of computers joined together with cables and connectors following standard communication protocols.
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Rajagopal, Dr. "Internet, Reengineering and Technology Applications in Retailing." In Business Information Systems, 1324–42. IGI Global, 2010. http://dx.doi.org/10.4018/978-1-61520-969-9.ch082.
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The rapid growth in computer technology and commercial needs have allowed significant changes in the information management systems. There emerged in early nineties many commercial network backbones to link with the NSFnet to provide market information to the business firms. The Internet today is a combination of NSFnet and commercially available backbone services disseminating information on the decentralized networks all over the world. It is estimated that there are over 30,000 computer networks connecting over 2 million computers with each other on the Web. In view of the increasing use of electronic information sources through the networks the Transmission Control Protocol/Internet Protocol (TCP/IP) has been designed and made essential for each user networks to abide with the protocol standards which enables the data transfer and retrieval at source.
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Louvros, Spiros. "Towards Unified Services in Heterogeneous Wireless Networks Based on Soft-Switch Platform." In Encyclopedia of Multimedia Technology and Networking, Second Edition, 1416–22. IGI Global, 2009. http://dx.doi.org/10.4018/978-1-60566-014-1.ch191.
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The last two decades, after the telecommunication and computer technology convergence, the world of telecommunication applications has changed dramatically. The traffic needs of the customers have moved from circuit switched applications towards packet switched applications (Cox, 1995). Data traffic, with the characteristics of information transmission in the form of packets and the bursty flow characteristics rather than constant rate, nowadays accounts for slightly more than 60% of the traffic that is transmitted over the backbone telecommunication networks (Esmailzadeh, Nakagawa, & Jones, 2003). In addition to data traffic, multimedia applications like video calls, IP TV, and multimedia messaging traffic (variable rate with real time constraints) was made possible by low cost video digitizing equipment (Houssos, Alonistioti, Merakos, Mohyeldin, Dillinger, Fahrmair, & Schoenmakers, 2003). Different Radio Access Technology (RAT) networks offer different services to their subscribers. This is a big problem for the multimedia industry since it poses certain constraints to the subscribers regarding specific technology handsets. The ideal solution might be a unified handset with a unified service subscriber identity module (SIM) card (Louvros & Iossifides, 2004). This handset should be able to access the service by any radio access network, like Global System Mobile (GSM) (Siegmund, Redl, Weber, & Oliphant, 1995), General Packet Radio System (GPRS), Universal Mobile Telecommunications System (UMTS), and IEEE802.11 standard (WiFi or WLAN) towards a common core platform. In order to achieve such a unification, the service request should be seamless to the radio access technology network and the core platform should support certain protocols to provide again seamless to the user access to the requested service. Such a platform is already designed and is known as the soft-switch solution. The idea behind the soft-switch solution is the layering of the core network management procedures (mobility management, call control, session management, charging) in such a way that the operator can support all requests as a unified routing process. Moreover the operator can deploy its core switch and transmission network based on a common backbone, designed according to the 3GPP standards on IP or ATM infrastructure, and also to be able to accommodate in the future any new radio access technology network simply and without any serious rearrangement of the existing backbone, thus eliminating cost implementation. Asynchronous Transfer Mode (ATM) technology is proposed by the telecommunication industry to accommodate multiple traffic types (packet and voice) in a high speed wire-line backbone network. Briefly, ATM is based on very fast (on the order of 2.5 Gbits/sec or higher (Q.2931 ATM Network Signaling Specification, ITU)) packet switching technology with 53 byte long packets called cells being transmitted through wireline networks running usually on fiber optical equipment (Louvros, Karaboulas, Iossifides, & Kotsopoulos, 2003).