978 resultados para linux kernel network tcp ip
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The publication comments on certain moments of the method of teaching the types of addresses and their use in the TCP/IP protocol stack.
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Se realizar una investigación para formar una base teórica que permita desarrollar un manual de referencia que sirva como consulta general sobre el uso del Protocolo de Internet versión 6, IPv6, y de las especificaciones básicas necesarias para migrar a esta nueva tecnología. Verificar el uso que hacen las empresas o instituciones en nuestro medio de aplicaciones operando en redes basadas en el Protocolo de Internet versión 4, con la idea de desarrollar pautas que sirvan para definir una metodología general para la migración de estas redes y sus aplicaciones en operación, al nuevo Protocolo de Internet IPv6
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Treball de final de carrera que descriu el procés d'implementació de TaFanet v1.0.
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Mestrado em Engenharia Electrotécnica e de Computadores
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En aquest treball s'ha desenvolupat una aplicació capaç de localitzar adreces IP.
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TCP flows from applications such as the web or ftp are well supported by a Guaranteed Minimum Throughput Service (GMTS), which provides a minimum network throughput to the flow and, if possible, an extra throughput. We propose a scheme for a GMTS using Admission Control (AC) that is able to provide different minimum throughput to different users and that is suitable for "standard" TCP flows. Moreover, we consider a multidomain scenario where the scheme is used in one of the domains, and we propose some mechanisms for the interconnection with neighbor domains. The whole scheme uses a small set of packet classes in a core-stateless network where each class has a different discarding priority in queues assigned to it. The AC method involves only edge nodes and uses a special probing packet flow (marked as the highest discarding priority class) that is sent continuously from ingress to egress through a path. The available throughput in the path is obtained at the egress using measurements of flow aggregates, and then it is sent back to the ingress. At the ingress each flow is detected using an implicit way and then it is admission controlled. If it is accepted, it receives the GMTS and its packets are marked as the lowest discarding priority classes; otherwise, it receives a best-effort service. The scheme is evaluated through simulation in a simple "bottleneck" topology using different traffic loads consisting of "standard" TCP flows that carry files of varying sizes
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Memòria del projecte final de carrera que mostra un mapamundi on es representa el recorregut que realitzen els paquets de dades per arribar al seu destí.
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VDSL on teknologia, joka mahdollistaa nopeat Internet-yhteydet tavallista puhelinlinjaa käyttäen. Tätä varten käyttäjä tarvitsee VDSL-modeemin ja Internet-operaattori reitittimen, johon VDSL-linjat kytketään. Reitittimen on oltava suorituskykyinen, jotta kaikki VDSL-liikenne voidaan reittittää eteenpäin. Tehokkuutta haetaan tekemällä suuri osa reitityksestä erityisillä reititinpiireillä. Tässä diplomityössä käsitellään reititinpiirien teoriaa ja niiden hallintaa. Lisäksi vertailtiin kolmen suuren valmistajan tuotteita. Tuotteiden tarjoamat ominaisuudet vaikuttivat hyvin yhteneväisiltä. Ominaisuuksien hallinta ja toteutus olivat erilaisia. Työn tavoitteena oli löytää ohjelmistoarkkitehtuuri piirien ohjaamiseen niin, että Linux-käyttöjärjestelmän ytimen palveluja voitaisiin käyttää mahdollisimman hyödyllisesti. Työssä havaittiin, että ohjelmistoarkkitehtuurin voi määritellä monella eri tavalla riippuen siitä, miten piiri on kytketty prosessoriin, mitä piirin ominaisuuksia halutaan käyttää ja miten arkkitehtuuria halutaan jatkossa laajentaa.
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Tässä diplomityössä selvitetään SNMP:n (Simple Network Management Protocol) hallintatietokantojen sisältämiä palvelunlaatutietoja. Tarkastelun kohteena ovat IP:n (Internet Protocol) ja ATM:n (Asynchronous Transfer Mode) hallintatietokannat. Työn teoriaosassa tarkastellaan verkonhallinnan eri osa-alueita, TCP/IP-verkkojen (Transmission Control Protocol) hallintaan tarkoitettua SNMP-protokollaa ja sen eri versioita. Lisäksi käsitellään palvelunlaadun kannalta IP ja ATM-verkkojen erilaisia toteutuksia. Työn kokeellissa osassa arvioidaan eri hallintatietokantojen sisältöä palvelunlaadun kannalta. Työssä todetaan palvelunlaatutietojen puutteellisuus hallintatietokannoissa ja sekä IP:lle että ATM:lle toteutetaan soveltuvat hallintatietokannat.
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Neljännen sukupolven mobiiliverkot kokoaa kaikki tietoliikenneverkot ja palvelut Internetin ympärille. Tämä mullistus muuttaa vanhat vertikaaliset tietoliikenneverkot joissa yhden tietoliikenneverkon palvelut ovat saatavissa vain kyseisen verkon päätelaitteille horisontaaliseksi malliksi jossa päätelaitteet käyttävät omaa verkkoansa pääsynä Internetin palveluihin. Tämä diplomityö esittelee idean paikallisista palveluista neljännen sukupolven mobiiliverkossa. Neljännen sukupolven mobiiliverkko yhdistää perinteiset televerkkojen palvelut ja Internet palvelut sekä mahdollistaa uuden tyyppisten palveluiden luonnin. TCP/IP protokollien ja Internetin evoluutio on esitelty. Laajakaistaiset, lyhyen kantaman radiotekniikat joita käytetään langattomana yhteytenä Internetiin on käsitelty. Evoluutio kohti neljännen sukupolven mobiiliverkkoja on kuvattu esittelemällä vanhat, nykyiset ja tulevat mobiiliverkot sekä niiden palvelut. Ennustukset palveluiden ja markkinoiden tulevaisuuden kehityksestä on käsitelty. Neljännen sukupolven mobiiliverkon arkkitehtuuri mahdollistaa paikalliset palvelut jotka ovat saatavilla vain yhdessä paikallisessa 4G verkossa. Paikalliset palvelut voidaan muunnella jokaiselle käyttäjälle erikseen käyttäen profiili-informaatiota ja paikkatietoa. Työssä on pohdittu paikallisten palveluiden käyttökelpoisuutta ja mahdollisuuksia käyttäen Lappeenrannan teknillisen korkeakoulun 4G projektin palvelupilotin tuloksia.
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TCP flows from applications such as the web or ftp are well supported by a Guaranteed Minimum Throughput Service (GMTS), which provides a minimum network throughput to the flow and, if possible, an extra throughput. We propose a scheme for a GMTS using Admission Control (AC) that is able to provide different minimum throughput to different users and that is suitable for "standard" TCP flows. Moreover, we consider a multidomain scenario where the scheme is used in one of the domains, and we propose some mechanisms for the interconnection with neighbor domains. The whole scheme uses a small set of packet classes in a core-stateless network where each class has a different discarding priority in queues assigned to it. The AC method involves only edge nodes and uses a special probing packet flow (marked as the highest discarding priority class) that is sent continuously from ingress to egress through a path. The available throughput in the path is obtained at the egress using measurements of flow aggregates, and then it is sent back to the ingress. At the ingress each flow is detected using an implicit way and then it is admission controlled. If it is accepted, it receives the GMTS and its packets are marked as the lowest discarding priority classes; otherwise, it receives a best-effort service. The scheme is evaluated through simulation in a simple "bottleneck" topology using different traffic loads consisting of "standard" TCP flows that carry files of varying sizes
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T'his dissertation proposes alternative models to allow the interconnectioin of the data communication networks of COSERN Companhia Energética do Rio Grande do Norte. These networks comprise the oorporative data network, based on TCP/IP architecture, and the automation system linking remote electric energy distribution substations to the main Operatin Centre, based on digital radio links and using the IEC 60870-5-101 protoco1s. The envisaged interconnection aims to provide automation data originated from substations with a contingent route to the Operation Center, in moments of failure or maintenance of the digital radio links. Among the presented models, the one chosen for development consists of a computational prototype based on a standard personal computer, working under LINUX operational system and running na application, developesd in C language, wich functions as a Gateway between the protocols of the TCP/IP stack and the IEC 60870-5-101 suite. So, it is described this model analysis, implementation and tests of functionality and performance. During the test phase it was basically verified the delay introduced by the TCP/IP network when transporting automation data, in order to guarantee that it was cionsistent with the time periods present on the automation network. Besides , additional modules are suggested to the prototype, in order to handle other issues such as security and prioriz\ation of the automation system data, whenever they are travesing the TCP/IP network. Finally, a study hás been done aiming to integrate, in more complete way, the two considered networks. It uses IP platform as a solution of convergence to the communication subsystem of na unified network, as the most recente market tendencies for supervisory and other automation systems indicate
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Since the appearance of downsized and simplified TCP/IP stacks, single nodes from Wireless Sensor Networks (WSNs) have become directly accessible from the Internet with commonly used networking tools and applications (e.g., Telnet or SMTP). However, TCP has been shown to perform poorly in wireless networks, especially across multiple wireless hops. This paper examines TCP performance optimizations based on distributed caching and local retransmission strategies of intermediate nodes in a TCP connection, and proposes extended techniques to these strategies. The paper studies the impact of different radio duty-cycling MAC protocols on the end-to-end TCP performance when using the proposed TCP optimization strategies in an extensive experimental evaluation on a real-world sensor network testbed.
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To interconnect a wireless sensor network (WSN) to the Internet, we propose to use TCP/IP as the standard protocol for all network entities. We present a cross layer designed communication architecture, which contains a MAC protocol, IP, a new protocol called Hop-to-Hop Reliability (H2HR) protocol, and the TCP Support for Sensor Nodes (TSS) protocol. The MAC protocol implements the MAC layer of beacon-less personal area networks (PANs) as defined in IEEE 802.15.4. H2HR implements hop-to-hop reliability mechanisms. Two acknowledgment mechanisms, explicit and implicit ACK are supported. TSS optimizes using TCP in WSNs by implementing local retransmission of TCP data packets, local TCP ACK regeneration, aggressive TCP ACK recovery, congestion and flow control algorithms. We show that H2HR increases the performance of UDP, TCP, and RMST in WSNs significantly. The throughput is increased and the packet loss ratio is decreased. As a result, WSNs can be operated and managed using TCP/IP.
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Nowadays, we can send audio on the Internet for multiples uses like telephony, broadcast audio or teleconferencing. The issue comes when you need to synchronize the sound from different sources because the network where we are going to work could lose packets and introduce delay in the delivery. This can also come because the sound cards could be work in different speeds. In this project, we will work with two computers emitting sound (one will simulate the left channel (mono) of a stereo signal, and the other the right channel) and connected with a third computer by a TCP network. The last computer must get the sound from both computers and reproduce it in a speaker properly (without delay). So, basically, the main goal of the project is to synchronize multi-track sound over a network. TCP networks introduce latency into data transfers. Streaming audio suffers from two problems: a delay and an offset between the channels. This project explores the causes of latency, investigates the affect of the inter-channel offset and proposes a solution to synchronize the received channels. In conclusion, a good synchronization of the sound is required in a time when several audio applications are being developed. When two devices are ready to send audio over a network, this multi-track sound will arrive at the third computer with an offset giving a negative effect to the listener. This project has dealt with this offset achieving a good synchronization of the multitrack sound getting a good effect on the listener. This was achieved thanks to the division of the project into several steps having constantly a good vision of the problem, a good scalability and having controlled the latency at all times. As we can see in the chapter 4 of the project, a lack of synchronization over c. 100μs is audible to the listener. RESUMEN. A día de hoy, podemos transmitir audio a través de Internet por varios motivos como pueden ser: una llamada telefónica, una emisión de audio o una teleconferencia. El problema viene cuando necesitas sincronizar ese sonido producido por los diferentes orígenes ya que la red a la que nos vamos a conectar puede perder los paquetes y/o introducir un retardo en las entregas de los mismos. Así mismo, estos retardos también pueden venir producidos por las diferentes velocidades a las que trabajan las tarjetas de sonido de cada dispositivo. En este proyecto, se ha trabajado con dos ordenadores emitiendo sonido de manera intermitente (uno se encargará de simular el canal izquierdo (mono) de la señal estéreo emitida, y el otro del canal derecho), estando conectados a través de una red TCP a un tercer ordenador, el cual debe recibir el sonido y reproducirlo en unos altavoces adecuadamente y sin retardo (deberá juntar los dos canales y reproducirlo como si de estéreo de tratara). Así, el objetivo principal de este proyecto es el de encontrar la manera de sincronizar el sonido producido por los dos ordenadores y escuchar el conjunto en unos altavoces finales. Las redes TCP introducen latencia en la transferencia de datos. El streaming de audio emitido a través de una red de este tipo puede sufrir dos grandes contratiempos: retardo y offset, los dos existentes en las comunicaciones entre ambos canales. Este proyecto se centra en las causas de ese retardo, investiga el efecto que provoca el offset entre ambos canales y propone una solución para sincronizar los canales en el dispositivo receptor. Para terminar, una buena sincronización del sonido es requerida en una época donde las aplicaciones de audio se están desarrollando continuamente. Cuando los dos dispositivos estén preparados para enviar audio a través de la red, la señal de sonido multi-canal llegará al tercer ordenador con un offset añadido, por lo que resultará en una mala experiencia en la escucha final. En este proyecto se ha tenido que lidiar con ese offset mencionado anteriormente y se ha conseguido una buena sincronización del sonido multi-canal obteniendo un buen efecto en la escucha final. Esto ha sido posible gracias a una división del proyecto en diversas etapas que proporcionaban la facilidad de poder solucionar los errores en cada paso dando una importante visión del problema y teniendo controlada la latencia en todo momento. Como se puede ver en el capítulo 4 del proyecto, la falta de sincronización sobre una diferencia de 100μs entre dos canales (offset) empieza a ser audible en la escucha final.