995 resultados para DNP3 over TCP


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This paper presents simulation results of the DNP3 communication protocol over a TCP/IP network, for Smart Grid applications. The simulation was performed using the NS-2 network simulator. This study aimed to use the simulation to verify the performance of the DNP3 protocol in a heterogeneous LAN. Analyzing the results it was possible to verify that the DNP3 over a heterogeneous traffic network, with communication channel capacity between 60 and 85 percent, it works well with low packet loss and low delay, however, with traffic values upper 85 percent, the DNP3 usage becomes unfeasible because the information lost, re-transmissions and latency are significantly increased. © 2013 IEEE.

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In dealing with computer networks, these allow the flow of information through the resources of various equipment's. This work describes the implementation through the encapsulation of Protocol DNP3, usually employed in Smart Grid communication, in a simulator of discrete events. The NS-2 is a simulator in open source of network events, that facilitate the development of communication networks scenarios considering the protocols involved, in wireless or wired technologies. The objective of this work is to develop the DNP3 protocol encapsulation over a TCP/IP in the in the discrete event Simulator NS-2, allowing an analysis of behavior of a middle or large network sized in Smart Grid applications. © 2013 IEEE.

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This work presents a packet manipulation tool developed to realize tests in industrial devices that implements TCP/IP-based communication protocols. The tool was developed in Python programming language, as a Scapy extension. This tool, named IndPM- Industrial Packet Manipulator, can realize vulnerability tests in devices of industrial networks, industrial protocol compliance tests, receive server replies and utilize the Python interpreter to build tests. The Modbus/TCP protocol was implemented as proof-of-concept. The DNP3 over TCP protocol was also implemented but tests could not be realized because of the lack of resources. The IndPM results with Modbus/TCP protocol show some implementation faults in a Programmable Logic Controller communication module frequently utilized in automation companies

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There is a wide range of video services over complex transmission networks, and in some cases end users fail to receive an acceptable quality level. In this paper, the different factors that degrade users' quality of experience (QoE) in video streaming service that use TCP as transmission protocol are studied. In this specific service, impairment factors are: number of pauses, their duration and temporal location. In order to measure the effect that each temporal segment has in the overall video quality, subjective tests. Because current subjective test methodologies are not adequate to assess video streaming over TCP, some recommendations are provided here. At the application layer, a customized player is used to evaluate the behavior of player buffer, and consequently, the end user QoE. Video subjective test results demonstrate that there is a close correlation between application parameters and subjective scores. Based on this fact, a new metrics named VsQM is defined, which considers the importance of temporal location of pauses to assess the user QoE of video streaming service. A useful application scenario is also presented, in which the metrics proposed herein is used to improve video services(1).

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Coordenação de Aperfeiçoamento de Pessoal de Nível Superior (CAPES)

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The goal of the this paper is to show that the DGPS data Internet service we designed and developed provides campus-wide real time access to Differential GPS (DGPS) data and, thus, supports precise outdoor navigation. First we describe the developed distributed system in terms of architecture (a three tier client/server application), services provided (real time DGPS data transportation from remote DGPS sources and campus wide data dissemination) and transmission modes implemented (raw and frame mode over TCP and UDP). Then we present and discuss the results obtained and, finally, we draw some conclusions.

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Dissertation submitted in partial fulfillment of the requirements for the Degree of Master of Science in Geospatial Technologies.

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In this work we deal with video streams over TCP networks and propose an alternative measurement to the widely used and accepted peak signal to noise ratio (PSNR) due to the limitations of this metric in the presence of temporal errors. A test-bed was created to simulate buffer under-run in scalable video streams and the pauses produced as a result of the buffer under-run were inserted into the video before being employed as the subject of subjective testing. The pause intensity metric proposed in [1] was compared with the subjective results and it was shown that in spite of reductions in frame rate and resolution, a correlation with pause intensity still exists. Due to these conclusions, the metric may be employed in layer selection in scalable video streams. © 2011 IEEE.

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Faced with an imminent restructuring of the electric power system, over the past few years many countries have invested in a new paradigm known as Smart Grid. This paradigm targets optimization and automation of electric power network, using advanced information and communication technologies. Among the main communication protocols for Smart Grids we have the DNP3 protocol, which provides secure data transmission with moderate rates. The IEEE 802.15.4 is another communication protocol also widely used in Smart Grid, especially in the so-called Home Area Network (HAN). Thus, many applications of Smart Grid depends on the interaction of these two protocols. This paper proposes modeling, in the traditional network simulator NS-2, the integration of DNP3 protocol and the IEEE 802.15.4 wireless standard for low cost simulations of Smart Grid applications.

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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.

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Purpose: The purpose of this paper is to investigate the use of 802.11e MAC to resolve the transmission control protocol (TCP) unfairness. Design/methodology/approach: The paper shows how a TCP sender may adapt its transmission rate using the number of hops and the standard deviation of recently measured round-trip times to address the TCP unfairness. Findings: Simulation results show that the proposed techniques provide even throughput by providing TCP fairness as the number of hops increases over a wireless mesh network (WMN). Research limitations/implications: Future work will examine the performance of TCP over routing protocols, which use different routing metrics. Other future work is scalability over WMNs. Since scalability is a problem with communication in multi-hop, carrier sense multiple access (CSMA) will be compared with time division multiple access (TDMA) and a hybrid of TDMA and code division multiple access (CDMA) will be designed that works with TCP and other traffic. Finally, to further improve network performance and also increase network capacity of TCP for WMNs, the usage of multiple channels instead of only a single fixed channel will be exploited. Practical implications: By allowing the tuning of the 802.11e MAC parameters that have previously been constant in 802.11 MAC, the paper proposes the usage of 802.11e MAC on a per class basis by collecting the TCP ACK into a single class and a novel congestion control method for TCP over a WMN. The key feature of the proposed TCP algorithm is the detection of congestion by measuring the fluctuation of RTT of the TCP ACK samples via the standard deviation, plus the combined the 802.11e AIFS and CWmin allowing the TCP ACK to be prioritised which allows the TCP ACKs will match the volume of the TCP data packets. While 802.11e MAC provides flexibility and flow/congestion control mechanism, the challenge is to take advantage of these features in 802.11e MAC. Originality/value: With 802.11 MAC not having flexibility and flow/congestion control mechanisms implemented with TCP, these contribute to TCP unfairness with competing flows. © Emerald Group Publishing Limited.

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To exploit the popularity of TCP as still the dominant sender and protocol of choice for transporting data reliably across the heterogeneous Internet, this thesis explores end-to-end performance issues and behaviours of TCP senders when transferring data to wireless end-users. The theme throughout is on end-users located specifically within 802.11 WLANs at the edges of the Internet, a largely untapped area of work. To exploit the interests of researchers wanting to study the performance of TCP accurately over heterogeneous conditions, this thesis proposes a flexible wired-to-wireless experimental testbed that better reflects conditions in the real-world. To exploit the transparent functionalities between TCP in the wired domain and the IEEE 802.11 WLAN protocols, this thesis proposes a more accurate methodology for gauging the transmission and error characteristics of real-world 802.11 WLANs. It also aims to correlate any findings with the functionality of fixed TCP senders. To exploit the popularity of Linux as a popular operating system for many of the Internet’s data servers, this thesis studies and evaluates various sender-side TCP congestion control implementations within the recent Linux v2.6. A selection of the implementations are put under systematic testing using real-world wired-to-wireless conditions in order to screen and present a viable candidate/s for further development and usage in the modern-day heterogeneous Internet. Overall, this thesis comprises a set of systematic evaluations of TCP senders over 802.11 WLANs, incorporating measurements in the form of simulations, emulations, and through the use of a real-world-like experimental testbed. The goal of the work is to ensure that all aspects concerned are comprehensively investigated in order to establish rules that can help to decide under which circumstances the deployment of TCP is optimal i.e. a set of paradigms for advancing the state-of-the-art in data transport across the Internet.

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In this paper we describe a real-time industrial communication network able to support both controlrelated and multimedia traffic. The industrial communication network is based on the PROFIBUS standard, with multimedia capabilities being provided by an adequate integration of TCP/IP protocols into the PROFIBUS stack. From the operational point of view the integration of TCP/IP into PROFIBUS is by itself a challenge, since the master-slave nature of the PROFIBUS MAC makes complex the implementation of the symmetry inherent to IP communications. From the timeliness point of view the challenge is two folded. On one hand the multimedia traffic should not interfere with the timing requirements of the "native" control-related PROFIBUS traffic (typically hard real-time). On the other hand multimedia traffic requires certain levels of quality-of-service to be attained. In this paper we provide a methodology that enables fulfilling the timing requirements for both types of traffic in these real-time industrial LAN. Moreover, we describe suitable algorithms for the scheduling support of concurrent multimedia streams.