• Title/Summary/Keyword: packet size

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Design of ATM Switch-based on a Priority Control Algorithm (우선순위 알고리즘을 적용한 상호연결 망 구조의 ATM 스위치 설계)

  • Cho Tae-Kyung;Cho Dong-Uook;Park Byoung-Soo
    • The Journal of the Korea Contents Association
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    • v.4 no.4
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    • pp.189-196
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    • 2004
  • Most of the recent researches for ATM switches have been based on multistage interconnection network known as regularity and self-routing property. These networks can switch packets simultaneously and in parallel. However, they are blocking networks in the sense that packet is capable of collision with each other Mainly Banyan network have been used for structure. There are several ways to reduce the blocking or to increase the throughput of banyan-type switches: increasing the internal link speeds, placing buffers in each switching node, using multiple path, distributing the load evenly in front of the banyan network and so on. Therefore, this paper proposes the use of recirculating shuffle-exchange network to reduce the blocking and to improve hardware complexity. This structures are recirculating shuffle-exchange network as simplified in hardware complexity and Rank network with tree structure which send only a packet with highest priority to the next network, and recirculate the others to the previous network. after it decides priority number on the Packets transferred to the same destination, The transferred Packets into banyan network use the function of self routing through decomposition and composition algorithm and all they arrive at final destinations. To analyze throughput, waiting time and packet loss ratio according to the size of buffer, the probabilities are modeled by a binomial distribution of packet arrival. If it is 50 percentage of load, the size of buffer is more than 15. It means the acceptable packet loss ratio. Therefore, this paper simplify the hardware complexity as use of recirculating shuffle-exchange network instead of bitonic sorter.

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Development of PLC by using micro controller for the distributed fire alarm system (마이크로 콘트롤러를 이용한 분산형 방재 시스템용 중계기 개발)

  • Han, Kyong-Ho;Lee, Ki-Sik;Hwang, Seok-Yong;Kim, Jong-Chol
    • Proceedings of the KIEE Conference
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    • 1996.07b
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    • pp.1243-1247
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    • 1996
  • In this paper, a small PLC system is designed and developed for the distributed building control and fire control system. It uses 8031, a 8-bit micro controller from Intel Inc. The size of the PLC is $7cm{\times}8cm$ and the PLC can accept 4 sensor inputs and drive 4 relay outputs, which operates at 24V. Multiple access of PLC by one host computer is implemented by assigning a unique 10 to each PLC, which ranges from 0 to 126. The operation starts by sending a command packet from host computer to a PLC and the PLC of the same 10 fetches the command packet by comparing the first byte of the command packet with its own 10. The PLC is programmed to perform a various functions and the function is selected by the content of the command byte, which is the second byte of the command packet. The third byte, which is the last byte, is a checksum byte. The checksum byte is the sum of the first byte and the second byte and is used to detect the communication error. Depending on the content of the command byte, PLC performs the desired function and returns the response packet back to the host computer. The response packet is also a three-byte packet, 10 byte, response byte and checksum byte. For the independent operation of PLC without being controlled by the host computer, variable length RULE data packet is sent to PLC. In case the communication line is broken, the PLC perform the independent operation by referencing the RULE data. The applicable areas are; building automation system, distributed factory automation, measurement of temperature of toxic or dangerous area.

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Image Denosing Based on Wavelet Packet with Absolute Average Threshold (절대평균임계값을 적용한 웨이블릿 패킷 기반의 영상 노이즈 제거)

  • Ryu, Kwang-Ryol;Sclabassi, Robert J.
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2007.06a
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    • pp.605-608
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    • 2007
  • The denoising for image restoration based on the Wavelet Packet with absolute average threshold is presented. The Existing method is used standard deviation estimated results in increasing the noise and threshold, and damaging an image quality. In addition that is decreased image restoration PSNR by using the same threshold in spite of changing image because of installing a threshold in proportion of noise size. In contrast, the absolute average threshold with wavelet packet is adapted by changing image to set up threshold by statistic quantity of resolved image and is avoided an extreme impart. The results on the experiment has improved 10% and 5% over than the denoising based on simple wavelet transform and wavelet packet respectively.

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Performance Analysis of Packet Switch Interconnection Networks with Output Buffer Modules (출구 버퍼모듈을 갖는 패킷 교환식 상호 연결 망의 성능 분석)

  • Chu, Hyeon-Seung;Park, Gyeong-Rin
    • The Transactions of the Korea Information Processing Society
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    • v.6 no.4
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    • pp.1045-1057
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    • 1999
  • Packet-switched multistage interconnection networks(MINs) have been widely used for digital switching systems and super computers. In this paper we show that multiple packets in a switching element can move to the succeeding switching element in one network cycle by fully utilizing the cycle bandwidth. Only one packet movement was usually assumed in typical MINs. we present an analytical model for the MNs with the multiple packet movement scheme, and validate it by computer simulation. Comparisons with the traditional MINs of single packet movement reveal that the throughput is increased up to about 30% for practical size MINs. Similar result was also obtained for delays. The performance increase is more significant when the network traffic is nonuniform.

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TCP Performance Enhancement over the Wireless Networks by Using CPC and ZWSC (CPC와 ZWSC를 이용한 무선 망에서의 TCP 성능 향상 방안)

  • Lee, Myung-Sub;Park, Young-Min;Chang, Joo-Seok;Park, Chang-Hyeon
    • IEMEK Journal of Embedded Systems and Applications
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    • v.1 no.1
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    • pp.24-30
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    • 2006
  • With the original Transmission Control Protocol(TCP) design, which is particularly targeted at the wired networks, a packet loss is assumed to be caused by the network congestion. In the wireless environment where the chances to lose packets due to transmission bit errors are not negligible, though, this assumption may result in unnecessary TCP performance degradation. In these days, many papers describe about wireless-TCP which has suggested how to avoid congestion control when packet loss over the wireless network. In this paper, an enhancement scheme is proposed by modifying SNOOP scheme. To enhance the original SNOOP scheme, CPC(Consecutive Packet Control) and ZWSC(Zero Window Size Control) are added. The invocation of congestion control mechanism is now minimized by knowing the cause of packet loss. We use simulation to compare the overhead and the performance of the proposed schemes, and to show that the proposed schemes improve the TCP performance compares to SNOOP by knowing the cause of packet loss at the base station.

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TCP Performance Improvement in Network Coding over Multipath Environments (다중경로 환경의 네트워크 코딩에서의 TCP 성능개선 방안)

  • Lim, Chan-Sook
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.11 no.6
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    • pp.81-86
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    • 2011
  • In one of the most impacting schemes proposed to address the TCP throughput problem over network coding, the network coding layer sends an acknowledgement if an innovative linear combination is received, even when a new packet is not decoded. Although this scheme is very effective, its implementation requires a limit on the coding window size. This limitation causes low TCP throughput in the presence of packet reordering. We argue that a TCP variant detecting a packet loss relying only on timers is effective in dealing with the packet reordering problem in network coding environments as well. Also we propose a new network coding layer to support such a TCP variant. Simulation results for a 2-path environment show that our proposed scheme improves TCP throughput by 19%.

A Study on the Application method of Server Router for Reliable Multicast (신뢰성 있는 멀티캐스트를 위한 서버라우터의 활용 방안에 관한 연구)

  • Choi, Won-Hyuck;Lee, Kwang-Jae;Kim, Jung-Sun
    • Proceedings of the Korea Information Processing Society Conference
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    • 2002.04b
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    • pp.1483-1486
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    • 2002
  • Multicast protocols are efficient methods of group communication, but they do not support the various transmission protocol services like a reliability guarantee, FTP, or Telnet that TCPs do. The purpose of this dissertation is to find a method to utilize sewer routers to form multicasts that can simultaneously transport multicast packets and TCP packets. For multicast network scalability and error recovery the existing SRM method has been used. Three packets per TCP transmission control window size are used for transport and an ACK is used for flow control. A CBR and a SRM is used for UDP traffic control. Divided on whether a UDP multicast packet and TCP unicast packet is used simultaneously or only a UDP multicast packet transport is used, the multicast receiver with the longest delay is measured on the number of packets and its data receiving rate. It can be seen that the UDP packet and the TCP's IP packet can be simultaneously used in a server router.

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A Study on Voice Quality and Speed Upgrade for Internet phone System (인터넷폰 시스템의 음질 및 속도향상연구)

  • 임종설;김성호;조남인;오춘석
    • Journal of the Korea Computer Industry Society
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    • v.3 no.5
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    • pp.631-640
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    • 2002
  • The internet phones that are currently available in use adopt packet exchange system, transferring through various routes and lacking sufficient band width with a result that there is an accompanied delay for packet transmission since the traffic is increased, accordingly affecting a lot in sound quality and speed. Two solutions for such troubles are suggested in this study to improve sound quality of internet phones. Firstly, we minimize the delay and damage regarding packet size based on traffic size by using the data algorithm from variable packets in order to supplement decreased sound quality due to the delay and damage of sound data. The second suggestion is to employ a method of Jitter compensation by giving an appropriate initial delay time with regenerating buffers to bypass troubles from Jitter, From employing the Jitter compensation method, we found that there is a sound quality improvement due to the less stoppage phenomenon.

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Performance Evaluation of the RSVP-capable Router using Latency-Optimized Fair Queuing Scheduler (최적 레이턴시 기반 공정 큐잉 스케줄러를 사용하는 RSVP-라우터의 성능 평가)

  • Kim, Tae-Joon
    • Journal of Korea Multimedia Society
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    • v.11 no.11
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    • pp.1536-1546
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    • 2008
  • RSVP-capable router supporting guaranteed services on the internet generally uses a packet scheduler based on the Weighted Fair Queuing (WFQ) algorithm to secure required qualities of traffic flows. A RSVP-capable router based on the LOFQ (Latency optimized fair queuing) algorithm had been proposed to improve the performance of the WFQ-based RSVP-router, but it required not only the RESV message to be expanded but also its performance could be evaluated only through simulation. This paper proposed a LOFQ based RSVP-capable router using the conventional RESV message and developed an algorithm to analyze the performance of the LOFQ based RSVP-capable routers. The performance evaluation using the developed algorithm showed that in terms of performance improvement the proposed router is inferior to the one using the expanded RESV message under a small packet size, but on the range of a large packet size both routers provide the same improvement.

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