• Title/Summary/Keyword: Long RTT

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Analysis of internet self-similar traffic through the round-trip time measurement (Round-Trip Time 측정을 통한 인터넷 트래픽의 자기 유사성 분석)

  • 황인수;송기평;이동철;박기식;김창호;김동일;최삼길
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 1999.11a
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    • pp.326-330
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    • 1999
  • 인터넷의 민감한 트래픽 특성으로 인해 기존의 트래픽 모델링 분석법으로 최적화된 네트워크 환경을 구성하기에는 부족한 점이 많다. 본 논문에서는 트래픽의 버스트 특성을 정확히 예측하고 모델링 하기 위한 방법으로 자기 유사 특성에 대해 분석하고자 한다. 실제 인터넷 네트워크에서의 RTT(Round-Trip Time)를 측정함으로써 계층, 거리별 링크간의 LRD(Long-Range Dependence) 와 노드 큐의 특성, 자기유사성 정도를 구하고 측정된 데이터의 확률 분포를 통해 실제 트래픽의 특성에 대해 분석하였다

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Enhanced TFRC for High Quality Video Streaming over High Bandwidth Delay Product Networks

  • Lee, Sunghee;Roh, Hyunsuk;Lee, Hyunwoo;Chung, Kwangsue
    • Journal of Communications and Networks
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    • v.16 no.3
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    • pp.344-354
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    • 2014
  • Transmission control protocol friendly rate control (TFRC) is designed to mainly provide optimal service for unicast applications, such as multimedia streaming in the best-effort Internet environment. However, high bandwidth networks with large delays present an environment where TFRC may have a problem in utilizing the full bandwidth. TFRC inherits the slow-start mechanism of TCP Reno, but this is a time-consuming process that may require many round-trip-times (RTTs), until an appropriate sending rate is reached. Another disadvantage inherited from TCP Reno is the RTT-unfairness problem, which severely affects the performance of long-RTT flows. In this paper, we suggest enhanced TFRC for high quality video streaming over high bandwidth delay product networks. First, we propose a fast startup scheme that increases the data rate more aggressively than the slow-start, while mitigating the overshooting problem. Second, we propose a bandwidth estimation method to achieve more equitable bandwidth allocations among streaming flows that compete for the same narrow link with different RTTs. Finally, we improve the responsiveness of TFRC in the presence of severe congestion. Simulation results have shown that our proposal can achieve a fast startup and provide fairness with competing flows compared to the original TFRC.

A Fast Code Propagation Scheme in Wireless Sensor Networks (무선 센서 네트워크에서 신속한 코드 전송 기법)

  • Lee, Han-Sun;Chung, Kwang-Sue
    • Journal of KIISE:Information Networking
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    • v.35 no.1
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    • pp.1-10
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    • 2008
  • Once the sensor node in wireless sensor networks is installed, it usually operates without human intervention for a long time. The remote code update scheme is required because it is difficult to recall the sensor node in many situations. Therefore, studies on the reliable and efficient transport protocol for code propagation in wireless sensor networks have been increasingly done. However, by considering only the stability aspect of transmission, most of previous works ignore the consideration on the fast code propagation. This results the energy inefficiency by consuming unnecessary energy due to the slow code propagation. In this paper, in order to overcome limitation of the previous code propagation protocols, we propose a new code propagation protocol called "FCPP(Fast Code Propagation Protocol)". The FCPP aims at improving the reliability at well as performance. For this purpose, the FCPP accomplishes the fast code propagation by using the RTT-based transmission rate control and NACK suppression scheme, which provides a better the network utilization and avoids a unnecessary transmission delay. Based on the ns-2 simulation result, we prove that the FCPP Improves significantly both reliability and performance.

An Adaptive UDT Rate Control Method on Network Traffic Condition (네트워크 트래픽 상태에 적응적인 UDT Rate Control 기법)

  • An, Do-Sik;Wang, Gi-Cheol;Kim, Seung-Hae;Cho, Gi-Hwan
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.48 no.3
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    • pp.19-27
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    • 2011
  • Due to the growth of optical fiber network technologies, most networks recently support several tens of Gb/s bandwidth. UDT(UDP-based Data Transfer protocol) is an optimized protocol for bulk data transmission on the network, which has the high bandwidth and long delay time. It periodically controls the sending rate on the fixed sync-interval, 0.01 sec. Here, if the network traffic status varies quickly, available network bandwidth is not able to be properly utilized in-between the sync-intervals. In this paper, we propose an adaptive rate control method with considering the network traffic status in-between the sync-interval periods. The network traffic status is determined based on the RTT variances. With dividing the network status into four classes, the proposed method performs a new rate control by adjusting the inter-packet sending period for a corresponding class. As a simulation result, the proposed method improves the message delivery throughput as well as stability than that of the existing UDT Protocol.

Performance Evaluation of PEP based TCP Splitting Scheme in Satellite Communication Systems (위성 통신 시스템에서 TCP연결 분할 기반 PEP의 성능 평가)

  • Weldegiorgis, Nathnael Gebregziabhe;Lee, Kyu-Hwan;Kim, Jong-Mu;Kim, Jae-Hyun
    • Journal of the Institute of Electronics and Information Engineers
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    • v.52 no.8
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    • pp.10-17
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    • 2015
  • A satellite communication system is one of viable solutions for Internet applications running in wide areas. However, the performance of TCP can be seriously degraded in the satellite networks due to long round-trip time (RTT) and high bit error rate (BER) over satellite links. Therefore, a performance enhancing proxy(PEP) based TCP splitting connection scheme is used in the satellite link to improve the TCP performance. In this paper, we implement PEP testbed and conduct experiment to evaluate the performance of TCP splitting connection by comparing with high-speed TCP solutions in various environments. In our experimental environment, we consider multiple connections, high packet loss, and limited bandwidth. The experiment results show that PEP improves the TCP throughput than high-speed TCP variants in various environments. However, there is no improvement of the TCP throughput with the limited bandwidth because there is packet loss caused by both the congestion and the channel error.

Improvement of physiological activity and processing quality through structural transformation of natural biomaterials based on radiation technology (방사선분자변환기술 기반 천연 생물소재 구조변환에 따른 가공적성 및 생리활성 증진 연구)

  • Byun, Eui-Baek;Song, Ha-Yeon;Kim, Hye-Min;Kim, Woo Sik;Lee, Seung Sik;Choi, Dae Seong;Lim, Sang-Yong;Chung, Byung Yeoup
    • Korean Journal of Food Science and Technology
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    • v.50 no.3
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    • pp.249-259
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    • 2018
  • Radiation technology (RT) has long been applied in various fields for increasing the safety and shelf-life of foods by controlling pathogen-induced poisoning. RT was introduced for the first time in Korea in the 1950s to eliminate harmful microorganisms in food materials. In the 1980s, RT had been scientifically proven to be effective for the sterilization of food and public health products. In recent years, irradiation with gamma rays has also been used for improving physiological properties through the structural modification of natural molecules, which has been proposed to be applicable to various industries. In particular, radiation transformation technology (RTT), which involves the development of new functional compounds through the molecular conversion of natural biomaterials, is becoming a new high-value technology as a fusion technique of RT and biotechnology. The present reports have suggested that RTT can be an effective tool for the development of new functional compounds and improvement of the physiological activity of biomolecules.

Implementation of Video Transfer with TCP-friendly Rate Control Protocol

  • Miyabayashi, Masaki;Wakamiya, Naoki;Murata, Masayuki;Miyahara, Hideo
    • Proceedings of the IEEK Conference
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    • 2000.07a
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    • pp.117-120
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    • 2000
  • As the use of real-time multimedia applications increases, a considerable amount of “greedy” UDP traffic would easily dominate network bandwidth and packet loss. As a result, bandwidth available to TCP connections is oppressed and their performance extremely deteriorates. In or-der that both TCP and UDP sessions fairly co-exist in the Internet, it is vital that we consider the fairness among both protocols. In this work, we implement a “TCP-friendly” rate control mechanism suitable to video applications and con-sider its applicability to a real system through observation of the video quality at the receiver and the connection state. It is shown that we can achieve high-quality and stable video transfer fairly sharing the network bandwidth with TCP by applying our rate control at a control interval of 32 times as long as RTT.

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Improving the TCP Retransmission Timer Adjustment Mechanism for Constrained IoT Networks

  • Chansook Lim
    • International Journal of Internet, Broadcasting and Communication
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    • v.16 no.1
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    • pp.29-35
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    • 2024
  • TCP is considered as one of the major candidate transport protocols even for constrained IoT networks..In our previous work, we investigated the congestion control mechanism of the uIP TCP. Since the uIP TCP sets the window size to one segment by default, managing the retransmission timer is the primary approach to congestion control. However, the original uIP TCP sets the retransmission timer based on the fixed RTO, it performs poorly when a radio duty cycling mechanism is enabled and the hidden terminal problem is severe. In our previous work, we proposed a TCP retransmission timer adjustment scheme for uIP TCP which adopts the notion of weak RTT estimation of CoCoA, exponential backoffs with variable limits, and dithering. Although our previous work showed that the proposed retransmission timer adjustment scheme can improve performance, we observe that the scheme often causes a node to set the retransmission timer for an excessively too long time period. In this work, we show that slightly modifying the dithering mechanism of the previous scheme is effective for improving TCP fairness.

On-the-fly Data Compression for Efficient TCP Transmission

  • Wang, Min;Wang, Junfeng;Mou, Xuan;Han, Sunyoung
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.7 no.3
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    • pp.471-489
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    • 2013
  • Data compression at the transport layer could both reduce transmitted bytes over network links and increase the transmitted application data (TCP PDU) in one RTT at the same network conditions. Therefore, it is able to improve transmission efficiency on Internet, especially on the networks with limited bandwidth or long delay links. In this paper, we propose an on-the-fly TCP data compression scheme, i.e., the TCPComp, to enhance TCP performance. This scheme is primarily composed of the compression decision mechanism and the compression ratio estimation algorithm. When the application data arrives at the transport layer, the compression decision mechanism is applied to determine which data block could be compressed. The compression ratio estimation algorithm is employed to predict compression ratios of upcoming application data for determining the proper size of the next data block so as to maximize compression efficiency. Furthermore, the assessment criteria for TCP data compression scheme are systematically developed. Experimental results show that the scheme can effectively reduce transmitted TCP segments and bytes, leading to greater transmission efficiency compared with the standard TCP and other TCP compression schemes.

Random Linear Network Coding to Improve Reliability in the Satellite Communication (위성 통신에서 신뢰성 향상을 위한 랜덤 선형 네트워크 코딩 기술)

  • Lee, Kyu-Hwan;Kim, Jae-Hyun
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.38B no.9
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    • pp.700-706
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    • 2013
  • In this paper, we propose a method for applying random linear network coding in satellite communication to improve reliability. In the proposed protocol, network-coded redundancy (NC-R) packets are transmitted in the PEP (Performance Enhancement Proxy). Therefore, if data packets is lost by wireless channel error, they can be recovered by NC-R packets. We also develop the TCP performance model of the proposed protocol and evaluate the performance of the proposed protocol. In the simulation results, It is shown that the proposed protocol can improve the TCP throughput as compared with that of the conventional TCP because the NC-R packets is sent by the sender-side PEP and the receiver-side PEP use these packets to recover the lost packets, resulting in reducing the packet loss in TCP.