• The KIPS Transactions:PartC
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• v.12C no.3 s.99
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• pp.437-442
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• 2005

The measurement of RTT (Round Trip Time) can be used for the analysis of Internet congestion. However, simple measuring of RTT which measures only hun around time of a packet can not infer a packet forward/backward delay variation. In this thesis, we present a new algorithm which can be used for the estimation of forward/backward delay variation of packets. These delay variations are implication of network congestion state. In this algorithm, the reference forward/backward delay can be determined based on the minimum RTT value. The delay variation of each packet can be calculated by comparing reference delay with the packet delay. We verified our proposed algorithm by NS-2 simulation and delay measuring in a real network.

• Journal of KIISE:Information Networking
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• v.37 no.1
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• pp.27-34
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• 2010

In this paper, we examine a long-range dependence in an active measurement of a network traffic which has been a well known characteristic from analyses of a passive network traffic measurement. To this end, we utilize RTT(Round Trip Time), which is a typical active measurement measured by PingER project, and perform a relevant analysis to a time series of both RTT and its volatilities. The RTT time series exhibits a long-range dependence or a 1/f noise. The volatilities, defined as a higher-order variation, follow a log-normal distribution. Furthermore, volatilities show a long-range dependence in relatively short time intervals, and a long-range dependence and/or 1/f noise in long time intervals. From this study, we find that the long-range dependence is a characteristic of not only a passive traffic measurement but also an active measurement of network traffic such as RTT. From these findings, we can infer that the long-range dependence is a characteristic of network traffic independent of a type of measurements. In particular, an active measurement exhibits a 1/f noise which cannot be usually found in a passive measurement.

• Journal of KIISE
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• v.43 no.8
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• pp.919-926
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• 2016

In a wireless network, TCP causes the performance degradation because of mistaking packet loss, which is caused by characteristics of wireless link and throughput oscillation due to change of devices connected on a limited bandwidth. Delay based TCP is not affected by packet loss because it controls window size by using the RTT. Therefore, it can solve the problem of unnecessary degradation of the rate caused by misunderstanding reason of packet loss. In this paper, we propose an algorithm for improving the remaining problems by using delay based TCP. The proposed scheme can change throughput adaptively by adding the RTT, which rapidly reflects the network conditions to BaseRTT. It changes the weight of RTT and the increases and decreases window size based on the remaining amount of the buffer. The simulation indicated that proposed scheme can alleviate the throughput oscillation problem, as compared to the legacy TCP Vegas.

• The KIPS Transactions:PartC
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• v.13C no.3 s.106
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• pp.353-358
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• 2006

Unlike TCP Reno, TCP Vegas recognizes network congestion through the measuring of RTT (Round Trip Time) and decides the main congestion control parameters, such as Windows size. But, congestion avoidance scheme of Vegas poorly reflects asymmetric characteristics of packet path because TCP Vegas uses the measuring of RTT that reflects forward/backward packet transmission delay as a forward delay. The RTT can't infer the forward/backward transmission delay variation because it only measures the packet's turn around time. In this paper, We have designed and implemented a new Vegas congestion control algorithm that can distinguish forward/backward network congestion. We have modified the source codes of TCP Vegas in Linux 2.6 kernel and verified their performance.

• Proceedings of the Korean Information Science Society Conference
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• 2004.04a
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• pp.520-522
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• 2004

본 논문은 CDMA2000 1x 망에서 이동단말 측의 측정정보와 TCP 동작 수정을 통하여 TCP 윈도우 크기를 제한하고 RTT(Round Trip Time)을 일정 수준으로 안정화시키는 방안을 제시한다. CDMA2000 1x망은 망 구성의 특성으로 인하여 큰 RTT 값을 가지면 변화가 큰 전송률을 제공한다. TCP는 이러한 CDMA2000망에서 지나치게 큰 전송 윈도우로 인하여 변동이 큰 RTT를 보이며 베이스 스테이션의 버퍼링 부하를 증가시킨다. 본 논문에서는 CDMA2000 1x망에서 사용되는 PPP프로토콜을 이용하여 무선 최대 전송률을 측정하고, TCP timestamp option을 이용하여 RTT를 측정하는 방안을 제시한다. 또한 이 측정값을 적용하여 TCP 수신측에서 RTT와 베이스 스테이션의 버퍼링을 일정수준으로 유지시키는 방안을 제시하고, 리눅스 프로토콜스택 수정을 통해 적용 결과를 보인다.

• Proceedings of the Korea Information Processing Society Conference
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• 2005.11a
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• pp.1299-1302
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• 2005

TCP 구현의 하나인 Vegas는 패킷의 유실을 망의 혼잡으로 인지하는 Reno와 달리 RTT(Round Trip Time) 측정값을 바탕으로 혼잡을 인지하며 윈도우 크기 등 혼잡 제어를 위한 주요 인자를 결정한다. TCP Vegas는 TCP Reno보다 더 효율적인 네트워크 대역폭과 처리율을 가진다. 그러나 Vegas의 혼잡 회피 방안이 TCP 패킷 경로의 비대칭적 특성을 제대로 반영하지 못하며, 이것은 양방향(순방향,역방향) 패킷 전송 상태를 반영하는 RTT 측정값을 순방향 경로의 상태 해석에 이용하기 때문이다. RTT는 패킷의 왕복 시간만을 측정하기 때문에 패킷의 송수신시 순방향과 역방향에서 어느 정도의 혼잡이 발생하였는지 알 수 없다. 본 논문에서는 리눅스 커널의 TCP 소스에서 RTT 측정값으로 혼잡도를 측정하는 기존의 Vegas 혼잡 제어 알고리즘을 수정하여 순방향 경로의 혼잡과 역방향 경로의 혼잡을 구별할 수 있는 새로운 Vegas 혼잡 제어 알고리즘을 설계하고 구현하여 그 성능을 분석하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.7B
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• pp.1260-1269
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• 2000

TCP 구현의 하나인 Vegas는 RTT 측정값을 바탕으로 혼잡을 인지하며 윈도우 크기 등 혼잡제어를 위한 주요 인자를 결정한다. 이러한 Vegas가 기존 TCP 구현에 비해 우수한 성능과 높은 공평성을 가짐은 여러 논문을 통해 보고되었다. 그런데 우리는 Vegas의 혼잡회피 방안이 TCP 데이터 흐름의 비대칭적 특성을 제대로 반영하지 못하며, 이것이 양쪽 방향 상태를 반영하는 RTT 측정값을 순방향 링크의 상태 해석에 이용하기 때문임을 발견하였다. 우리는 이를 Vegas RTT Ambiguity 문제라 하고, 네트워크 내의 패킷 스케쥴링 방법인 Small Get Priority Queue, TCP Timestamp Options을 이용한 수정 Vegas 구현 등 두 가지 해결 방안을 제시하였다. 우리는 시뮬레이션을 통해 Vegas RTT Ambiguity 문제와 제시한 해결 방안들을 검증하였다.

• The Transactions of the Korea Information Processing Society
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• v.7 no.5
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• pp.1520-1524
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• 2000

TCP congestion control algorithm prevents network congestion through the control of outgoing traffic size. The network, therefore, should monitor the incoming traffic size of a TCP to determine whether or not a TCP follows standard congestion control algorithms. Some TCP friendly test algorithms are proposed, But, these algorithms cannot be used in real environments because a router in a network does not know the RTT of a TCP flow. In this study, we propose a new RTT determination algorithm that can be used in a router. Our proposed algorithms is validated through the simulation studies.

• Journal of Digital Contents Society
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• v.16 no.3
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• pp.381-388
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• 2015

TCP, which performs congestion control in congestion condition, is able to help a reliable transmission. However, packet loss can be increased because congestion window is increased by the time the packet is dropped in the process of congestion avoidance. In this paper, to solve the above problem, we propose a new congestion estimation based TCP congestion control scheme using the weighted average value of the RTT. After measuring a SRTT, which means the weighted average value of RTTs, at this point of time when a buffer overflow is occurred by an overloaded packet, the proposed scheme estimates the time, when the same SRTT is made in packet transmission, as a congestion time and then decreases the congestion window. The simulation results show that the proposed schem has a good performance in terms of packet loss rate and throughput when the packet loss due to buffer overflow is larger than that due to wireless channel.

• The Journal of the Korea Contents Association
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• v.11 no.2
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• pp.69-78
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• 2011

Recently, the bandwidth available to an end user has been dramatically increasing with the advancing of network technologies. This high-speed network naturally requires faster and/or stable data transmission techniques. The UDT(UDP based Data Transfer protocol) is a UDP based transport protocol, and shows more efficient throughput than TCP in the long RTT environment, with benefit of rate control for a SYN time. With a NAK event, however, it is difficult to expect an optimum performance due to the increase of fixed sendInterval and the flow control based on the previous RTT. This paper proposes a rate control method on following a NAK, by adjusting the sendInterval according to some degree of RTT period which calculated from a set of experimental results. In addition, it suggests an improved flow control method based on the TCP vegas, in order to predict the network congestion afterward. An experimental results show that the revised flow control method improves UDT's throughput about 20Mbps. With combining the rate control and flow control proposed, the UDT throughput can be improved up to 26Mbps in average.