• 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.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.652-657
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• 2004

As the real-time multimedia applications through Internet increase, the bandwidth available to TCP connections is oppressed by the UDP traffic, result in the performance of overall system is extremely deteriorated. Therefore, developing a new transmission protocol is necessary. The TCP-friendly algorithm is an example meeting this necessity. The TCP-friendly (TFRC) is an UDP-based protocol that controls the transmission rate based on the available round transmission time (RTT) and the packet loss rate (PLR). In the data transmission processing, transmission rate is determined based on the conditions of the previous transmission period. If the one-step ahead predicted values of the control factors are available, the performance will be improved significantly. This paper proposes a prediction model of transmission rate control factors that will be used for the transmission rate control, which improves the performance of the networks. The model developed through this research is predicting one-step ahead variables of RTT and PLR. A multiplayer perceptron neural network is used as the prediction model and Levenberg-Marquardt algorithm is used for the training. The values of RTT and PLR were collected using TFRC protocol in the real system. The obtained prediction model is validated using new data set and the results show that the obtained model predicts the factors accurately.

• Journal of Institute of Control, Robotics and Systems
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• v.18 no.11
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• pp.1040-1044
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• 2012

We design a NCS (Networked Control System) where the communication between sensors and controllers takes place over a USN (Ubiquitous Sensor Network). In order to measure time delays between sensors and controllers in real time, we design an algorithm to measure RTT (Round Trip Time) between USN nodes, and implement it into TinyOS of USN. By using the measured time delays, we construct the Smith predictor to compensate the time delays between sensors and controllers in real-time. For the real time experiment, we simulate the dynamic plant model, controller, and USN interface using Real-Time Windows Target provided in MATLAB. The USN interface in the Simulink model consists of serial ports, which connect the plant output and controller with USN nodes. The experiment results show that the time delays between sensors and controllers are precisely measured in real time; the Smith predictor appropriately compensates the time delays; and the stability is achieved in the closed-loop of the NCS.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.17 no.3
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• pp.332-337
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• 2016

The rapid reconstruction of a transfer path is caused by frequent secession of nodes in many applications using overlay multicast. For this, however, many studies have been proposed, but a study of fast path recovery is required. This paper proposes a new method for fast path recovery to improve the path reconstruction time and for stable path maintain caused by the sudden secession of the parent node on overlay multicast. The proposed method uses the virtual transfer path and the candidate parent node for the fast path recovery. All nodes exchange periodically the RTT (Round Trip Time) information between the other nodes of similar position itself and neighboring node. All nodes have a candidate parent node list, and each node stores the exchanged RTT information on the list. Each node constructs the virtual transfer path to reduce the recovery time after deciding a candidate parent node that is one of them by the RTT order. In this way, when the parent node is seceded, all nodes can recover the transfer path quickly using the virtual transfer path. The simulation result showed that the recovery time of the proposed method is an average 30% faster than the known method.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2015.05a
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• pp.649-652
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• 2015

Previous marking policy for the AF service of TCP traffic in the Diffserv network have no sufficient consideration on the effect of RTT and target rate. In this paper, in order to improve fairness Index by the effect RTT difference of TCP traffic, we propose the modified TSW3CDM(Time Sliding Window Three Color Dynamic Marker) based on average transfer rate estimation and the flow state. The proposed algorithm is dynamic marking policy that do allocate band width in proportion to transmission rate. To evaluate the performance of the proposed algorithm, We accomplished a computer simulation using NS-2. From simulation results, the proposed TSW3CDM algorithm improves fairness index by comparison with TSW3CM.

• 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.

• Journal of KIISE:Information Networking
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• v.30 no.6
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• pp.764-775
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• 2003

Previous works for the assured service in the Diffserv network have no sufficient consideration on the fairness of bandwidth share based on RTTs and the target rates of TCP flows. In this paper, in order to solve these problems, we propose a TRA3CM(Target rate and RTT Aware 3 Color Marking) mechanism. The TRA3CM mechanism provides three color marking and fair transmission rates among aggregate flows by considering RTT and target rate simultaneously. In case of higher target rate than bottleneck bandwidth, the TRA3CM mechanism is able to mitigate the RTT effect and provides fair transmission rates. In the results of comparing the performance among existing mechanisms and the TRA3CM, the TRA3CM mechanism was able to mitigate the RTT effect better than the former. The TRA3CM is shown to provide good performance for transmission rates proportional to various target rates.

• Proceedings of the IEEK Conference
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• 2000.11a
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• pp.17-20
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• 2000

TCP is ill-suited to real-time multimedia applications. Its bursty transmission, and abrupt and frequent wide rate fluctuations cause high delay jitters and sudden quality degradation of multimedia applications. Deploying non congestion controlled traffic results in extreme unfairness towards competing TCP traffic. Therefore, they need to be enhanced with congestion control schemes that not only am at reducing loss ratios and improve bandwidth utilization but also are fair towards competing TCP connections. This paper proposes a differentiated rate adaptation algorithm based on loss and round trip time. Rate in a sender quickly responds to loss ratio and holds steady state. Additionally, this algorithm reduces loss ratio by loss prediction in a receiver.

• Proceedings of the Korean Information Science Society Conference
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• 2001.04a
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• pp.304-306
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• 2001

오늘날 인터넷에서 멀티미디어 컨텐트는 날로 다양해지고 있으며 특히 무선망에서 미디어 스트리밍에 대한 요구는 점점 커지고 있다. 그런데 무선망은 낮은 대역폭과 높은 전송 에러율로 인하여 유선망과 같은 스트리밍 서비스를 제공하기가 힘들다. 본 논문에서는 자바로 스트리밍 시스템을 구현하고, 이를 이용하여 무선망에서의 스트리밍 특성을 살펴본다. 본 논문에서 수행된 실험은 CDMA 2000 무선 인턴넷 환경을 배경으로 하며, 패킷 전송 시간(round trip time:RTT)과 패킷 손실 양상을 관찰하고 이에 따라 무선망에서 적합한 패킷 손실 복구나 버퍼링 기법의 필요성을 제안한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2004.05a
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• pp.1371-1374
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• 2004

본 논문에서는 내부 LAN(Local area network)에서 MSS(Maximum segment size)의 변화가 네트워크 성능에 미치는 영향에 대하여 분석한다. 내부 LAN에 웹 클라이언트와 웹 서버를 두고 이더넷(Ethernet)이 허용하는 사이즈 내에서 MSS 값의 변화가 네트워크의 성능에 미치는 영향을 분석한다. 처리량(Throughput), 이용률(Utilization), 분실율, 수신율, RTT(Round Trip Time)를 사용하여 성능을 평가하였다. 실험은 네트워크 시뮬레이터 중의 사실상 표준 도구인 NS-2로 한다.