• Journal of Institute of Control, Robotics and Systems
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• v.9 no.3
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• pp.196-203
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• 2003

Recent increase in accessibility to the internet makes it easy to use the internet-connected devices. The internet could allow any user can reach and command any device that is connected to the network. But these teleoperation systems using the internet connected device have several problems such as the network time delay, data loss and development cost of an application for the communication with each other. One feasible solution is to use local and external network line for the network time delay, transmission control protocol for data loss and Java language to reduce the development period and cost. In this study, web-based remote control system using Java language is newly proposed and implemented to a pneumatic servo control system to solve the time delay, data loss and development cost. We have conducted several experiments using pneumatic rodless cylinder through the internet and verified that the proposed remote control system was very effective.

• Journal of KIISE:Information Networking
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• v.36 no.2
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• pp.130-136
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• 2009

Delay and Disruption Tolerant Network (DTN) employs message delivery based on a store-and-forward method to conquer no guarantee of continuous end-to-end connectivity. Different from general networks, it is hard for the existing TCP/IP-based routing protocols to correctly work, due to the characteristics such as large latency and unstable link connectivity. Thus, many recent studies focus on routing protocols for DTN. In this paper, we propose a novel routing protocol for DTN with efficient message delivery utilizing mobility information such as direction or destination of mobile nodes. And this protocol is enhanced in terms of delivery ratio, decreases latency and overhead compared to the previously proposed solutions. Simulation results prove its superiority.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.1B
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• pp.29-40
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• 2004

In this paper, we propose a new early congestion detection and notification technique called QR-AQM. Unlike RED and it's variation, QR-AQM measures the total traffic rate from TCP sessions, predicts future network congestion, and determine the packet marking probability based on the measured traffic rate. By incorporating the traffic rate in the decision process of the packet marking probability, QR-AQM is capable of foreseeing future network congestion as well as terminating congestion resolution procedure in much more timely fashion than RED. As a result, simulation results show that QR-AQM maintains the buffer level within a fairly narrow range around a target buffer level that may be selected arbitrarily as a control parameter. Consequently, compared to RED and its variations, QR-AQM is expected to significantly reduce the jitter and delay variance of packets traveling through the buffer while achieving nearly identical link utilization.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.1
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• pp.45-57
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• 2005

In this paper, we extend the performance of bidirectional TCP connection over end-to-end network that uses transfer rate-based flow and congestion control. The sharing of a common buffer by TCP packets and acknowledgement has been known to result in an effect called ack compression, where acks of a connection arrive at the source bunched together, resulting in unfairness and degraded throughput. The degradation in throughput due to bidirectional traffic can be significant. Even in the simple case of symmetrical connections with adequate window size, the connection efficiency is improved about 20% for three levels of background traffic 2.5Mbps, 5.0Mbps and 7.5Mbps. Otherwise, the throughput of jitter is reduced about 50% because round trip delay time is smaller between source node and destination node. Also, we show that throughput curve is improved with connection rate algorithm which is proposed for TCP congetion avoidance as a function of aggressiveness threshold for three levels of background traffic 2.5Mbps, 5Mbps and 7.5Mbps. By analyzing the periodic bursty behavior of the source IP queue, we derive estimated for the maximum queue size and arrive at a simple predictor for the degraded throughput, applicable for relatively general situations.

• Journal of Communications and Networks
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• v.5 no.1
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• pp.65-72
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• 2003

Internet roundtrip delay/time (RTT) prediction plays an important role in detecting packet losses in reliable transport protocols for traditional web applications and determining proper transmission rates in many rate-based TCP-friendly protocols for Internet-based real-time applications. The widely adopted autoregressive and moving average (ARMA) model with fixed-parameters is shown to be insufficient for all scenarios due to its intrinsic limitation that it filters out all high-frequency components of RTT dynamics. In this paper, we introduce a novel parameter-varying RTT model for Internet roundtrip time prediction based on the information theory and the maximum entropy principle (MEP). Since the coefficients of the proposed RTT model are updated dynamically, the model is adaptive and it tracks RTT dynamics rapidly. The results of our experiments show that the MEP algorithm works better than the ARMA method in both RTT prediction and RTO estimation.

• Proceedings of the KIEE Conference
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• 2007.07a
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• pp.1713-1715
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• 2007

Measurement of network traffic have shown that the self-similarity is a ubiquitous phenomenon spanning across diverse network environments. In previous work, we have explored the feasibility of exploiting the long-range correlation structure in a self-similar traffic for the congestion control. We have advanced the framework of the multiple time scale congestion control and showed its effectiveness at enhancing performance for the rate-based feedback control. Our contribution is threefold. First, we define a modular extension of the TCP-a function called with a simple interface-that applies to various flavours of the TCP-e.g., Tahoe, Reno, Vegas and show that it significantly improves performance. Second, we show that a multiple time scale TCP endows the underlying feedback control with proactivity by bridging the uncertainty gap associated with reactive controls which is exacerbated by the high delay-bandwidth product in broadband wide area networks. Third, we investigate the influence of the three traffic control dimensions-tracking ability, connection duration, and fairness-on performance.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2014.05a
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• pp.404-407
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• 2014

Delay Tolerant Network (DTN) is a networking architecture that is designed to solve network problem in unstable and stressed environments using the Store-Carry-Forward method. Different form general networks, DTN does not guarantee the end-to-end connectivity, it is hard for the exiting TCP/IP based protocols to normally work due to the characteristic such as large latency and unstable link connectivity. And the condition that send massage without the information of the destination occurs frequently. Thus, suitable routing protocols for DTN are required and being researched. In this paper, we propose a novel routing protocol for DTN utilizing epidemic and prediction based algorithm to prevent the defect of previously DTN routing protocols such as the absence of the holistic view of the network. Proposed algorithm predicted destination using the mobility information recorded in neighbor node's history and utilize epidemic based algorithm when occurred condition without destination's information. This algorithm is enhanced in terms of delivery ratio, decreases latency and overhead in sparse network such as DTN.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.11
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• pp.2553-2562
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• 2013

It is difficult for TCP congestion control algorithm to ensure the bandwidth and delay bound required for media streaming services in broadband wireless network environments. In this paper, we propose the COIN TCP (COncave INcrease TCP) scheme for providing a high-quality media streaming services. The COIN TCP concavely increases the congestion window size by adjusting the increment rate of congestion window, that is inversely proportional to the amount of data accumulated in the router queue. As a result, our scheme can quickly occupy the available bandwidth and prevent the heavy congestion. It also improves the link utilization by adjusting the decrement rate of congestion window according to the packet loss rate with the random loss. Through the simulation results, we prove that our scheme improves the total throughput in broadband wireless network.

• Journal of information and communication convergence engineering
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• v.2 no.1
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• pp.9-14
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• 2004

In this paper the modeling of the RLP layer in CDMA2000 is presented, which uses the NAK based ARQ scheme for Random Error Channels. The RLP performs a partial link recovery through limited number of RLP frame transmission in case of frame error. In case when the RLP fails due to excessive frame error, the control is passed on to the higher (TCP) layer. The TCP layer provides the complete end-to-end recovery. Thus the reliable performance at the TCP/RLP is essential to maintain the required Quality of Service in the DS-CDMA wireless links. The modeling is done for the performance analysis of the system in terms of the throughput and the mean extra delays, which are calculated analytically and are compared with the results generated by the simulations. This paper studies the effect of the random errors over different types of RLP frame formats and also the performance of the NAK based ARQ mechanism used under these conditions. The simulation provides with the over all characteristics of the throughput and the mean extra delay in terms of realistic environment parameters like Eb/No and probability of packet error (PE), based on the channel conditions.

• Journal of Korea Multimedia Society
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• v.5 no.6
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• pp.697-703
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• 2002

In this paper, we propose the adaptive multi stream transmission technique based on SPIHT video signal for the highest quality service over the current Internet that does not guarantee QoS. In addition to the reliable transmission of the video stream over the asynchronous packet network, the proposed approach provides the transmission using the adaptive frame pattern control and multi steam over the TCP for continuous replay. The adaptive frame pattern control makes the transmission date scalable in accordance with the client's buffer status. Apart from this, the multi stream transmission improves the efficiency of video stream, and is robust to the network jitter problem, and maximally utilizes the bandwidth of the client's. As a result of the experiment, the DR(delay ratio) in the proposed adaptive multi-stream transmission is more close to zero than in the existing signal stream transmission, which enables the best-efforts service to be implemented.