• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.8
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• pp.2090-2095
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• 1998

In this paper, we measured voice packet loss over the Korean Internet and analyzed percentage articulation by variation of the packet loss. To do this, we reviewed real-time transmission service based on RTP/UDP/IP and test method of the transmission quality. and implemented the real-time speech transmission system using GSM and UDP/IP. Monosyllable list has been chosen for the percentage articulation test, each voice packet has been coded and compressed by GSM and it has sequence number to measured packet loss and to recover out-of-order packets. In transmission results using seven router over the Korean Internet, we have show that loss rates reached 1.6% (unload), 22.5%(load) and loss rates after packet recovery by resequencing and FEC are from 9% to 35%. Finally, we have shown that percentage articulations by variation of the network traffic are Table 4.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.1
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• pp.590-593
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• 2005

In this paper, it is analyzed a transmission time of ACL Packet in bluetooth wireless link. In order for segment to improve the transfer capability in bluetooth system, it is fragmented in TCP total messages that are coming down from upper layer and then the packets are sent one at time in baseband. And it is studied that transmission time for bluetooth wireless link according to DM1, DM3 or DM5 packet type in bluetooth piconet environment. From the results, we were able to obtain ACL packet transmission time, optimal TCP packet size and DM packet size.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.26 no.6
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• pp.1571-1582
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• 2016

One of the issues, which is dealed with in undirectional data transmission technology, is reducing the packet loss in TCP based data transfer. We can decrease the packet loss by using several well known error correction approaches. Although we utilize those previous approaches, the packet loss by both link errror and buffer overflow could be occurred. In this paper, we propose the RED(REliable Data diode). RED also uses the TCP proxy approach for supporting the TCP based data transfer which is similar with the existing unidirectional data transmission technologies. The RED transmission system could alleviate the packet loss caused by buffer overflow by exploiting the delaying transmission of TCP packets. Furthermore, in order to reduce the packett loss caused by link error in the unidirectional transmission link, the RED transmission system transmits one or more duplicated packets to the RED reception system by considering both the remaining resources and packet importance.

• IEMEK Journal of Embedded Systems and Applications
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• v.6 no.3
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• pp.149-156
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• 2011

Transmission delay and packet loss induced by a communication network can degrade the control performance and, even make the system unstable. This paper presents a method for compensating transmission delay and packet loss in a networked control system for unmanned underwater vehicle. The proposed method is based on Lagrange interpolation in order to satisfy the requirements of simplicity and model-independency. In this work, the lost/delayed data are estimated in real time by only using the past data without requiring any mathematical model of the controlled system. Consequently, the proposed method can be implemented independent of the controlled system, and also it can achieve fast and accurate compensation performance. The performance of the proposed technique is evaluated by numerical simulations with an unmanned underwater vehicle.

• Journal of Internet Computing and Services
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• v.16 no.5
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• pp.11-17
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• 2015

In the general transmission power control model that is used for wearable sensor systems, if RSSI value gets out of the Target RSSI Margin, then the sink node finds new transmission power by using TPC(Transmission Power Control) Algorithm. At this time, the sink node sends the control packet to the sensor node for delivering the newly calculated transmission power. However, when the wireless network channel condition is poor, even it is consuming a lot of control packets, the sink node could not find an appropriate transmission power so it only waste of energy. Therefore, we proposed a new control packet transmission decision method that the sink node changes the transmission power when the wireless network channel condition is stabilized. It makes waste of energy decline. In this paper, we apply control packet transmission decision method to Binary TPC algorithms and analyze the results to evaluate the proposed method. We propose three methods that judge the state of wireless network channel. We experiment that methods and analysis the results.

• Journal of Korean Institute of Industrial Engineers
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• v.33 no.3
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• pp.381-392
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• 2007

In this paper, we present an analytical method for estimating the saturation throughput of an 802.11 ad hoc LAN in the presence of noise distorting transmitted frames. This is the first method that allows studying analytically the 802.11 network performance with consideration of correlated channel failures usually inherent to realistic wireless channels. With the study, we consider the possible packet fragmentation that can be adopted to reduce the performance degradation caused by noise-induced distortions. In addition to the throughput, our method allows estimating the probability of a packet rejection occurring when the number of packet transmission retries attains its limit. The obtained numerical results of investigating 802.11 LANs by the developed method are validated by simulation and show high estimation accuracy as well as the method efficiency in determining the optimal fragmentation threshold.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.5
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• pp.1311-1321
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• 2010

Sensor network is used to obtain sensing data in various area. The interval to sense the events depends on the type of target application and the amounts of data generated by sensor nodes are not constant. Many applications exploit long sensing interval to enhance the life time of network but there are specific applications that requires very short interval to obtain fine-grained, high-precision sensing data. If the number of nodes in the network is increased and the interval to sense data is shortened, the amounts of generated data are greatly increased and this leads to increased amount of packets to transfer to the network. To transfer large amount of packets fast, it is necessary that the delay between successive packet transmissions should be minimized as possible. In this paper, we propose SET(SendDoneEventbasedTransmission Technique)which reduces the delay between successive packet transmissions by using SendDone Event which informs that a packet transmission has been completed. In SET, the delay between successive packet transmissions is shortened very much since the transmission of next packet starts at the time when the transmission of previous packet has completed, irrespective of the transmission time. Therefore SET could provide high packet transmission rate given large amount of packets.

• Proceedings of the IEEK Conference
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• 2000.06a
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• pp.213-216
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• 2000

In this paper, performance of a CDMA-based packet transmission system is analyzed and simulated in a wireless infrared channel. The indoor wireless infrared channel is modeled as a non-directed diffuse link. The pulse position modulation (PPM) is used as a modulation scheme. From the simulation results, it is shown that the RS coding is very effective in improving the packet-delay characteristics of the CDMA-based packet transmission system. The performance improvement is more significant especially when the infrared channel is in a worse condition.

• Journal of the Korea Society of Computer and Information
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• v.28 no.2
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• pp.39-46
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• 2023

We present an efficient packet transmission strategy for massively multiplayer online first-person shooter (MMOFPS) games using movement-adaptive packet transmission interval. The player motion in FPS games shows a wide spectrum of movement variability both in speed and orientation, where there is room for reducing the number of packets to be transmitted to the server depending on the predictability of the character's movement. In this work, the degree of variability (nonlinearity) of the player movements is measured at every packet transmission to calculate the next transmission time, which implements the adaptive transmission frequency according to the amount of movement change. Server-side prediction with a few auxiliary heuristics is performed in concert with the incoming packets to ensure reliability for synchronizing the connected clients. The comparison of our method with the previous fixed-interval transmission scheme is presented by demonstrating them using a test game environment.

• Journal of Communications and Networks
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• v.6 no.3
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• pp.235-244
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• 2004

There have been a lot of approaches to evaluate and predict transmission control protocol (TCP) performance in a numerical way. Especially, under the recent advance in wireless transmission technology, the issue of TCP performance over wireless links has come to surface. It is because TCP responds to all packet losses by invoking congestion control and avoidance algorithms, resulting in degraded end-to-end performance in wireless and lossy systems. By several previous works, although it has been already proved that overall TCP performance is largely dependent on its loss recovery performance, there have been few works to try to analyze TCP loss recovery performance with thoroughness. In this paper, therefore, we focus on analyzing TCP's loss recovery performance and have developed a simple model that facilitates to capture the TCP sender's behaviors during loss recovery period. Based on the developed model, we can derive the conditions that packet losses may be recovered without retransmission timeout (RTO). Especially, we have found that TCP Reno can retransmit three packet losses by fast retransmits in a specific situation. In addition, we have proved that successive three packet losses and more than four packet losses in a window always invoke RTO easily, which is not considered or approximated in the previous works. Through probabilistic works with the conditions derived, the loss recovery performance of TCP Reno can be quantified in terms of the number of packet losses in a window.