• Journal of Korea Multimedia Society
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• v.7 no.9
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• pp.1273-1281
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• 2004

There have been a number of researches on real-time packet scheduling based on EDF algorithm to support end-to-end delay bound guarantees for real-time traffic transmission. However, EDF-based packet scheduler could not guarantee the real-time requirements of real-time traffic if there exist non-real-time traffic. In this paper, we propose a new admission control algorithm and packet scheduling scheme considering non-real-time traffic in the real -time packet scheduler based on EDF policy. Proposed admission control algorithm has pseudo-polynomial time complexity, but we show through simulation that it can be used with little run-time overhead.

• Journal of Institute of Control, Robotics and Systems
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• v.13 no.12
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• pp.1173-1178
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• 2007

Ubiquitous sensor network (USN) is composed of many sensor nodes which are one of the simplest form of embedded system. In developing the sensor network system, a debugging tool is necessary to test and verify the system. Recently, a so-called packet analyzer has been developed for this purpose, and it supports IEEE 802.15.4 which is considered as the standard for the sensor network protocols. The major function of the packet analyzer is to take RF packets from sensor nodes and show the structure and the data of the packets graphically to the user. However, the conventional packet analyzers do not support remote control because they require a USB interface along with a personal computer. To make it available for remote control, we propose a new packet analyzer based on a server-client scheme in which a server program is implemented on embedded Linux and a client program is implemented on Windows for convenient use.

• Journal of Korea Game Society
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• v.2 no.1
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• pp.23-29
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• 2002

Multicast protocols are efficient methods of group communication such as video conference, Internet broadcasting and On-Line Game, but they do not support the various transmission protocol services like a reliability guarantee, FTP, or Telnet that TCPs do. The Purpose or this Paper is to find a method to utilize multicast routers can simultaneously transport multicast packets and TCP packets. For multicast network scalability and error recovery the existing SRM(Scalable Reliable Multicast)method has been used. Three packets per TCP transmission control window site are used for transport and an ACK is used for flow control. A CBR(Constant Bit Rate) and a SRM is used for UDP traffic control. Divided on whether a UDP multicast packet and TCP unicast packet is used simultaneously or only a UDP multicast packet transport is used, the multicast receiver with the longest delay is measured on the number of packets and its data receiving rate. It can be seen that the UDP packet and the TCP's IP packet can be simultaneously used in a server router.

• IEMEK Journal of Embedded Systems and Applications
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• v.1 no.1
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• pp.24-30
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• 2006

With the original Transmission Control Protocol(TCP) design, which is particularly targeted at the wired networks, a packet loss is assumed to be caused by the network congestion. In the wireless environment where the chances to lose packets due to transmission bit errors are not negligible, though, this assumption may result in unnecessary TCP performance degradation. In these days, many papers describe about wireless-TCP which has suggested how to avoid congestion control when packet loss over the wireless network. In this paper, an enhancement scheme is proposed by modifying SNOOP scheme. To enhance the original SNOOP scheme, CPC(Consecutive Packet Control) and ZWSC(Zero Window Size Control) are added. The invocation of congestion control mechanism is now minimized by knowing the cause of packet loss. We use simulation to compare the overhead and the performance of the proposed schemes, and to show that the proposed schemes improve the TCP performance compares to SNOOP by knowing the cause of packet loss at the base station.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.13 no.2
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• pp.171-181
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• 1988

In this study, a packet voice/data terminal(PVDT) that services both voice and data in the packet-switched network is implemented. The software structure of the PVDT is designed according to the OSI 7 layer architecture. The discrimination of voice and data is made in the link layer. Voice packets have priority over data packets in order to minimize the transmission delay, and are serviced by a simple protocol so that the overhead arising form the retransmission of packets may be minimized. The hardware structure of the PVDT is divided into five modules; a master control module, a speech proessing module, a speech activity detection module, a telephone interface module, and an input/output interface module. In addition to the hardware implementation, the optimal reconstruction delay of voice packets to reduce the influence of delay variance is analyzed.

• The Journal of the Korea institute of electronic communication sciences
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• v.10 no.2
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• pp.289-296
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• 2015

Wireless multimedia sensor network senses and transfers mass multimedia data. Also, it is sensitive to latency. This thesis proposes a routing technique based on traffic priority in order to improve the network efficiency by minimizing latency. In addition, it proposes a congestion control mechanism that uses packet service time, packet inter-arrival time, buffer usage, etc. In this thesis, we verified the reduction of packet latency in accordance with the quality level of packet as a result of the performance analysis through the simulation method. Also, we verified that the proposed mechanism maintained a reliable network state by preventing packet loss due to network overload.

• ETRI Journal
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• v.38 no.6
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• pp.1064-1073
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• 2016

An adaptive speech streaming method to improve the perceived speech quality of a software-based multipoint control unit (SW-based MCU) over IP networks is proposed. First, the proposed method predicts whether the speech packet to be transmitted is lost. To this end, the proposed method learns the pattern of packet losses in the IP network, and then predicts the loss of the packet to be transmitted over that IP network. The proposed method classifies the speech signal into different classes of silence, unvoiced, speech onset, or voiced frame. Based on the results of packet loss prediction and speech classification, the proposed method determines the proper amount and bitrate of redundant speech data (RSD) that are sent with primary speech data (PSD) in order to assist the speech decoder to restore the speech signals of lost packets. Specifically, when a packet is predicted to be lost, the amount and bitrate of the RSD must be increased through a reduction in the bitrate of the PSD. The effectiveness of the proposed method for learning the packet loss pattern and assigning a different speech coding rate is then demonstrated using a support vector machine and adaptive multirate-narrowband, respectively. The results show that as compared with conventional methods that restore lost speech signals, the proposed method remarkably improves the perceived speech quality of an SW-based MCU under various packet loss conditions in an IP network.

• 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.

• Proceedings of the KIEE Conference
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• 2002.11c
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• pp.509-512
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• 2002

We propose a control method for mobile robot system using the bluetooth. The control packet is defined and used for control of the mobile robot. The control packet is composed of behavior components and has reserved packets for future working. The control packet has to be simple and provide commands to the mobile robot, since the bluetooth has a limited bandwidth. The data transmission rate and the distance, which can control the mobile robot in various circumstances, for example, corridor, yard, and room are measured by some experiments.