• KSCE Journal of Civil and Environmental Engineering Research
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• v.28 no.6D
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• pp.801-807
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• 2008

This study attempts to evaluate and compare the land consumption efficiencies of various urban transport modes, utilizing the time-area method. With the careful considerations of travel conditions and modal characteristics, equations for the computation of time-area for various different modes are derived. With the appropriate assumed values of parameters, time-area consumptions per person per km for each mode both for peak and off-peak situations are computed and evaluated. The relationships between the time-area consumed and operating speed for each mode are graphically demonstrated and discussed. An example of the time-area consumed for a hypothetical commuter round-trip using various modes are also presented in order to clearly show the consumption of parking requirements by specific modes. It shows that regardless of facilities used, auto users are by far the worst in terms of area efficiency and that transit, especially rail rapid transit, is a superior mode. Pedestrian and bicycle, though consuming more than expected, also demonstrate the advantage in case of a short-distance trip in terms of area efficiency compared with the bus transit modes.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.25 no.4
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• pp.266-274
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• 2015

Vibration exposure of domestic high speed trains(KTX and ITX-saemaeulho) to passenger traveling on the Gyung-Bu line was measured to evaluate health effect which is based on ISO 2631-1. The vibration exposure was compared with the frequency weighted root mean square(r.m.s.) and vibration dose value(VDV) of the two trains’ vibration. It is concluded that vibration exposure of the two train during round trip of Busan-Seoul in single day is evaluated to be safe for the passenger’s health according to the health guidance caution defined on ISO 2631-1. Futhermore KTX’s vibration exposure is found to be significantly lower than ITX-saemaeulho thanks to lower vibration magnitude as well as shorter trip time.

• Journal of Internet Computing and Services
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• v.5 no.3
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• pp.99-108
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• 2004

We have designed and implemented a Voice Messenger System(VMS) for sending and receiving voice data in order to measure and analyze the performance of a multimedia data transfer system using PDAs(Personal Digital Assistants) in a fast wireless LAN environment. The VMS has been configured as an infrastructure network by integrating a wireless intra-network based on IEEE 802,11 b standard with a wired network based on Ethernet. A sending time, a receiving time and a round-trip time(RTT) have been measured by varying its pecket size from 256 to 6656 bytes, Besides, we have performed the experiments to see which NIC is more stable between CF and PCMCIA type in such a wireless LAN environment, The results of such experiments show that the NIC with PCMClA type is better than that with CF type from the points of both efficiency and stability. The performance of a notebook computer with the attachment of a wireless NIC that is used as a VMS client is almost constant by varying its packet size when transferring voice data. However, with PDA client, in the case of receiving packets from the VMS server, the performance is the best when the pecket size is 6656 bytes, In addition, in the case of sending packets from PDA client to the VMS server, the performance is the best when the packet size is 4096 bytes. Even with the same pecket size, the time of receiving packets from the VMS server to PDA is longer than the time of sending packets from PDA to the VMS Server, Thus, we conclude that when PDA is used as a client by applying different pecket size when sending and receiving packets should be achieved a better performance over a wireless LAN environment.

• 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 Society of Broadcast Engineers Conference
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• 2011.07a
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• pp.99-101
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• 2011

본 논문에서는 WiMAX 의 신호세기(CINR: Carrier to Interference + Noise Ratio)의 변화와 예측 방법에 대해 연구한다. WiMAX 와 같은 무선 네트워크에서는 신호세기가 불규칙하게 변화하기 때문에 비디오 스트리밍시 에러나 지연이 발생할 수 있다. CINR 은 클라이언트 측의 가용 대역폭을 결정하는 중요한 요소이다. 하지만 기존의 RTCP(Real-time Transport Control Protocol)를 이용한 비트율 제어는 RTT(Round Trip time), PLR(Packet Loss Rate) 등의 정보를 사용하기 때문에 부정확하거나 지연이 발생될 수 있다. 이를 보완하기 위한 CINR 을 직접 비트율 제어에 사용하는 방법에 대해 연구하기에 앞서, 본 논문에서는 CINR 을 분석하고 미래의 값을 예측하는 방법에 대해 연구한다. 본 논문의 분석을 통해 CINR 의 예측을 비교적 정확하게 수행할 수 있다면 앞으로의 가용 대역폭을 비교적 정확하게 예측할 수 있고 효율적인 비디오 스트리밍 시스템을 제안할 수 있다.

• International Journal of Control, Automation, and Systems
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• v.5 no.5
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• pp.584-593
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• 2007

BACnet is a standard data communication protocol especially designed for building automation and control networks. BACnet uses the Master-Slave/Token-Passing (MS/TP) protocol as one of its field-level networks. In this study, the performance of the BACnet MS/TP protocol is evaluated using an experimental model. The network performance is investigated and evaluated in terms of token rotation time and round trip time. The experimental results show that the performance of the MS/TP network is heavily influenced by the network traffic load, the assignment of MS/TP node address and the network configuration parameter called $N_{max_info-frames}$. Based on the experimental results, simple practical designing guidelines for BACnet MS/TP network systems are also proposed.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.6
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• pp.1147-1165
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• 2011

We present a simple and cost effective rate control scheme for streaming video over a wireless channel by using the information of mobile devices' buffer level. To prevent buffer fullness and emptiness at receivers, the server should be able to adjust sending rate according to receivers' buffer status. We propose methods to adjust sending rate based on the buffer level and discrete derivative of the buffer occupancy. To be compatible with existing network protocols, we provide methods to adjust sending rate by changing the inter-packet delay (IPD) at the server side. At every round-trip time, adjustments of sending rate are made in order to achieve responsiveness to sudden changes of buffer availabilities. A series of simulations and the prototype system showed that the proposed methods did not cause buffer overflows and it can maintain smoother rate control and react to bandwidth changes promptly.

• Proceedings of the Korean Information Science Society Conference
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• 2004.04a
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• pp.781-783
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• 2004

최근 종단 호스트에서 나타나는 데이터 전송의 병목현상이 두드러지면서 종단 호스트에 관한 많은 연구가 진행되고 있다. 기존의 연구를 통해 종단 호스트의 병목현상의 원인으로 TCP의 비효율적인 버퍼관리가 제기되었다. 특히 임베디드 환경에서는 시스템 자원이 제한적일 수밖에 없기 때문에 TCP의 버퍼관리의 효율성이 매우 중요하다. 본 논문에서는 새로운 TCP 버퍼할당 기법인 RTBA(Rate-Based TCP Buffer Allocation) 기법을 제안하였다. RTBA는 플로우의 RTT(Round Trip Time), RTO(Retransmit Time Out), 패킷 손실율 둥을 고려한 TCP의 기대전송율을 기반으로 버퍼를 동적으로 할당함으로써 상대적으로 적은 버퍼를 사용하는 임베디드 환경에서도 다수의 TCP 플로우들이 높은 전송성능을 얻게 한다. 실험을 통해 제안한 RTBA 기법이 기존의 고정적인 버퍼할당 기법에 비해 우수한 성능을 보임을 확인할 수 있었다.

• Journal of Communications and Networks
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• v.15 no.6
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• pp.635-644
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• 2013

Adaptive hypertext transfer protocol (HTTP) streaming has become a new trend to support adaptivity in video delivery. An HTTP streaming client needs to estimate exactly resource availability and resource demand. In this paper, we focus on the most important resource which is bandwidth. A new and general formulation for throughput estimation is presented taking into account previous values of instant throughput and round trip time. Besides, we introduce for the first time the use of bitrate estimation in HTTP streaming. The experiments show that our approach can effectively cope with drastic changes in connection throughput and video bitrate.

• Proceedings of the Korean Information Science Society Conference
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• 2001.04a
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• pp.292-294
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• 2001

VoIP(Voice over IP) system은 현재 크게 2가지 형태로 진행되어가고 있다. 첫 번째는 H.323을 이용한 방법이고, 두 번째는 SIP(Session Initiation Protocol)를 이용한 방법이다. H.323은 실제 데이터를 전송하기전 호처리에 많은 signaling이 이루어지는 관계로 SIP보다 많은 RTT(Round Trip Time)를 소모하게 된다. 따라서 매우 복잡하고, LAN환경을 바탕으로 만들어서 확장성면에서도 여러 문제점을 가지고 있다. 그래서 본 논문은 호처리는 SIP를 이용하고, 실제 음성전송은 RTP(Real-Time Transport Protocol)와 RTCP(RTP Control Protocol)를 이용하는 시스템 구현을 제시한다. RTP는 실시간 특성을 가지는 데이터에 대해서 종단간 전송 서비스를 제공해주는 프로토콜로, 어떠한 인코딩에도 적합한 프레임워크를 제공한다. 그런데, RTP는 완전한 하나의 프로토콜이 되기 위해서는 RTP와 페이로드 포맷이 함께 제공되어야 하므로, 구현시스템은 음성신호를 PCM(Pulse Code Modulation), ADPCM(Adaptive Differential PCM)등의 여러 압축기술을 이용하여 파일을 생성하여 실시간으로 RTP와 RTCP를 이용하여 전송하는 방법을 제시한다.