• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4A
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• pp.385-392
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• 2006

It is widely accepted that high user densities in mobile multimedia environments can only be achieved with micro- and pico-cell. The smaller cell size causes frequent handovers between cells, and decreases permissible handover processing delay. This may result in the handover failure, in addition to the loss of some packets during the handover. In these cases, re-transmission is needed in order to compensate errors, which in turn triggers a rapid degradation of throughput. In this paper, we propose a new handover scheme for the next generation mobile communication systems, in which the handover setup process is done in advance before a handover request by predicting the handover cell based on mobile terminal's current position and moving direction. Simulation is performed in order to analyze the handover failure rate and packet loss rate. Simulation results show that our proposed method provides a better performance than the conventional method.

• Journal of KIISE:Information Networking
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• v.29 no.4
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• pp.375-385
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• 2002

The IETF has defined the Intserv model and the RSVP signaling protocol to improve QoS capability for a set of newly emerging services including voice and video streams that require high transmission bandwidth and low delay. However, since the current Intserv model requires each router to maintain the states of each service flow, the complexity and the overhead for processing packets in each rioter drastically increase as the size of the network increases, giving rise to the scalability problem. This motivates our work; namely, we investigate and devise new control schemes to enhance the scalability of the Intesev model. To do this, we basically resort to the SCORE network model, extend it to fairly well adapt to the three services presented in the Intserv model, and devise schemes of the QoS scheduling, the admission control, and the edge and core node architectures. We also carry out the computer simulation by using ns-2 simulator to examine the performance of the proposed scheme in respects of the bandwidth allocation capability, the packet delay, and the packet delay variation. The results show that the proposed scheme meets the QoS requirements of the respective three services of Intserv model, thus we conclude that the proposed scheme enhances the scalability, while keeping the efficiency of the current Intserv model.

• 전자공학회논문지 IE
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• v.43 no.3
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• pp.41-48
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• 2006

In this paper, we propose a new active queue management (AQM) scheme by utilizing the predictability of the Internet traffic. The proposed scheme predicts future traffic input rate by using the auto-regressive (AR) time series model and determines the future congestion level by comparing the predicted input rate with the service rate. If the congestion is expected, the packet drop probability is dynamically adjusted to avoid the anticipated congestion level. Unlike the previous AQM schemes which use the queue length variation as the congestion measure, the proposed scheme uses the variation of the traffic input rate as the congestion measure. By predicting the network congestion level, the proposed scheme can adapt more rapidly to the changing network condition and stabilize the average queue length and its variation even if the traffic input level varies widely. Through ns-2 simulation study in varying network environments, we compare the performance among RED, Adaptive RED (ARED), REM, Predicted AQM (PAQM) and the proposed scheme in terms of average queue length and packet drop rate, and show that the proposed scheme is more adaptive to the varying network conditions and has shorter response time.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.11b
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• pp.1193-1196
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• 2002

본 연구는 접속 요구가 많은 웹 서버의 부하를 두개의 웹 서버로 분산시켜 보다 효율적인 웹 서비스를 제공함과 동시에, 두 웹 서버의 용도에 맞게 재전송 타임아웃(RTO:Retransmission Timeout)을 독립적으로 설정하여 네트워크의 효율성을 제고하기 위해 시도되었다. 이를 위해 Linux 기반의 PC에 라우터를 구축하고, 접속의 요구에 따라 두개 중 하나의 Web 서버에 접속되게끔 라우터를 설정하였다. 웹 서버는 내부 접속용과 외부 접속용으로 구분하였으며, 각각의 서버마다 재전송 타임아웃(RTO:Retransmission Timeout)값을 다르게 설정하여 네트워크에서의 지연(delay)을 최적화시켜 클라이언트의 요청에 대한 응답시간을 최소화시켰다. 전송거리가 짧은 내부 접속용 웹 서버에는 패킷 손실이 거의 없으므로 작은 RTO 값을 설정하고, 외부 접속용 웹 서버에는 큰 RTO 값을 설정하였는데 RTT(Round Trip Time:응답시간)와 Tput(Throughput:처리율)의 출력 결과를 통해 Web 서버의 성능 향상을 확인 할 수 있다.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10a
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• pp.862-864
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• 2003

DoS Attack (Denial-of-Service Attack)이란 공격자가 침입대상 시스템의 시스템 자원과 네트워크 자원을 악의적인 목적으로 소모시키기 위하여 대량의 패킷을 보냄으로써 정상 사용자로 하여금 시스템이 제공하는 서비스를 이용하지 못하도록 하는 공격을 의미한다. 현재 이러한 대부분의 DoS 공격은 인터넷 프로토콜 중 TCP 프로토콜을 주로 이용하고 있다. 이에 호스트의 연결요청이 발생하였을 경우 TCP 프로토콜을 기준으로 서버의 트래픽 변화율을 확인함으로써 DoS 공격을 분석하는 것이다. 서버의 트래픽 변화율은 호스트가 요청한 연결의 발생빈도에 따라 변화하는 서버의 연결요청 처리 시간 변화율을 확인하는 방법으로 확인할 수 있다. 이와 같은 방법으로 확인된 서버의 트래픽 변화율은 일정 시간동안의 변화율 증강을 모니터링 함으로써 DoS 공격에 참여하는 호스트가 요청하는 연결의 발생빈도를 간접적으로 확인하는데 이용할 수 있다. 따라서 본 논문에서는 호스트의 연결요청과 서버의 트래픽 변화율간 관계를 이용하여 DoS 공격의 특성을 분석하고자 한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2003.05b
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• pp.1369-1372
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• 2003

가까운 장래에 싼 가격의 블루투스가 여러 가지 이동, 휴대 장치에 내장되어 다양한 응용서비스를 제공할 것이다. 본 논문에서는 공항이나 쇼핑몰 등 사람들이 많이 몰리는 공공장소와 같은 핫스팟(hot spot) 지역에서 블루투스 기지국을 통한 음성서비스 제공을 위하여 다수의 기지국을 유서 LAN으로 협동시켜 각각 기지국별로 따로 제공되던 음성 채널들을 집합화(aggregation) 함으로써 보다 많은 음성 서비스 채널들을 동시에 수용할 수 있는 방법을 제시하고 성능을 분석하였다. 단독으로 동작하는 기지국은 ACL채널을 사용하는 경우 7개까지의 채널 수용이 가능하므로 이상적인 경우에서도 오버랩 영역의 기지국의 개수가 n이면 7n 만큼 채널의 증가가 가능한 반면 협동으로 동작하면 블록킹 율도 개선되면서 집합화된 채널들을 보간(interpolation) 기법을 통한 통계적 사용으로 음성 패킷의 허용 손실 범위 내에서 음성채널의 수를 약 14n 정도까지 확장할 수 있다. 또한 음성전송과 데이터 전송이 혼합 서비스 될 때 수용하는 음성 채널의 수에 따른 데이터 전송지연시간 특성을 분석하였다.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2006.05a
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• pp.729-732
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• 2006

This paper proposes and implements consistency algorithm that is serviced streaming data efficiently in VOD system. The media data is loaded into each node by Round Robin method. The barrier mechanism is changed into the minimum data factor(SH. GOP) of media data in this paper. In addition in order to fast media data service, the additional features are implemented in the consistency algorithm. Additional feature of the consistency algorithm is added. First, time synchronization algorithm is added the suggested consistency algorithm. Second, the prepaging mechanism supports efficient buffering service. I experimented the suggested consistency algorithm on two DSM systems.

• KIPS Transactions on Computer and Communication Systems
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• v.5 no.4
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• pp.87-94
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• 2016

This paper proposes an energy transmission method to maximize energy efficiency of a based station. This method makes use of classification of service type to solve an inefficient use of transmission power, which is from exponential relationship between the legacy data throughput and transmission power. The proposed one is a way to find the most energy-efficiency points with the transmitted optimal amount of data on users in a base station of wireless network environment. For this, we propose EETA (Energy-Efficient Transmission Algorithm) which can control the amount of data and the holding time at the base station. As a result, the proposed method can improve the energy efficiency of about 10% compared to the legacy base station.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.12 no.3
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• pp.96-102
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• 2013

In recent years, a variety of ITS services are available such as driving information, road conditions, V2X messages as well as navigation and traffic jams notification. The development of ITS services is accelerating by V2X communication technologies for high-speed vehicles. In this paper, WAVE communication devices based on the IEEE802.11p standard is introduced as a solution of V2X communication technologies. The H/W and S/W structures of the WAVE communication device and the characteristics of RF/antenna are described. The performance is evaluated in the test road by measuring throughput, PER and latency. The implemented WAVE communication device has 6~7 Mbps throughput with 10% PER at 1km coverage. The packet latency is less than 3ms for the whole test road. It is shown that the implemented WAVE technology is satisfactory to provide ITS services and Internet video-streaming services.