• 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.

• 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.

• Journal of the Korea Society of Computer and Information
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• v.13 no.6
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• pp.233-241
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• 2008

The overlay multicast that has been recently presented as an alternative for the IP multicast needs scheduling method that can apply the system resource and the network bandwidth for the many users. We propose a efficient overlay multicasting network policy for multimedia services with composite stream service topology. And in order to evaluate the ability of the proposed scheme's performance, We test the performance of multi_casting nodes with RTT values, the bandwidth attrition rate for building the network topology and MOR effects with Overcast[5] The result of simulation shows improves about $8{\sim}20%$ of performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.12B
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• pp.1052-1062
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• 2004

Recent studies have pointed out that existing high-speed TCP protocols have a severe unfairness and TCP friendliness problem. As the congestion window achieved by a high-speed TCP connection can be quite large, there is a strong possibility that the sender will transmit a large burst of packets. As such, the current congestion control mechanisms of high-speed TCP can lead to bursty traffic flows in hi인 speed networks, with a negative impact on both TCP friendliness and RTT unfairness. The proposed solution to these problems is to evenly space the data sent into the network over an entire round-trip time. Accordingly, the current paper evaluates this approach with a high bandwidth-delay product network and shows that pacing offers better TCP friendliness and fairness without degrading the bandwidth scalability.

• 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.

• Journal of the Korea Society for Simulation
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• v.20 no.1
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• pp.61-67
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• 2011

Smart TV System will lead to drastic change of communication and media industries as one of the emerging next generation network services. However, when the number of concurrent users increases rapidly, the issue of service quality degradation occurs because providing services to many users simultaneously stresses both the server and the network. The server limitation can be circumvented by deploying server clusters. but the network limitation is far less easy to cope with, due to the difficulty in determining the cause and location of congestion and in provisioning extra resources. In order to alleviate these problems, a number of schemes have been developed. Prior works mostly focus on reducing user-centric performance metrics of individual connection, such as the round-trip time(RTT), downloading time or packet loss rate, but tend to ignore the network loads caused by the concurrent connections or global network load balance. In this work, we make an in-depth investigation on the issue of user grouping for massive Smart TV services through simulations on actual Internet test-bed, PlanetLab.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.6
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• pp.1083-1088
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• 2001

In this paper, we propose a new CBT(core-based tree)multicast routing Mechanism by load balancing mechanism. CBT may result in traffic concentration and bottlenecks near the core routers since traffic from all sources nodes the same set of links as approaches the core. So the router have to load packet to new core router for such congestion. Congestion information is given by rtt(round trip time) between designed core router and the other router.

• Proceedings of the Korea Information Processing Society Conference
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• 2017.11a
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• pp.232-234
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• 2017

사이버전 피해평가에 있어서 유무선 통합 환경에 대한 공격의 탐지와 이에 대한 평가가 필요한 상황이다. 특히 회사, 정부 및 군 시설 등에서 인가되지 않은 AP를 사용하여 공격이 발생하는 경우 각종 바이러스 및 해킹 공격에 의한 피해가 발생한 가능성이 높다. 띠라서 인가된 AP와 인가되지 않은 AP를 탐지해서 찾아 내야한다. 본 논문에서는 인가된 AP와 인가 되지 않은 AP를 탐지하기 위해 RTT(Round Trip Time)값을 데이터셋으로 만들고 각 기계학습 알고리즘 SVM(Support Vector Machine), J48(C4.5), KNN(K nearest neighbors), MLP(Multilayer Perceptron)의 결과를 비교해 성능의 차이를 밝히고 이를 통하여 공격을 탐지하여 피해평가에 연결이 되도록 한다.

• Bulletin of the Korean Space Science Society
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• 1993.04a
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• pp.7.1-7
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• 1993

위성의 정밀 거리 결정을 위해 1993년 9월 5일부터 IS일간 중국의 상해 천문대 Sheshan관측소와 장춘 인공위성 관측소에서 LAGEOS 11 (Laser Geodynamics Satellite II)에 대한 SLR (Satellite Laser Ranging) 관측을 수행하였다. SLR 관측에서는 지상의 관측소에서 발사한 LASER 펄스 (pulse)가 반사경들(retroflectors)로 둘러싸인 인공위성에 반사되어 돌아오는 RTT (Round Trip Time)를 측정하여 위성까지의 거리를 결정하는데, 관측된 시간과 거리 자료는 많은 잡음(noise)를 포함하고 있기 때문에 정확한 자료를 얻기 위해서는 많은 보정이 필요하다. 관측된 시간, 거리 자료를 지상 목표물 조준(ground target ranging )에 의한 system보정, 원자시계와 GPS에서 수신된 시간과의 시간 비교, 측정된 온도, 기압, 상대 습도에 따른 대기 영향의 보정 등을 통해 오차를 줄이고 다시 LASERF beam의 대기 굴절에 따른 거리 변화 보정, 위성의 질량 중심 거리(offset) 보정, 조석력에 의한 변화값 보정, 전자기적 지연(electromagnetic delay)에 의한 상대론적 보정등을 통해서 정밀한 거리 자료를 얻었다.