• The Journal of the Korea institute of electronic communication sciences
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• v.14 no.4
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• pp.657-662
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• 2019

The power consumption of wearable sensors and electrocardiogram regulators should be very low to extend the network lifetime and anticipated QoS( : Quality of Service) control such as error correction and authentication of data processed by WBAN( : Wireless Body Area Network) nodes is important. Therefore, QoS control is the most urgent concern to implement WBAN in health monitoring regulations. For optimal QoS control, we compare the energy efficiency and the average number of transmissions with IEEE 802.15.6 and the error correction method considering energy efficiency. The performance of the proposed error correction technique shows that the energy efficiency and the transmission rate are improved by adjusting the coding rate appropriately using the channel estimation.

• Proceedings of the Korea Multimedia Society Conference
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• 2003.05b
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• pp.104-108
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• 2003

모바일에서의 TCP 데이터전송은 무선 고유의 특성인 높은 에러율로 인하여 TCP의 성능에 많은 장애를 가져 오고있다. 특히, 셀간 핸드오프 등으로 인해 TCP의 빈번한 혼잡제어가 발생하므로서 대역폭 저하가 발생할 수 있다. 본 논문에서는 대역폭 저하가 발생하는 것을 억제하기 위하여 모바일 노드가 높은 에러 발생지역에 위치한 경우 시그널링을 이용하여 패킷 전송을 중지시키는 방법으로 핸드오프 영역에서 효율적인 전송이 이루어지도록 함으로써 TCP의 성능을 향상시킬 수 있는 방법을 제안하였다.

• Proceedings of the Korean Information Science Society Conference
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• 2000.04a
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• pp.448-450
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• 2000

인터넷을 통한 실시간 멀티미디어 전송시 패킷손실에 따른 재생지연과 서비스 질이 낮아지는 문제가 발생한다. I-frame만을 전송하는 경우 패캣손실에 따른 에러전파를 제거할 수 있지만 네트워크의 대역폭이 작은 경우에는 압축율을 높여 I-frame을 전송하기 때문에 압축과 정에서 데이터의 손실이 발생하게 된다. 본 논문에서는 압축된 비디오 데이터를 멀티캐스트와 유니캐스트를 동시에 사용하여 I-frame을 FEC방식으로 전송하는 프로토콜을 제안한다. 기존의 단일 프로토콜을 사용하는 방식과는 달리 유니캐스트기법을 동시에 사용하여 허용범위 내에서 수신자들의 에러제어를 가능하게 한다. 유니캐스트 연결을 선별적으로 흐름제어 하여 추가적인 작업 없이 우선순위별로 QoS를 보장한다.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.10
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• pp.1225-1236
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• 1999

기존의 오류 제어 방법들은 실시간 특성과 신축적인 신뢰성을 가지는 멀티미디어 통신 응용의 특징을 충분히 반영하지 못하고 있다. 본 연구는 기존의 재전송을 기반으로 하는 방법에 두 가지 기법을 추가하여 이 문제에 접근하였다. 첫째, 실시간 데이타의 마감 시간을 고려하여 미리 재전송을 실시하는 사전 재전송과 시간적으로 불가능하다고 판단될 때 복구를 포기하는 재전송 포기를 통한 적시 재전송 기법이다. 둘째, 요구되는 신뢰성에 근접하도록 오류가 많이 발생하였을 때만 재전송을 하는 선택적 오류 제어 기법이다. 본 연구는 두 기법을 적용한 오류 제어 프로토콜을 구현하고 네트워크 환경을 바꾸어 가며 성능을 검사하였다. 실험 결과, 제안된 오류 제어 방법은 전체적인 오류 제어 부하는 감소시키면서 오류 복구율을 높인다는 점을 확인하였다.Abstract The current error control schemes do not fit well to the characteristics of multimedia communications: real-time transmission and flexible reliability requirements. This research proposes a new error control scheme, which extends the retransmission-based error control with the following two mechanisms. First, error recovery is performed in two timely fashions: one is retransmission in advance where a retransmission is performed early enough for the deadline if an error is suspected, and the other is retransmission abort where a retransmission is given up if its recovery within the deadline seems to be impossible. Second, error control is selectively performed only when an actual error rate approaches to the threshold on a given reliability requirement. The proposed scheme has been implemented and experimented in various network environments. The performance results show that it has lower control overhead and higher error recovery than the existing schemes.

• The KIPS Transactions:PartC
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• v.13C no.5 s.108
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• pp.567-574
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• 2006

VG-AnyLAN is a local area network standard developed by the IEEE 802.12 project. While preserving the frame format of IEEE 802.3, VG-AnyLAN adopts a new medium access control called Demand Priority where transmission requests of stations are arbitrated by a control hub in a round-robin manner. Unlike CSMA/CD which is the medium access control of IEEE 802.3, the Demand Priority, while providing the maximum bound on the packet delay, does not put the limit on the network segment size. In this paper, we analyze the delay and the channel utilization performances of the medium access control of IEEE 802.12 VG-AnyLAN. We develope an analytic model of the system under assumptions that each station generates traffic of the equal priority and that the packets are of fixed length. Using the analytic model, we obtain the recursive expression of the average channel utilization and the average access delay The numerical results obtained via analysis are compared to the simulation results of the system for a partial validation of our analysis.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.10
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• pp.35-43
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• 2004

An arbitration method is proposed to resolve a collision and arbitrate beacon transmissions in an indoor Positioning system consisting of multiple beacons and listeners. Although two or more beacons may compete to transmit signals simultaneously, a single winner in the competition is determined autonomously through the arbitration process while they are transmitting. So, it can continue to send its data, but the others give up their transmissions during the arbitration process. As a consequence, update rate for location information and channel utilization can be improved by avoiding that all beacons fail due to a collision. Once a beacon succeeds in transmitting its signal, a low-level priority is assigned to it. And a high-level priority is allocated to a beacon which gave up its transmission during arbitration process. This will guarantee every beacon has fair transmission opportunity with the arbitration method. As no centralized control is required among beacons, a Positioning system can still be easily deployed and expanded with this arbitration method.

• Journal of Korea Multimedia Society
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• v.3 no.3
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• pp.272-279
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• 2000

When the multiple video sources are together transmitted through the channel of fixed bandwidth, the efficiently joint bandwidth allocation is necessary. This paper presents the joint bandwidth allocation method to keep the constant picture duality ratio among the video sources. We first find an approximated model of distortion and bitrate for the multiplexing system of multiple sources. Then we obtain the bitrate for each source to have constant distortion ratio among the sources by using the approximated model parameters for simple implementation. It is shown by simulation that the proposed bandwidth allocation can keep almost constant picture quality ratio among the sources in comparison to an independent bandwidth allocation method.

• Proceedings of the Korea Multimedia Society Conference
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• 2002.05d
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• pp.752-757
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• 2002

IP 멀티캐스트는 다자간 통신을 위한 효과적인 방법이지만 신뢰성 보장과 FTP나 Telnet등 TCP제공하는 다양한 전송프로토콜을 지원하지 않는다. 본 논문에서는 멀티캐스트의 UDP와 TCP계층의 전송 프로토콜을 동시에 사용할 수 있는 신뢰성 있는 멀티캐스트 서버라우터를 제안한다. 신뢰성 있는 멀티캐스트가 반드시 고려해야 할 확장성과 오류회복, 흐름제어를 위해, 그룹의 확장성과 오류회복에 적합한 기존의 SRM방식을 이용하였다. SRM방식은 신뢰성 있는 다-대-다 멀티캐스트 패킷을 전달한다. 각 참여자는 동시에 수신자와 송신자 역할을 할 수 있다. 모든 그룹 멤버는 낮은 주파수의 세션 메시지를 교환하고 그룹내의 주변 가입자에 대한 정보를 얻게 되며 패킷 손실 판단에 필요한 지연시간을 측정하고 복구하는 방식이다. UDP의 트래픽 기반은 멀티캐스트에 적합한 CBR과 SRM을 사용하였다. 본 연구에서는 멀티캐스트의 UDP 패킷과 TCP 패킷을 동시에 보낸 수신율과 멀티캐스트의 UDP만 보낸 수신율을 지연이 가장 큰 멀티캐스트 수신자측에서 패킷 번호와 수신율을 측정하고, 시뮬레이션을 통해서 검증을 통하여 기존의 멀티캐스트 전송방법과 비교하였다.

• Journal of Korea Society of Industrial Information Systems
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• v.10 no.3
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• pp.48-56
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• 2005

This paper proposes a dynamic end-to-end flow control algorithm that is more effective than previous methods considering either the Rate of Packet Loss (RPL) or Round Trip Time (RTT). When the RPL is under normal conditions, the current network status will be in one of three defined states by using the RTT and this makes bandwidth control precise before serious packet loss occurs. If the RPL exceeds a critical level, then the network is considered to be in a fourth state. Suitable transmission rates are determined depending on the network status and are controlled by adjusting not only the number of transmitted frames but also the quality of the frames. In this paper, we present some experimental results of the proposed algorithm. According to our quantitative analysis, the proposed method performs 1.6 to 6 times better than the previous method in terms of the RPL. At the same time, the total number of transmitted packets is increased, which indicates that the proposed method can provide greater bandwidth capacity than previous methods.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.8
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• pp.1505-1512
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• 2009

This paper propose a new frame level rate control algorithm for improving video quality and decreasing quality variation of an entire video sequence in a very low bit rate environment. In the proposed scheme, the allocated bits to a GOP are distributed to each frame properly according to the frame characteristics as well as the buffer status and the channel bandwidth. The H.264 standard uses various coding modes and optimization methods to improve the compression performance, which makes it difficult to control the generated traffic accurately. In this paper, proper prediction models for low bit rate environments are lust proposed, and a target distortion is determined using the models. According to the target distortion, the bit budget is allocated to each frame. It is shown by experimental results that the new algorithm can generate the PSNR performance better than that of the existing rate control algorithm.