• Journal of KIISE:Information Networking
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• v.37 no.5
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• pp.363-373
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• 2010

The existing QoS mechanisms for video streaming are short of the consideration for various user environments and the characteristic of streaming applying programs. In order to overwhelm this problem, studies on the video streaming protocols exploiting scalable video coding (SVC), which provide spatial, temporal, and qualitative scalability in video coding, are progressing actively. However, these protocols also have the problem to deepen network congestion situation, and to lower fairness between other traffics, as they are not equipped with congestion control mechanisms. SVC based streaming protocols also have the problem to overlook the property of videos encoded in SVC, as the protocols transmit the streaming simply by extracting the bitstream which has the maximum bit rate within available bandwidth of a network. To solve these problems, this study suggests TCP-friendly network adaptive SVC streaming(T-NASS) protocol which considers both network status and SVC bitstream property. T-NASS protocol extracts the optimal SVC bitstream by calculating TCP-friendly transmission rate, and by perceiving the network status on the basis of packet loss rate and explicit congestion notification(ECN). Through the performance estimation using an ns-2 network simulator, this study identified T-NASS protocol extracts the optimal bitstream as it uses TCP-friendly transmission property and perceives the network status, and also identified the video image quality transmitted through T-NASS protocol is improved.

• 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.

• Proceedings of the Korean Information Science Society Conference
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• 1998.10a
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• pp.480-482
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• 1998

최근, 멀티캐스트 기법을 사용하는 멀티미디어 응용 프로그램들이 인터넷에 등장하고 있다. 이들 응용 프로그램들의 성공 여부는 수신자들에게 전송되는 음성/영상의 품질에 의해 좌우된다. 인터넷은 응용프로그램의 QoS(Quality of Service) 에 대한 요구를 보장할 수 없기 때문에 멀티케스트 트래픽(multicast traffic)을 위하여 인터넷의 성능을 최대한 효율적으로 이용할 수 있도록 흐름제어에 대한 많은 연구가 진행되고 있다. 그 중 IVS(INRIA Video conferencing System)에서 제안한 멀티캐스트 트래픽 흐름제어 알고리즘은 수신자가 주기적으로 전달하는 RTCP 의 패킷손실 정보에 의해 송신자가 전송율을 조절하는 것이다. 그러나 이 알고리즘은 네트워크 상태가 무부하(unload)임에도 불구하고 느린 피드백으로 인하여 가용 네트워크 대역폭을 빠르게 파악하지 못하기 때문에, TCP트래픽과 경쟁 상태에서 네트워크 대역폭을 불공정(unfairness)하게 사용하게 되고 네트워크 상태에 알맞는 전송율을 결정하지 못한다. 본 논문에서는 더욱 공정하게 대역폭을 공유할 수 있고 전체 링크 이용율을 높이는 두 가지 기법을 제안한다. 첫째, 측정된 네트워크 혼잡상태에 따라 RTCP 피드백의 전송 빈도를 동적으로 조절하는 것이다. 둘째, TCP와 같이 전송율을 증가/감소시킴으로써 공정하게 네트워크를 공유하도록 하는 것이다. 본 논문에서는 이 두 가지 기법들이 TCP 트래픽에 영향을 주지 않고 또한 RTCP피드백의 양을 증가시키지 않으면서도 공정하게 네트워크 대역폭을 공유함으로써 링크의 이용율을 높일 수 있다는 것을 시뮬레이션을 통하여 보여준다.안 모니터링 기 능 등으로 조사되었다.도 멜-켑스트럼을 사용한 경우 67.5%, K-L계수를 사용한 경우 75.3%로 7.8%의 향상된 인식률을 보였으며 K-L계수와 회귀계수를 결합한 경우에서도 비교적 높은 인식률을 보여 숫자음에 대해서도 K-L계수의 유효성을 확인할 수 있었다..rc$ 구입할 때 중점적으로 살펴보는 사항은 신선도와 순수재래종 여부, 위생상태였다. 한편 소비자가 언제나 구입할 수 없다는 의견이 85.2%나 되어 원활한 공급과 시장조성이 아직 정착되지 않고 있었다. $\bigcirc$ 현재 유통되고 있는 재래종닭은 소비자 대부분이 잡종으로 인식하고 있었으며, 재래종과 일반육계와의 구별은 깃털색, 피부색, 정강이색등 외관상으로 구별하고 있었다. 체중에 대한 반응은 너무 작다는 의견이었고, 식품으로의 인식도는 비교적 고급식품으로 인식하고 있다. $\bigcirc$ 재래종닭고기의 브랜드화에 대한 견해는 젊고 소득이 높은 계층에서 브랜드화의 필요성을 강조하고 있다. $\bigcirc$ 재래종달걀의 소비형태는 대부분의 소비자가 좋아하였으나 아직 먹어보지 못한 응답자가 많았다. 재래종달걀의 맛에 대해서는 고소하고 독특하여 차별성을 느끼고 있었다. $\bigcirc$ 재래종달걀의 구입장소는 계란판매점(축협.농협), 슈퍼, 백화점, 재래닭 사육 농장등 다양하였으며 포장단위는 10개를 가장 선호하였고, 포장재료는 종이, 플라스틱, 짚의 순으로 좋아하였다. $\bigcirc$ 달걀의 가격은 200원정도를 적정하다고 하였으며, 크기는 (평균 52g)는 가장 적당하다고

• The KIPS Transactions:PartC
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• v.19C no.3
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• pp.191-208
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• 2012

Proxy Mobile IPv6 (PMIPv6) is designed to provide network-based mobility management support to an MN without any involvement of the MN in the mobility related signalling, hence, the proxy mobility entity performs all related signalling on behalf of the MN. The new principal functional entities of PMIPv6 are the local mobility anchor (LMA) and the mobile access gateway (MAG). In PMIPv6, all the data traffic sent from the MN gets routed to the LMA through a tunnel between the LMA and the MAG, but it still has the single point of failure (SPOF) and bottleneck state of traffic. To solve these problems, various approaches directed towards PMIPv6 performance improvements such as route optimization proposed. But these approaches add additional signalling to support MN's mobility, which incurs extra network overhead and still has difficult to apply to multiple-LMA networks. In this paper, we propose a improved route optimization in PMIPv6-based multiple-LMA networks. All LMA connected to the proxy internetworking gateway (PIG), which performs inter-domain distributed mobility control. And, each MAG keeps the information of all LMA in PMIPv6 domain, so it is possible to perform fast route optimization. Therefore, it supports route optimization without any additional signalling because the LMA receives the state information of route optimization from PIG.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.9A
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• pp.737-748
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• 2005

Two flexible admission control schemes for integrated voice and stream-type data services are proposed in DS-CDMA systems. Most Previous studies on admission control have focused on integration of short, bursty Packet-type data services and conventional voice services. However, stream-type data services with a relatively long service holding time are expected to be a considerable portion of data traffic in future generation cellular systems. Scheme I is a basic scheme that accommodates both voice and data services with full bandwidth. However, voice services are given priority over data services using the duration difference between the holding times for these services. Scheme ll uses a different method to efficiently give priority to voice services over stream-type data services. An additional interference margin for voice services is provided by suppressing interference from stream-type data services according to voice access requests and a varying interference status. Performance of the two schemes is evaluated by developing Markovian models. Numerical results show that the voice capacity is highly sensitive to the service holding time of data services while the performance measures of data services are not highly sensitive. Scheme H is a significant improvement over Scheme I for accommodating voice and stream-type data services

• The KIPS Transactions:PartA
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• v.9A no.2
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• pp.181-190
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• 2002

IEEE1394 is a de facto standard for the home network interfaces of digital multimedia home devices including digital A/V systems, digital camcorders, and PCs. Recently, it has been used in applications to guarantee the real-time characteristics such as home automation system and IICP (Instrument and Industrial Control Protocol). In order to guarantee real-time requirements in these IEEE1394-based real-time applications, this thesis proposes the software architecture of an IEEE1394 based home network that supports the guarantee for service's react-time characteristics. The proposed architecture has a real-time IEEE1394 device driver and event service architecture for guarantee real-time characteristics. The real-time device driver supports priority-based queueing of packets and mechanism to reduce the interrupt latency time in ISR. The event service architecture supports a real-time events delivery based on home network service using real-time event channel. This architecture can accommodate the real-time requirements of various applications and services such as digital multimedia services with QoS guarantees. home automation system required real-tine characteristics.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.8
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• pp.1631-1636
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• 2015

With the advance of wireless communication technology in terrestrial area, underwater communication is also evolving very fast from a simple point-to-point transmission to an elaborate networked communications. Underwater acoustic channel has quite different features comparing with the terrestrial radio channel in terms of propagation delay, Doppler shift, multipath, and path loss. Thus, existing technologies developed for terrestrial communication might not work properly in underwater channel. Especially medium access control (MAC) protocols which highly depend on propagation phenomenon should be newly designed for underwater network. CSMA/CA has drawn lots of attention as a candidate of underwater MAC protocol, since it is able to resolve a packet collision and the hidden node problem. However, a received signal could be degraded by the interferences from the nodes locating outside the receiver's propagation radius. In this paper, we study the effects of interference on the CSMA/CA based underwater network. We derived the SNR with the interference using the sonar equation and analyzed the degradation of the RTS/CTS effects. These results are compared with the terrestrial results to understand the differences. Finally we summarized the design considerations in CSMA/CA based underwater network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.11B
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• pp.1133-1141
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• 2009

In this paper, we present a distributed fault-tolerant topology control protocol that configure a wireless sensor network to achieve k-connectivity and (k+1)-coverage. One fundamental issue in sensor networks is to maintain both sensing coverage and network connectivity in order to support different applications and environments, while some least active nodes are on duty. Topology control algorithms have been proposed to maintain network connectivity while improving energy efficiency and increasing network capacity. However, by reducing the number of links in the network, topology control algorithms actually decrease the degree of routing redundancy. Although the protocols for resolving such a problem while maintaining sensing coverage were proposed, they requires accurate location information to check the coverage, and most of active sensors in the constructed topology maintain 2k-connectivity when they keep k-coverage. We propose the fault-tolerant topology control protocol that is based on the theorem that k-connectivity implies (k+1)-coverage when the sensing range is at two times the transmission range. The proposed distributed algorithm does not need accurate location information, the complexity is O(1). We demonstrate the capability of the proposed protocol to provide guaranteed connectivity and coverage, through both geometric analysis and extensive simulation.

• The KIPS Transactions:PartC
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• v.10C no.7
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• pp.891-898
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• 2003

This paper analyzes and proposes a scheme that improves the performance of the RTP that is developed to support the end-to-end transmission function and QoS monitor function for real-time multimedia data transmission. Although the existing RTP module supports real-time transmission, it has some problems in guaranteeing QoS parameters. To solve this problem, we propose a new Selective Repeat Adaptive Rate Control (SRARC). The SRARC can support QoS by referring to the data transmission status from the client and then classifying the network status into three levels. It selectively transmits multimedia data and dynamically controls transmission rates based on such information as bandwidth, packet loss rate, and latency that can be calculated in data transfer phase. To verify the SRARC, we implement it in real local area networks and compare the QoS parameters of the SRARC with those of the SR and RTP By the experimental results, the SRARC shows better performance in the aspects of bandwidth usage rate, packet loss rates, and transmission delays than the existing RTP schemes.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.149-158
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• 2005

This paper proposes a congestion avoidance algorithm which provides stable throughput and transmission rate regardless of buffer size by limiting the TCP congestion window in fixed bandwidth networks. Additive Increase, Multiplicative Decrease (AIMD) is the most commonly used congestion control algorithm. But, the AIMD-based TCP congestion control method causes unnecessary packet losses and retransmissions from the congestion window increment for available bandwidth verification when used in fixed bandwidth networks. In addition, the saw tooth variation of TCP throughput is inappropriate to be adopted for the applications that require low bandwidth variation. We present an algorithm in which congestion window can be limited under appropriate circumstances to avoid congestion losses while still addressing fairness issues. The maximum congestion window is determined from delay information to avoid queueing at the bottleneck node, hence stabilizes the throughput and the transmission rate of the connection without buffer and window control process. Simulations have performed to verify compatibility, steady state throughput, steady state packet loss count, and the variance of congestion window. The proposed algorithm can be easily adopted to the sender and is easy to deploy avoiding changes in network routers and user programs. The proposed algorithm can be applied to enhance the performance of the high-speed access network which is one of the fixed bandwidth networks.