• Proceedings of the Korean Information Science Society Conference
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• 2008.06d
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• pp.272-275
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• 2008

종단에서 네트워크 용량에 대한 정보는 사용자뿐 아니라 전송 계층 프로토콜 상에서 전송량 등을 조절하기 위한 중요한 용도로 활용된다. 이러한 네트워크 용량(또는 대역폭)을 종단에서 측정하기 위해 활용되는 방법 중 하나가 바로 패킷 페어 기법이다. 패킷 페어 기법은 계산이 간단하고 빠른 측정이 가능하다는 장점이 있으나, 각 프로빙 패킷에 대한 즉각적인 응답 패킷을 필요로 하기 때문에 지연 ACK 방식이 존재하는 TCP 상에서는 적용하기 어렵다는 단점이 있다. 본 논문에서는 이러한 지연 ACK의 효과를 그대로 유지하면서 네트워크 용량의 추정이 가능한 지연 패킷 페어 기법을 제안한다. 시뮬레이션 결과를 통해 제안된 지연 패킷 페어 기법이 기존 방식의 정확성, 빠른 측정 시간 등의 장점을 보존하면서 지연 ACK의 사용 여부에 관계없이 정확히 동작할 수 있음을 보였다.

• Proceedings of the Korean Information Science Society Conference
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• 2000.10c
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• pp.207-209
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• 2000

인터넷전화 서비스는 저렴한 가격과, 타 서비스와 통합 및 가치부가(Value Added)면에서 기존의 전화에 비해 많은 장점을 가지고 있으나, 상대적으로 낮은 음질로 인하여 사용자의 요구를 만족시키지 못하고 있다. 이것은 현재 인터넷은 best-effort형 패킷 전달 서비스만을 제공하고 있기 때문에 전송지연, 패킷손실, 지터등을 보장할 수 있는 방법이 없기 때문이다. 본 논문에서는 인터넷전화에서 패킷손실이나 전송지연으로 인한 음질 저하문제를 SOLA 알고리즘을 이용해 보완하였다. 제시된 알고리즘에서는 송신측에서 패킷을 전송하면 수신측에서는 수신 패킷에 SOLA 알고리즘을 적용하여 수신 패킷을 사람이 인지하지 못하는 수준에서 확장하여 전송지연으로 인한 패킷손실을 감소시킨다. 시뮬레이션 결과 전송지연으로 인한 패킷 손실 확률이 상당히 감소되었고 음질 또한 상당히 개선되었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.11C
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• pp.123-132
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• 2001

This paper proposes a sound multicast packet-switching method which can less affect QoS(Quality of Service) degradation. The method includes a switch fabric with extra switching paths dedicated f()r multicast packets. Presented also are both a buffering structure and a scheduling algorithm for the proposed method. Simulation analysis for the method shows that the switching delay of unicast packets is decreased even though arrival rate of multicast packets is increased.

• Journal of the Korea Society of Computer and Information
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• v.10 no.6 s.38
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• pp.227-232
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• 2005

A network of Packet based switch Mode will be required to carry the traffics(CVR,VBR, UBR, ABR) generated by a wide range of services. Packet based Network services the quality-of-Service (QoS) management of traffic sources and bandwidth. Besides efficiency and throughput services are achieved in the multimedia traffic sent in the network. In this paper. the scheduler transmits the safe packet, drop the unsafe packet and evaluate unsafe packet as the requirement of the delay avoiding the network congestion for improving the QoS of the multimedia network In this paper. we Propose the scheduling algorithm which evaluates and drops the packet . The suggested model performance of the switch is estimated and simulated in terms of the delay by computer.

• Journal of the Korea Society of Computer and Information
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• v.10 no.5 s.37
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• pp.245-250
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• 2005

A network of Packet based switch Mode will be required to carry the traffics(CVR,VBR, UBR, ABR) generated by a wide range of services. Packet based Network services the Qualify-of-Service (QoS) management of traffic sources and bandwidth. Besides efficiency and throughput services are achieved in the multimedia traffic sent in the network. In this paper, the scheduler transmits the safe packet, drop the unsafe packet and evaluate unsafe Packet as the requirement of the delay avoiding the network congestion for improving the QoS of the multimedia network. In this paper, we Propose the scheduling algorithm which evaluates and drops the packet The suggested model Performance of the switch is estimated and simulated in terms of the delay by computer.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.1
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• pp.205-214
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• 2008

To transfer large amount of packets fast in sensor network, it is necessary that the delay between successive packet transmissions should be minimized as possible. In Sensor network, since the Operating Systems are worked on the event driven, the Timer Event is used to transfer packets successively. However, since the transferring time of packet completely is varies very much, it is very hard to set appropriate interval. If interval is too long, delay also becomes too long but if interval is too short, the fail of transfer request would increase. In this paper, we propose ESTEO which reduces the delay between successive packet transmissions by using SendDone Event which informs that a packet transmission has been completed. In ESTEO, the delay between successive packet transmissions is shortened very much since the transmission of next Packet starts at the time when the transmission of previous packet has completed, irrespective of the transmission time. Therefore ESTEO could provide high packet transmission rate given large amount of packets.

• Journal of the Institute of Electronics and Information Engineers
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• v.49 no.11
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• pp.9-16
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• 2012

In this paper, we focus on end-to-end transfer delay improvement by using network coding and propose a scheme where a message is divided into several packets which are network coded generating additional redundancy with the results that the number of transmitted packets increases. In networks with high coefficient of variation of transfer time, increased number of packets could reduce the transfer time of the message to a destination. For the proposed scheme, we investigate the optimum number of divided packets and redundancy considering transfer delay reduction and additional transmission cost caused by using network coding under the restriction of maximum transmission packet size. From the results of the investigation, we see that the proposed scheme is effective in networks having high variability of transfer time and would be very useful and practical especially for the case that expedited deliveries of messages are needed.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.10
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• pp.2191-2204
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• 1997

An efficient packet transmit scheduling algorithm should be able to allocate the resource to each connection fairly based on the bandwidth and quality of service requirements negotiated during the call admission procedure and it should be able to isolate the effects of users that are behaving badly. In this paper, we propose an effective round-robin packet transmit scheduling mechanism, which we call the delay tolerant packet reserving scheme (DTPRS) based on delay QoS requirments. The proposed scheme can not only provide fairness and but also reduce delay, delay variation, and packet loss rate by reserving output link time slots of delay tolerant packets and assigning the reserved slotsto delay urgent packets. Our scheme is applicable to high speed networks including ATM network because it only requires O(1) work to process a packet, and is simple enough to implement.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.1
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• pp.20-33
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• 2002

Usage of ATM AAL2 packets becomes dominant to increase transmission efficiency of voice traffic in the backbone network. In case of voice service that uses AAL2 mechanism, if resources of network are enough, connection of new call is accepted. However, due to packets generated by the new call, transmission delay of packets from old calls can increase sharply. To control this behavior, in this paper we present an AAL2 buffer management scheme that allocates a virtual buffer to each call and after calculating its propagation delay variation(PDV), decides to drop packets coming from each call according to the PDV value. We show that this packet dropping algorithm can effectively prevent abrupt QoS degradation of old calls. To do this, we analyze AAL2 packet composition process to find a critical factor in the process that influences the end-to-end delay behavior and model the process by K-policy M/D/1 queueing system and MIN(K, Tc)-policy M/D/1 queueing system. From the mathematical model, we derive the probability generating function of AAL2 packets in the buffer and mean waiting time of packets in the AAL2 buffer. Analytical results show that the AAL2 packet dropping algorithm can provide stable AAL2 packetization delay and ATM cell generation time even if the number of voice sources increases dramatically. Finally we compare the analytical result to simulation data obtained by using the COMNET Ⅲ package.

• Proceedings of the KAIS Fall Conference
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• 2011.05a
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• pp.207-210
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• 2011

무선으로 근거리 디바이스들을 연결하는 무선 랜에서는 CSMA/CA를 기반으로 하는 IEEE 802.11이 대표적인 프로토콜로 사용되고 있다. IEEE 802.11 무선망에서 지연에 민감한 실시간 멀티미디어 응용 서비스들의 요구가 증가함에 따라 MAC 계층에서의 지연시간에 대한 연구는 중요하다. IEEE 802.11 무선랜의 기본적인 액세스 방법으로 사용하는 DCF는 경쟁 스테이션이 적은 상황에서는 비교적 우수한 성능을 보이나 경쟁 스테이션의 수가 많은 경우 처리율, 패킷지연 관점에서 성능이 저하되는 문제점이 있다. 본 논문에서는 IEEE 802.11 무선 랜 환경에서 전송에 성공한 패킷의 평균지연시간과 전송에 실패하여 패킷이 폐기되기까지의 평균시간에 대해 제안된 방식들의 성능을 분석하고 시뮬레이션을 통해 이를 입증한다.