• 대한전자공학회:학술대회논문집
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• 대한전자공학회 1999년도 추계종합학술대회 논문집
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• pp.121-124
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• 1999

In a packet-switched wireless cellular network, a packet destined to a mobile station is queued at a base station and then broadcast over the base station's cell. When an active mobile station leaves a cell, there re-main packets which are destined to the mobile and not yet delivered to it at the cell's base station. For application which are sensitive to packet losses, such back- logged packets must be forwarded to the new base station. Otherwise, an end-to-end retransmission may be required. However, an increase in packet delay is incurred by employing the packet forward scheme, since a packet may be forwarded many times before it is delivered to the destined mobile station. For an enhanced quality-of-service level, it is preferred to reduce tile packet delay time. In this paper, we develop an analytical approximation method for deriving mean packet delay times. Using the approximation and simulation methods, we investigate the effect of network parameters on the packet delay time.

• 대한산업공학회지
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• 제28권2호
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• pp.139-146
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• 2002

The third generation mobile communication needs to provide multimedia service with increased data rates. Thus an efficient allocation of radio and network resources is very important. This paper models the 'burst switching' as an efficient radio resource allocation scheme and the performance is compared to the circuit and packet switching. In burst switching, radio resource is allocated to a call for the duration of data bursts rather than an entire session or a single packet as in the case of circuit and packet switching. After a stream of data burst, if a packet does not arrive during timer2 value ($\tau_{2}$), the channel of physical layer is released and the call stays in suspended state. Again if a packet does not arrive for timerl value ($\tau_{1}$) in the suspended state, the upper layer is also released. Thus the two timer values to minimize the sum of access delay and queuing delay need to be determined. In this paper, we focus on the decision of $\tau_{2}$ which minimizes the access and queueing delay with the assumption that traffic arrivals follow Poison process. The simulation, however, is performed with Pareto distribution which well describes the bursty traffic. The computational results show that the delay and the packet loss probability by the burst switching is dramatically reduced compared to the packet switching.

• Journal of Communications and Networks
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• 제16권5호
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• pp.548-558
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• 2014

Load distribution is vital to the performance of multipath transport. The task becomes more challenging in real-time multimedia applications (RTMA), which impose stringent delay requirements. Two key issues to be addressed are: 1) How to minimize end-to-end delay and 2) how to alleviate packet reordering that incurs additional recovery time at the receiver. In this paper, we propose sub-packet based multipath load distribution (SPMLD), a new model that splits traffic at the granularity of sub-packet. Our SPMLD model aims to minimize total packet delay by effectively aggregating multiple parallel paths as a single virtual path. First, we formulate the packet splitting over multiple paths as a constrained optimization problem and derive its solution based on progressive approximation method. Second, in the solution, we analyze queuing delay by introducing D/M/1 model and obtain the expression of dynamic packet splitting ratio for each path. Third, in order to describe SPMLD's scheduling policy, we propose two distributed algorithms respectively implemented in the source and destination nodes. We evaluate the performance of SPMLD through extensive simulations in QualNet using real-time H.264 video streaming. Experimental results demonstrate that: SPMLD outperforms previous flow and packet based load distribution models in terms of video peak signal-to-noise ratio, total packet delay, end-to-end delay, and risk of packet reordering. Besides, SPMLD's extra overhead is tiny compared to the input video streaming.

• 한국통신학회논문지
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• 제13권2호
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• pp.171-181
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• 1988

본 논문에서는 packet-switched network에서 음성을 서비스하는데 있어서 고려해야 할 여러가지 점들을 살펴보고, 실제로 음성과 데이터를 동시에 서비스하는 packet voice/data terminal을 구현하였으며 그 성능 분석을 시도하였다. PVDT의 software는 OSI 7 layer architecture에 맞추어 설계하였으며 음성과 데이터를 link level부터 구별하여 서비스하였다. 또한 음성 packet의 전송 delay를 작게 하기 위해 데이터보다 음성을 우선적으로 서비스하도록 하였으며 간략화된 protocol로 재전송에 의한 overhead를 없앴다. PVDT의 hardware의 구성은 기능별로 master control module, speech processing module, speech activity detection module, telelphone interface module, input/output inteface module로 나누어진다. Packet음성통신망에 대한 해석으로는 음성 packet의 전송 delay의 variance에 의한 영향을 줄이기 위한 최적 재생지연시간을 전송 delay의 분포를 통해 계산하였다.

• 인터넷정보학회논문지
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• 제18권6호
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• pp.65-73
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• 2017

Congestion occurring at wireless sensor networks(WSNs) causes packet delay and packet drop, which directly affects overall QoS(Quality of Service) parameters of network. Network congestion is critical when important data is to be transmitted through network. Thus, it is significantly important to effectively control the congestion. In this paper, new mechanism to guarantee reliable transmission for the important data is proposed by considering the importance of packet, configuring packet priority and utilizing the settings in routing process. Using this mechanism, network condition can be maintained without congestion in a way of making packet routed through various routes. Additionally, congestion control using packet service time, packet inter-arrival time and buffer utilization enables to reduce packet delay and prevent packet drop. Performance for the proposed mechanism was evaluated by simulation. The simulation results indicate that the proposed mechanism results to reduction of packet delay and produces positive influence in terms of packet loss rate and network lifetime. It implies that the proposed mechanism contributes to maintaining the network condition to be efficient.

• 한국통신학회논문지
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• 제15권2호
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• pp.98-111
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• 1990

본 논문에서는 CCITT에서 권고한 ISDN D-채널 access protocol의 queueing 모델을 제시하였으며 그 모델을 이용하여 signalling message와 packet message의 delay의 Laplace transform을 구하였다. 그 분석결과는 simulation을 통하여 검증하였으며, 또한 D-채널 access 시스템에서의 packet 및 signalling message 의 delay 특성에 대하여 비교 설명하였다.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제9권6호
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• pp.2180-2200
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• 2015

This paper considers mobility control to improve packet delivery in delay-tolerant networks (DTNs) under group mobility. Based on the group structure in group mobility, we propose two mobility control techniques; group formation enforcement and group purposeful movement. Both techniques can be used to increase the contact opportunities between groups by extending the group's reachability. In addition, they can be easily integrated into some existing DTN routing schemes under group mobility to effectively expedite the packet delivery. This paper is divided into 2 parts. First, we study how our proposed mobility control schemes reduce the packet delivery delay in DTNs by integrating them into one simple routing scheme called group-epidemic routing (G-ER). For each scheme, we analytically derive the cumulative density function of the packet delivery delay to show how it can effectively reduce the packet delivery delay. Then, based on our second proposed technique, the group purposeful movement, we design a new DTN routing scheme, called purposeful movement assisted routing (PMAR), to further reduce the packet delay. Extensive simulations in NS2 have been conducted to show the significant improvement of PMAR over G-ER under different practical network conditions.

• 한국통신학회논문지
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• 제30권10A호
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• pp.896-905
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• 2005

본 논문에서는 N-채널 SAW(Stop and Wait) ARQ 재전송 기법을 채택하고 있는 HSDPA 시스템에서 HS-DSCH 채널의 성능 향상을 목적으로 선택적 패킷 지연(SPD :Selective Packet Delay) 기법을 제안한다. SPD 기법은 무선 채널 상태에 따라 적응적으로 패킷 전송을 제어하는 방법으로 채널 상태가 나쁜 경우에는 해당 패킷 전송을 일정시간 동안 강제 지연하고 대신에 해당 타임 슬롯을 채널 상태가 양호한 다른 사용자에게 할당해 주는 기법이다. 따라서 제안하는 기법은 버스트 에러 환경에서 연속적인 패킷 손실로 인해 발생하게 되는 긴 재전송 지연을 효과적으로 줄일 수 있는 특징을 가진다. 또한 SPD 기법에 효율적으로 적용될 수 있는 SPD-LDPF(Long Delayed Packet First)와 SPD-DCRR(Deficit Compensated Round Robin)의 두 종류의 패킷 스케줄링 기법을 제안한다. 시뮬레이션 결과, 제안한 기법은 패킷 평균지연 특성, 처리량 성능, 그리고 사용자간 공평성에서 모두 우수한 성능을 보임을 확인하였다.

• 산업기술연구
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• 제27권B호
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• pp.235-243
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• 2007

Sensor network is used to obtain sensing data in various area. The interval to sense the events depends on the type of target application and the amounts of data generated by sensor nodes are not constant. Many applications exploit long sensing interval to enhance the life time of network but there are specific applications that requires very short interval to obtain fine-grained, high-precision sensing data. If the number of nodes in the network is increased and the interval to sense data is shortened, the amounts of generated data are greatly increased and this leads to increased amount of packets to transfer to the network. To transfer large amount of packets fast, 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. The longer the interval, the higher the delay and the shorter the delay, the larger the fail of transfer request. 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 timee. Therefore ESTEO could provide high packet transmission rate given large amount of packets.

• 정보처리학회논문지C
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• 제12C권3호
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• pp.437-442
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• 2005

인터넷 망의 종단간 혼잡상태를 유추할 수 있는 가장 일반적인 방법은 RTT (Round Trip Time)값을 측정하는 것이며, 이 방법으로 망의 흔잡 정도를 측정할 수 있다. 그러나 RTT는 패킷의 왕복 시간만을 측정하기 때문에 패킷의 송수신시 순방향과 역방향에서 어느 정도의 혼잡과 전송지연이 발생하였는가는 알 수가 없다. 본 논문에서는 패킷이 전송될 때 순방향/역방향 전송 지연을 계산하여, 망의 흔잡 상태를 정확하게 유추할 수 있는 새로운 알고리즘을 제시한다. 본 알고리즘에서는 여러 RTT 값들 중에서 가장 작은 RTT 값을 기준으로 하여 기준이 되는 순방향/역방향 전송 시간을 결정하고, 이 값과 각 패킷이 전송될 때 측정된 전송 시간을 비교하여 순방향/역방향 전송 지연 시간을 계산한다. 본 연구에서는 NS-2에서의 시뮬레이션과 실제 네트워크 상에의 측정을 통하여 제안된 방법의 올바른 동작을 확인하였다.