• The KIPS Transactions:PartC
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• v.12C no.1 s.97
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• pp.121-128
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• 2005

Tho existing TCP(Transmission Control Protocol) is known to be unsuitable for a network with the characteristics of high RDP(Bandwidth-Delay Product) because of the fixed small or large buffer size at the TCP sender and receiver. Thus, some trial cases of adjusting the buffer sizes automatically with respect to network condition have been proposed to improve the end-to-end TCP throughput. ATBT(Automatic TCP fluffer Tuning) attempts to assure the buffer size of TCP sender according to its current congestion window size but the ATBT assumes that the buffer size of TCP receiver is maximum value that operating system defines. In DRS(Dynamic Right Sizing), by estimating the TCP arrival data of two times the amount TCP data received previously, the TCP receiver simply reserves the buffer size for the next arrival, accordingly. However, we do not need to reserve exactly two times of buffer size because of the possibility of TCP segment loss. We propose an efficient TCP buffer tuning technique(called TBT-PLR: TCP buffer tuning algorithm based on packet loss ratio) since we adopt the ATBT mechanism and the TBT-PLR mechanism for the TCP sender and the TCP receiver, respectively. For the purpose of testing the actual TCP performance, we implemented our TBT-PLR by modifying the linux kernel version 2.4.18 and evaluated the TCP performance by comparing TBT-PLR with the TCP schemes of the fixed buffer size. As a result, more balanced usage among TCP connections was obtained.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.6C
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• pp.614-622
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• 2006

In this paper, we propose an efficient packetization method to reduce the packetization overhead. For the purpose, we firstly verify the relationship between packet length and packet loss rate. The empirical results show that as the packet length is larger than the path MTU, the packet loss rate is drastically increased, producing poor visual quality at the receiver side. However, as the length of the packet is reduced, we should transmit more packets per frame and the packetization overhead will be increased. This increase in the packetization overhead reduces the number of bits allocated to the video data, resulting in the low visual quality. Therefore, each packet should be packetized to have the packet length close to the path MTU. In this paper, we show that the this process of the packetization with the constraint on the packet length is very similar to the dynamic storage allocation in the operating system. We had thoroughly surveyed the dynamic storage allocation methods used in the recent operating systems and propose to use the allocation methods for the video packetization. We empirically show that the proposed method can reduce the packetization overhead upto 28.3%, compared with the conventional sequential packetization method which have been widely used in Internet video transmission.

• Journal of Internet Computing and Services
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• v.10 no.2
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• pp.171-177
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• 2009

Optical packet switching(OPS) is a strong candidate for the next-generation internet, since it has a fine switching granularity at the packet level for providing flexible bandwidth, and provides seamless integration between WDM layer and IP layer. Optical packet switching have been studied in two categories: OPS in synchronous and OPS in asynchronous networks. In this article we are focused on contention resolution of OPS in asynchronous networks. The hybrid buffer have been addressed, to reduce packet loss further as one of the alternative buffer structures for contention resolution of asynchronous and variable length packets, which consists of the FDL buffer and the electronic buffer. The OPS design issue for the limited number of TWCs and internal wavelengths is important in the aspect of switch cost and resource efficiency. Therefore, an hybrid buffer structured optical packet switch and its scheduling algorithm is presented for considering the limited number of TWCs and internal wavelengths, for contention resolution of asynchronous and variable length packets. The proposed algorithm could lead to the packet loss improvement compared to the legacy LAUC-VF algorithm with only the FDL buffer.

• Proceedings of the IEEK Conference
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• 2001.06c
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• pp.93-96
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• 2001

In this paper, we propose a delay-constrained retransmission method to control packet error or loss in common internet. The Delay Regulator(or Jitter Buffer) which is used to control errors caused by unreliable UDP connection, stores received data packets fDr a small amount of time to prevent network jitter from affecting display quality, which causes constant delay. In this paper, we propose a retransmission method to increase efficiency of ARQ(Automatic Repeat reQuest) by using characteristic of delay regulator.

• Journal of Information Processing Systems
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• v.2 no.1
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• pp.34-38
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• 2006

In the wireless network, TCP performs poorly because it was originally designed for wired networks and does not take into consideration wireless characteristics such as mobility, high-loss probability, and hidden-terminal problems. In particular, in the wireless multi-hop networks, a large congestion window increases the probability of contention and packet losses, and TCP performance is degraded severely as a result. So, it is necessary to limit the TCP congestion window size in order keep the probability of contention loss in the system to a minimum. In this paper, we propose a new scheme for determining the maximum congestion window size based on the measured bandwidth and Round-Trip-Time (RTT). Using ns-2 simulation, we show that the proposed scheme reduces the probability of packet contention and improves TCP performance.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.10
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• pp.2529-2547
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• 2012

In this paper, we use a quadrotor-based mobile sink to gather sensor data from the terrestrial deployed wireless sensor network. By analyzing the flight features of the mobile sink node, we theoretically study the flight constraints of height, velocity, and trajectory of the mobile sink node so as to communicate with the terrestrial wireless sensor network. Moreover, we analyze the data amount which the mobile sink can send when it satisfies these flight constraints. Based on these analysis results, we propose a data acquisition approach for the mobile sink node, which is discussed detailed in terms of network performance such as the transmission delay, packet loss rate, sojourning time and mobile trajectory when given the flying speed and height of the mobile sink node. Extensive simulation results validate the efficiency of the proposed scheme.

• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.549-551
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• 2004

This paper used OPNET to simulate video streaming service a test IP network and MPLS network with the traffic shaping that have with CQ_ LLQ algorithm, LSP of fixed bandwidth, policy of limitation users and measures parameters such as delay, throughput, packet loss. To verify the performance of video streaming service in IP network and MPLS network, two scenario that have same topology and traffic source. One is the simulation for best-effort service in pure IP network. The other is the simulation for QoS-enabled service in MPLS Network. Based on simulation result, the MPLS network with CQ_ LLQ algorithm and fixed LSP show advantage of the video streaming service QoS, specially delay and packet loss

• Proceedings of the IEEK Conference
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• 2003.07d
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• pp.1629-1632
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• 2003

Generally an increasing number of recent applications have found TCP too limiting. There are some characteristics in the transmission of document and binary data which some transmission delay are tolerant but the content must completely be transferred. However voice signals are more sensitive with not some packet loss but some transmission delay. Therefore, Stream Control Transmission Protocol(SCTP) is proposed to minimize the delay and packet loss in the field of delivery of voice signal. SCTP is designed to transport PSTN signalling messages over IP networks, but is capable of broader applications. In this paper, the architecture of SCTP implementation is designed and some interface of SCTP software library which are implemented are specified.

• Journal of the Korea Society of Computer and Information
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• v.10 no.6 s.38
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• pp.261-268
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• 2005

Recently, Many People have studied in Mobile Ipv6 to provide seamless service. FMIPv6 is designed to reduce the procedure of handover and will be used SIP, VoIP and Mobile Internet moving picture service. But, If mobile node moves fast, it will happen to packet loss and interruption problem in handover procedure. So, In this paper, designed and evaluated traffic control mechanism using WFQ scheduling to reduce the packet loss that occurred to handover procedure.

• Journal of the Korean Operations Research and Management Science Society
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• v.37 no.3
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• pp.69-78
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• 2012

Reduction of power consumption has been a major issue and an interesting challenge to maximize the lifetime of wireless sensor networks. We investigate the practical meaning of N-policy in queues as a power saving technique in a WSN. We consider the N-policy of a finite M/M/1 queue. We formulate the optimization problem of power consumption considering the packet loss probability. We analyze the trade-off between power consumption and the packet loss probability and demonstrate the operational characteristics of N-policy as a power saving technique in a WSN with various numerical examples.