• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.234-236
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• 2004

Transmission control protocol(TCP) is protocol used in internet. TCP is seldom transmission error and is protocol based on wire environment. TCP uses 3 way handshake ways, data transmission control through windows size, data transmission control through reception confirmation, sliding window for packet delivery. In this study, designed TCP packet ion module for analyze the TCP segments & correct information about TCP. TCP capture in internet using designed TCP module and analysed TCP segments composition. Through this, could analyze the correct information of protocol in network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.2B
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• pp.91-97
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• 2006

As increasing the network bandwidth, the threat of a network also increases with emerging various new services. For a high-performance network security, It is generally used that high-speed packet classification methods which employ hardware like TCAM. There needs an method using these devices efficiently because they are expensive and their capacity is not sufficient. In this paper, we propose an efficient packet classification using a Ternary-CAM(TCAM) which is widely used device for high-speed packet classification in which we have applied Snort rule set for the well-known intrusion detection system. In order to save the size of an expensive TCAM, we have eliminated duplicated IP addresses and port numbers in the rule according to the partitioning of a table in the TCAM, and we have represented negation and range rules with reduced TCAM size. We also keep advantages of low TCAM capacity consumption and reduce the number of TCAM lookups by decreasing the TCAM partitioning using combining port numbers. According to simulation results on our TCAM partitioning, the size of a TCAM can be reduced by upto 98$\%$ and the performance does not degrade significantly for high-speed packet classification with a large amount of rules.

• Journal of Advanced Navigation Technology
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• v.6 no.2
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• pp.158-167
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• 2002

In this paper, it is analyzed that WAP packet transmission time to improve performance of WAP using SAR-QT algorithm in Bluetooth channel. The order for SAR-QT algorithm to improve the transfer capability, it is fragmented in WTP total messages that are coming down from upper layer and then the packets are sent one at time in baseband. And it is studied that transmission time for WAP over Bluetooth according to DMI, DM3 or DM5 packet type using SAR-QT algorithm in Bluetooth piconet environment. This SAR-QT algorithm decreases WAP packet transmission time of L2CAP baseband packets by sending packet that are spanning multiple slots. From the results, in WAP over Bluetooth channels, it is found out that WTP packet size ought to be increased to decrease transmission time of WAP packet. In addition, considering BER in wireless channel, optimal WTP packet size is achieved for WAP over Bluetooth in a Rician fading environment.

• The KIPS Transactions:PartC
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• v.10C no.7
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• pp.937-942
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• 2003

In this paper, we measured and examined RTT delays and packet losses according to the changes of stationary loads for two typical stream-type traffics, a DV and a MPGE2 on the R＆D Gigabit Network testbed, JGN. As the result of our actual measurements, we realized that the packet size of stationary load have no effects on a DV and a MPGE2 stream on the very high-speed network(50Mbps, IP over ATM). When its bandwidth and stationary load exceeds 95％ of network bandwidth, packet losses appeared and RTT delay increased rapidly. Also we realized that the number and size of Receive ＆ Transmit buffer on the end systems have no effects on packet losses and RTT delays.

• Journal of Internet of Things and Convergence
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• v.6 no.3
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• pp.1-6
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• 2020

This paper aims to integrate the previous models about mean object transfer latency in one framework and analyze the result through the computational experience. The analytical object transfer latency model assumes the multiple packet losses and the Internet of Things(IoT) environment including multi-hop wireless network, where fast re-transmission is not possible due to small window. The model also considers the initial congestion window size and the multiple packet loss in one congestion window. Performance evaluation shows that the lower and upper bounds of the mean object transfer latency are almost the same when both transfer object size and packet loss rate are small. However, as packet loss rate increases, the size of the initial congestion window and the round-trip time affect the upper and lower bounds of the mean object transfer latency.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.9
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• pp.3159-3176
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• 2014

This work presents an analysis of the buffer features of an access router, especially the size, the impact on delay and the packet loss rate. In particular, we study how these features can affect the Quality of Service (QoS) of multimedia applications when generating traffic bursts in local networks. First, we show how in a typical SME (Small and Medium Enterprise) network in which several multimedia flows (VoIP, videoconferencing and video surveillance) share access, the upgrade of the bandwidth of the internal network may cause the appearance of a significant amount of packet loss caused by buffer overflow. Secondly, the study shows that the bursty nature of the traffic in some applications traffic (video surveillance) may impair their QoS and that of other services (VoIP and videoconferencing), especially when a certain number of bursts overlap. Various tests have been developed with the aim of characterizing the problems that may appear when network capacity is increased in these scenarios. In some cases, especially when applications generating bursty traffic are present, increasing the network speed may lead to a deterioration in the quality. It has been found that the cause of this quality degradation is buffer overflow, which depends on the bandwidth relationship between the access and the internal networks. Besides, it has been necessary to describe the packet loss distribution by means of a histogram since, although most of the communications present good QoS results, a few of them have worse outcomes. Finally, in order to complete the study we present the MOS results for VoIP calculated from the delay and packet loss rate.

• Proceedings of the Optical Society of Korea Conference
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• 2009.02a
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• pp.307-308
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• 2009

To reduce delay for high priority packets in the ring-based EPON, the EPON packet classifier groups services as their priorities and frames for services are dynamically framed as their priorities. Since dynamic framing for a packet priority dynamically changes assignment spaces in the maximum framing packet size as network traffics, it makes services with high priority to improve quality of services with relative low threshold time for transmitting.

• ETRI Journal
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• v.16 no.3
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• pp.27-41
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• 1994

This paper provides a performance study of multistage interconnection networks in packet switching environment. In comparison to earlier work, the model is more extensive - it includes several parameters such as multiple-packet messages, variable buffer size, and wait delay at a source. The model is also uniformly applied to several representative networks and thus provides a basis for fair comparison as well as selection of optimal values for parameters. The complexity of the model required use of simulation. However, a partial analytical model is provided to measure the congestion in a network.

• Proceedings of the IEEK Conference
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• 2003.07a
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• pp.477-480
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• 2003

Packet losses in the Internet can dramatically degrade quality of multimedia streams. Forward Error Correction (FEC) is one of the best methods that can protect data from packet erasures by means of sending additional redundant information. Proposed control algorithm provides the possibility of receiving real-time multimedia streams of given quality wifth minimal traffic overhead. The traffic optimization is reached by adjusting packet size as well as block code parameters. Calculations and simulation results show that for non-bursty network conditions traffic optimization can lead to more than 50% bandwidth reduction.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.11B
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• pp.953-959
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• 2006

This paper provides optimum TCP/IP packet size that maximizes the throughput efficiency of ATM layer as a function of TCP/IP packet length for several values of channel BER over wireless ATM LAN links applying data link error control schemes to reduce error problems encountered in using wireless links. For TCP/IP delay-insensitive traffc requiring reliable delivery, it is necessary to adopt data link layer ARQ protocol. So ARQ error control schemes considered in this paper include GBN ARQ, SR ARQ and type-I Hybrid ARQ, which ARQ is needed, but FEC can be used to reduce the number of retransmissions. Especially adaptive type-I Hybrid ARQ scheme is necessary for a variable channel condition to make the physical layer as SONET-like as possible.