• Journal of the Institute of Electronics Engineers of Korea SD
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• v.41 no.7
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• pp.71-79
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• 2004

Packet-type data transmission is characterized by the continuous transmission of massive data with relatively constant rate. In such transmission, the dynamic power consumed on buses is influenced by the sequence of transmitted data. A new coding scheme called Sequence-Switch Coding (SSC) is proposed in this paper. SSC reduces the number of bus transitions in the transmission of packet-type data by changing the sending order of the data. Some simple algorithms are presented, In. The simulation results show that SSC outperforms the well-known Bus-Invert Coding with these algorithms. SSC is not a specific algerian but a method to reduce the number of bus-transitions. There could be lots of algorithms for realizing SSC. The variety of SSC algorithms provides circuit designers a wide range of trade-off between performance and circuit complexity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38B no.7
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• pp.553-563
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• 2013

When the IEEE 802.11 wireless LAN-based broadcasting services, we use broadcast packets to broadcast multimedia contents to a large number of users using limited wireless resources. However, broadcast transmission is difficult to recover the loss packets compared with unicast transmission. Therefore, analysis of packet loss characteristics is required to perform efficient packet recovery. The packet loss in wireless transmissions is often bursty with high loss data rate. Even if loss patterns have the same average packet loss, they are different in the recovery rate of random loss and burst loss depending on the nature. Therefore, the analysis and research of the nature of the loss are needed to recover loss packets considering bursty characteristics. In this paper, we experimented Wi-Fi broadcast transmission according to transmission rate and analyzed bursty characteristics of loss patterns using 4-state markov model.

• The KIPS Transactions:PartC
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• v.9C no.4
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• pp.563-572
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• 2002

Wireless mobile networks tend to drop a large portion of packets due to propagation errors rather than congestion. To Improve reliability over noisy wireless channels, wireless networks can employ forward error correction (FEC) techniques. Static FEC algorithms, however, can degrade the performance by poorly matching their overhead to the degree of the underlying channel error, especially when the channel path loss rate fluctuates widely. This paper investigates the benefits of an adaptable FEC mechanism for wireless networks with severe packet loss by analytical analysis or measurements over a real wireless network called sensor network. We show that our adaptive FEC named FECA (FEC-level Adaptation) technique improves the performance by dynamically tuning FEC strength to the current amount of wireless channel loss. We quantify these benefits through a hybrid simulation integrating packet-level simulation with bit-level details and validate that FECA keeps selecting the appropriate FEC-level for a constantly changing wireless channel.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.103-105
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• 2002

802.11과 같은 무선 네트워크에서는 전송오류에 의한 패킷손실이 많이 발생한다. 802.11 MAC 프로토콜에서는 에러 복구를 위해 ARQ방식을 통한 재전송을 통하여 에러를 정정하나 채널 에러 율이 증가하면 재전송 방식의 효율은 급격히 저하된다. 또한 재전송을하는데 있어서 다시 RTS와 CTS를 전송하여 데이터를 보낼 수 있는 채널을 확보해야 하므로 상당한 전송부하가 발생한다. 이에 재전송 없이 효율적인 에러 복구를 위해서는 FEC방식이 필요하다. 그러나 정적인 FEC방식은 연속적으로 변화하는 무선 채널의 전송 오류율에알맞은 정정 코드를 채택하지 못해 과도한 대역폭 낭비로 인하여 효율이 떨어지는 문제가 있다. 이러한 문제를 개선하기 위해서는 채널의 상태에 따라 정정 코드를 동적으로 변경하는 것이 필요하다. 본 논문은 FEC방식을 802.11 MAC 프로토콜에 적용할 수 있는 방안에 대해서 기술하고 채널 에러 변화에 따라 능동적으로 정정 코드 양을 조절하여 재 전송하는적응적 FEC 알고리즘을 제안한다. 본 논문에서 제안한 적응적 FEC 알고리즘을 802.11 MAC 프로토콜에 적용하여 성능을 측정한 결과 최대 80%정도 성능이 향상된 것을 확인할 수 있었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.11
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• pp.1054-1062
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• 1991

In this paper a communication system using direct sequence spread spectrum (DS/SS) technique is constructed to transmit burst format data over power line channel with impulsive noise and narrowband interferences. Fast code synchronization is acquired by digital matched filter and data decision is accomplished by sampling pulses. In order to examine the performance of the power line communication system, but error rate and packet loss rate are measured over the simulation channel with various noise sources. When the packet composed of 1-bit preamble and 63-bit data is transmitted under very high burst impulsive noise, the bit error rate is about 10$^3$-10$^4$ and the packet loss rate is below 0.07.

• Proceedings of the Korea Information Processing Society Conference
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• 2007.11a
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• pp.1147-1150
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• 2007

인터넷에서 발생하고 주고 받는 통신 데이터 패킷의 양 중 다양한 컨텐트 애플리케이션 사이의 암호화된 통신이 차지하는 비중은 날로 증가하고 있다. 하지만, 인터넷의 기술적 특성과 한계로 인하여 아직은 완벽한 통신 패킷의 보안 시스템을 만든다는 것이 어려운 상황이며, 특히 데이터양이 많고 실시간 전송이 중요한 컨텐트 애플리케이션들에서는 모든 데이터 패킷을 암호화하고 복호화하는 과정을 거치는 것이 개발상의 번거로움과 시스템의 부하를 가중시키는 일로 인식되고 있다. 본 논문에서는 실시간으로 연속적이며 다양한 형태의 데이터 패킷을 주고 받는 인터넷 컨텐트 애플리케이션을 위하여 개발이 간단하면서도 효율적인 암호화 기법에 대해 제안한다.

• Proceedings of the Korea Information Processing Society Conference
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• 2016.10a
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• pp.119-122
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• 2016

대표적인 네트워크 기반의 이동성 기법인 PMIPv6(Proxy Mobility IPv6)은 이동 단말에 대한 끊김 없는 서비스 지원이 가능하다. 하지만 PMIPv6 기법은 단말과 홈 에이전트 간 IP Tunnel을 생성해야 하며, 그에 따른 시그널링 비용이 발생하게 된다. 또한, 단말 이동에 따른 연속적인 서비스 지원만 고려할 뿐 이동 중 발생하는 패킷 손실은 고려하지 않는 문제점이 있다. 본 논문은 이러한 문제점을 해결하기 위해 SDN(Software Defined Networks)을 적용하였다. 데이터 플레인에서 Anchor 스위치를 선정하여 이동 전 위치의 단말로 전송되던 패킷을 이동한 위치의 단말로 Re-direction 함으로써 패킷 손실을 줄이기 위한 기법을 제안한다.

• Journal of Digital Contents Society
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• v.10 no.2
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• pp.279-289
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• 2009

As the TCP is the protocol designed for the wired network that packet loss probability is very low, because TCP transmitter takes it for granted that the packet loss by the wireless network characteristics is occurred by the network congestion and lowers the transmitter's transmission rate, the performance is degraded. In this article, we suggest the newly improved algorithm using two parameters, the local retransmission time value and the local retransmission critical value to the BS based on the Snoop. This technique adjusts the base stations local retransmission timer effectively according to the wireless link status to recover the wireless packet loss rapidly. We checked that as a result of the suggested algorithm through various simulations, A-Snoop protocol improve more the wireless TCP transmission rate by recovering the packet loss effectively in the wireless link that the continuous packet loss occur than the Snoop protocol.

• Journal of KIISE:Computing Practices and Letters
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• v.14 no.4
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• pp.352-362
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• 2008

In the mobile ad hoc network, network nodes are continuously capable of movement, but have battery limitations and these characteristics of MANET cause network link failure. Therefore, when performing ubiquitous services with the application of MANET, a technique is required to improve the packet transmission rates as to enable continuous packet transmission while flexibly adapting to topology changes caused by the mobility of nodes. In this paper, we are proposing PAOMDV(Possibility based AOMDV) and OLSR-ME(OLSR with Modified Energy Efficiency) which are mixed-type protocols to solve the problem for design a more efficient protocol. Especially by classifying mobile ad hoc networks as clusters, OLSR-ME protocol is implemented on communication within the clusters, while PAOMDV is used in protocols between clusters. In addition, a selection algorithm for forwarding nodes that responding packet relays in the case of increased hops was proposed for designing a more efficient protocol than existing protocols. We verified the performance of the proposed protocols through a comparative experiment with AODV, AOMDV, ZRP, and the existing protocols. We confirmed the results of the experiment revealed that the proposed protocol had the best packet transmission rate, the shortest delay between nodes, and showed better results in routing overhead.

• The Journal of the Korea Contents Association
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• v.10 no.8
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• pp.59-65
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• 2010

This paper proposes an implementation of a hardware module for transmitting MPEG-2 TS data over the internet protocol (IP)-based network. This implementation consists of two modules; one is an encapsulation module which bridges between n TS packets, where $1\;{\leq}\;n\;{\leq}\;7$, and an IP packets, the other is a packet conversion module which extracts an DSM-CC PS packet from consecutive TS packets and then reconstructing an IP packet. So, these IP packets are carried over 150 megabits per second. Although overall work flow of the proposed DeMUX is based on the reference design of ALTERA, the DeMUX is enhanced by modifying it and performs more functions by adding a packet conversion module. The DeMUX is described by Verilog-HDL (hardware description language) and shows the faithful functionality and throughput through the simulation.