• 한국정보통신학회논문지
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• 제11권2호
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• pp.416-421
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• 2007

ZigBee는 저전력 특성을 기반으로 무선 센서 네트워크의 전송 프로토콜로 활용되고 있다. ZigBee 네트워크에서 연속적인 데이터 전송률이 ZigBee에서 처리 가능한 최대 전송률 수준으로 요구되는 경우에 데이터를 중계하는 라우터가 수신과 송신을 동시에 처리 할 수 없기 때문에 실질 전송률은 절반 이하로 떨어지게 된다. 본 논문에서는 2개의 송수신 모듈을 장착하여 수신과 송신을 동시에 처리할 수 있는 라우터를 설계하고 구현하였다. 또한, 송수신을 동시에 처리할 수 있도록 서로 다른 채널을 할당하는 중앙집중식 채널할당 알고리즘을 제시하였다. 실험을 통해 제안한 이중 채널 라우터의 최대 전송률이 일반 라우터의 2배에 가까운 150kbps에 이르는 것을 보였다.

• 한국정보기술응용학회:학술대회논문집
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• 한국정보기술응용학회 2005년도 6th 2005 International Conference on Computers, Communications and System
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• pp.211-214
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• 2005

There are emerging many eScience applications. More and more scientists want to collaborate on their investigation with international partners without space limitation by using these applications. Since these applications have to analyze the massive raw data, scientists need to send and receive the data in short time. So today's network related requirement is high speed networking. The key point of network performance is transport protocol. We can use TCP and UDP as transport protocol but we use TCP due to the data reliability. However, TCP was designed under low bandwidth network, therefore, general TCP, for example Reno, cannot utilize the whole bandwidth of high capacity network. There are several TCP variants to solve TCP problems related to high speed networking. They can be classified into two groups: loss based TCP and delay based TCP. In this paper, I will compare two approaches of TCP variants and propose a hybrid approach for high speed networking.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제9권9호
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• pp.3594-3610
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• 2015

Recent advancements in broadband wireless communication technologies enable mobile users to receive video streaming services with various smart devices. The long term evolution (LTE) network provides high bandwidth and low latency for several emerging mobile applications. This paper proposes the buffer aware scheduling (BAS) approach to schedule the downlink video traffic in LTE network. The proposed BAS scheme applies the weighting function to heuristically adjust the scheduling priority by considering the buffer status and channel condition of UE so as to reduce the time that UE stays in the connected state without receiving data. Both of 1080P and 2160P resolution video streaming sources were applied for exhaustive simulations to examine the performance of the proposed scheme by comparing to that of the fair bandwidth (FB) and the best channel quality indicator (CQI) schemes. The simulation results indicate that the proposed BAS scheme not only achieves better performance in power saving, streaming delivery time, and throughput than the FB scheme while maintaining the similar performance as the best CQI scheme in light traffic load. Specifically, the proposed scheme reduces streaming delivery time and generates less signaling overhead than the best CQI scheme when the traffic load is heavy.

• ETRI Journal
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• 제38권6호
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• pp.1074-1084
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• 2016

This paper presents a transmission method for improving human body communications in terms of spectral efficiency, and the performances of bit-error-rate (BER) and frame synchronization, with a highly simplified structure. Compared to the conventional frequency selective digital transmission supporting IEEE standard 802.15.6 for wireless body area networks, the proposed scheme improves the spectral efficiency from 0.25 bps/Hz to 1 bps/Hz based on the 3-dB bandwidth of the transmit spectral mask, and the signal-to-noise-ratio (SNR) by 0.51 dB at a BER of $10^{-6}$ with an 87.5% reduction in the detection complexity of the length of the Hamming distance computation. The proposed preamble structure using its customized detection algorithm achieves perfect frame synchronization at the SNR of a BER of $10^{-6}$ by applying the proposed pre-processing to compensate for the distortions on the preamble signals due to the band-limit effects by transmit and receive filters.

• 정보처리학회논문지C
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• 제10C권7호
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• pp.937-942
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• 2003

본 논문은 일본 기가비트 네트워크 테스트베드인 JNG상에서 대표적인 스트림형 트래픽인 DV와 MPEG2에 대한 최종 사용자간에 실측을 통하여 초고속 네트워크 상에서 정상 부하의 변화에 따른 각 스트림 트래픽의 전송 지연과 패킷 손실에 대하여 고찰한 것이다. 이 결과에 의하면 50Mbps의 IP over ATM상에서 DV와 MPEG2 스트림을 전송할 때, 부가된 정상 부하의 패킷 크리(512바이트 이상)에 무관하게 부하율이 네트워크 대역폭의약 95％에 달하는 시점에서 패킷 손실이 다량 발생하고 전송 지연도 급증하였다. 또한 최종 사용자 시스템상의 버퍼 크기와 송수신 버퍼 수는 전송 지연과 패킷 손실에 영향을 미치지 않음을 확인하였다.

• ETRI Journal
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• 제30권4호
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• pp.526-534
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• 2008

This paper presents a fully integrated 0.13 ${\mu}m$ CMOS MB-OFDM UWB transmitter chain (mode 1). The proposed transmitter consists of a low-pass filter, a variable gain amplifier, a voltage-to-current converter, an I/Q up-mixer, a differential-to-single-ended converter, a driver amplifier, and a transmit/receive (T/R) switch. The proposed T/R switch shows an insertion loss of less than 1.5 dB and a Tx/Rx port isolation of more than 27 dB over a 3 GHz to 5 GHz frequency range. All RF/analog circuits have been designed to achieve high linearity and wide bandwidth. The proposed transmitter is implemented using IBM 0.13 ${\mu}m$ CMOS technology. The fabricated transmitter shows a -3 dB bandwidth of 550 MHz at each sub-band center frequency with gain flatness less than 1.5 dB. It also shows a power gain of 0.5 dB, a maximum output power level of 0 dBm, and output IP3 of +9.3 dBm. It consumes a total of 54 mA from a 1.5 V supply.

• Journal of Communications and Networks
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• 제18권3호
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• pp.299-310
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• 2016

In practical wireless relay communication systems, non-destination nodes are assumed to be idle not receiving signals while the relay sends messages to a particular destination node, which results in reduced bandwidth efficiency. To improve the bandwidth efficiency, we relax the idle assumption of non-destination nodes and assume that non-destination nodes may receive signals from sources. We note that the message relayed to a particular node in such a system gives rise to interference to other nodes. To study such a more general relay system, we consider, in this paper, a relay system in which the relay first listens to the source, then routes the source message to the destination, and finally produces interference to the destination in sending messages for other systems. We obtain capacity upper and lower bounds and study the optimal method to deal with the interference as well as the optimal routing schemes. From analytic results obtained, we find the conditions on which the direct transmission provides higher transmission rate. Next, we find the conditions, by numerical evaluation of the theoretical results, on which it is better for the destination to cancel and decode the interference. Also we find the optimal source power allocation scheme that achieves the lower bound depending on various channel conditions. We believe that the results provided in this paper will provide useful insights to system designers in strategically choosing the optimal routing algorithms depending on the channel conditions.

• 한국전자파학회논문지
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• 제28권2호
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• pp.97-104
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• 2017

본 논문에서는 C 대역, X 대역, Ku 대역을 포함하는 다채널 영상레이다를 위한 송수신기를 설계, 제작하고, 전기적인 성능을 검증하기 위한 연구를 수행하였다. 송수신기는 대역별 송신기, 대역별 수신기, 대역별 신호선택기, 고안정발진기, 주파수합성기, 제어기, 전원분배기로 구성된다. 송수신기는 운용모드에 따라 경로 선택 및 수신 대역폭 선택 기능을 가진다. 송수신기는 C, X, Ku 대역 3가지 대역을 모두 송신 및 수신할 수 있으며, 각 대역에서 최대 300 MHz의 대역폭을 가지고, T/R 모듈을 구동시키기에 적합하도록 각 대역별 송수신기 최종 송신 출력은 20 dBm 이상이다. 수신 대역폭은 필요 요구 기능에 따라 모드 선택이 가능하며, 수신 이득은 대역별 C 대역 52 dB, X 대역 50 dB, Ku 대역 60 dB 정도를 가지고 있으며, 잡음 지수는 대역별 H, V 편파에서 Ku 대역 V 편파 최대 4.28 dB의 성능을 보였다. 전기적 성능 시험 결과, 다중대역 송수신기는 다채널 영상레이다에서 요구되는 성능을 만족하였다.

• International journal of advanced smart convergence
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• 제13권3호
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• pp.1-12
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• 2024

Recently, Audio systems transform the configuration of conventional sound reinforcement and public address systems using audio over internet protocol (AoIP), whereby audio signals are transmitted and received based on internet protocol (IP). Currently, AoIP technologies are leading the audio market, and various technologies have been released. Audio networks and the control hierarchy over peer-to-peer (Anchor) technology based on AoIP transmit and receive audio signals over a wide bandwidth without an audio mixer. Audio system based on Anchor technology is constructed by connecting the on-site audio center (OAC), a device that can transmit and receive audio sources and output equipment over IP. Receiving OAC of the Anchor technology can receive and mix audio signals transmitted from different IPs; consequently, novel audio systems can be configured by replacing conventional audio mixers. However, the Anchor technology does not have an equalizer function for improving the quality of audio equipment. Therefore, tone distortion may occur owing to signal loss between equipment, poor audio-signal clarity, and howling due to audio deformation according to different architectural structures and environments. In this study, we implemented an audio effect device capable of tone control using the Audio Processor Core. Using Anchor technology, tone control was realized through an audio effect device in the receiving OAC. The output of the incoming OAC was received by the audio effect device, which adjusted the tone and then outputted it. Thus, the tone issues in Anchor technology were overcome by the receiving OAC and audio effect devices. In future, audio system configurations using Anchor technology could be the standard for audio equipment.

• 한국전자파학회논문지
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• 제13권9호
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• pp.858-864
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• 2002

본 논문에서는 동일 주파수 대역을 사용하는 기존 시스템에 의한 간섭 환경에서 TH/UWB(Time Hopping/ Ultra Wideband 이하 Impulse Radio) 시스템의 성능을 분석하였다. 간섭 점유율이 0.1 이하인 경우는 5 GHz 대역에서 ISM 대역의 80 MHz 대역폭이나 802.113의 사용대역폭 20 MHz와 같은 IR 시스템에 비해 매우 협소한 대역을 사용하는 경우로써 향후 3.1 - 10.6 GHz 대역에서 공유되는 IR 시스템에서 간섭으로 영향을 미칠 경우를 상정하여 분석하였다. 결과에 의하면, 간섭 전력의 크기보다는 간섭의 점유 대역폭에 따라서 성능 변화가 크게 나타남을 알 수 있었다. 또한, 부분대역 간섭 환경에서는 펄스반복개수(Ns)를 증가시킴으로써 간섭 점유율이 큰(0.1이상) 협대역 간섭일수록 큰 폭의 성능 개선을 얻을 수 있었다. 그러나, 간섭 점유율이 적은(0.1이하) 협대역 간섭일 때는 시스템 성능에 거의 영향을 미치지 않음을 알 수 있었다. 따라서, 정확한 간섭 점유율 추정을 통해 간섭의 대역폭 점유율에 따라 최적의 펄스반복개수를 설정하고 적정한 간섭억압기법을 적용함으로써 최상의 시스템 성능 및 전송효율을 얻을 수 있을 것이다.