• Journal of KIISE:Information Networking
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• v.36 no.5
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• pp.369-375
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• 2009

Residential broadband networks (RBN) are quickly evolving from xDSL/Cable networks to FTTH services with lOOMbps bandwidth. With the high-speed Internet services, subscribers tend to use web, multimedia, game or file sharing applications. Therefore, performance of RBN is important to the quality of services for users. Typically, Internet service providers (ISPs) are providing their RBN performance results that are measured by subscribers with the dedicated measurement tools. However, they do not make their results public to people. In order to obtain the performance results measured by users, we have to induce a lot of subscribers, which will take a lot of cost and time. Monarch, that is used in this paper, could provide the 3rd party measurement method without the help of ISPs and subscribers. With the extensive experiments through Monarch, we have shown that 3rd party measurement results are not much different from those of user-initiated measurement. In addition, we have observed that bandwidth is around $80{\sim}90Mbps$ and RTT is less than 30ms in FTTH services provided by three major Korean ISPs.

• Journal of Mechanical Science and Technology
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• v.17 no.4
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• pp.511-520
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• 2003

In room acoustics, reverberation time is an important acoustic parameter. However it is often difficult to measure short reverberation times at low frequencies with a traditional band pass filter bank if the product of filter bandwidth (B) and reverberation time (T) is small. It it well known that the minimum permissible product of bandwidth and reverberation time of the traditional band pass filter is at least 16. This strict requirement makes it difficult to measure short reverberation times of an acoustic room at low frequencies exactly. In order to reduce this strict requirement, in the previous paper, the wavelet filter bank was developed and the minimum permissible product of bandwidth and reverberation time was replaced with 4. In the present paper it is demonstrated how the short reverberation times of an practical room at low frequencies are successfully measured by using the wavelet filter bank and the results are compared with the traditional method using a band past filer bank.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.21 no.4
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• pp.104-114
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• 2007

In this paper, an Ad-hoc Routing Protocol that works in wireless Ad-hoc communication networks with multiple radios and multiple channels and that controls call admission based on bandwidth measurement is proposed. Unlike the conventional Ad-hoc node with a single radio using a single channel, an Ad-hoc node of the protocol proposed, the MCQosR(Multiple Channel Quality of Service Routing), has multiple radios and uses multiple channels, which allows full duplex transmission between wireless Ad-hoc nodes, and reduces intra interference on the route. Also, a fixed channel only for reception at each node enables the estimation of the available bandwidth, which is used to control the call admission for QoS provision. The performance of the MCQosR was verified by simulation.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2002.05a
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• pp.1077-1080
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• 2002

In building acoustics, reverberation time is an important acoustic parameter. However, it is often difficult to measure short reverberation times at low frequencies using the traditional band pass filter bank if the product of bandwidth (B) and reverberation time (T) is small. It is well known that the minimum permissible product of bandwidth and reverberation time of the traditional band pass filter is at least 16 ［F. Jacobsen, J. Sound Vib. 115, 163-170 (1987)]. This strict requirement makes it difficult to measure short reverberation times of an acoustic room at low frequencies exactly. In order to reduce this strict requirement, recently, the wavelet filter bank is developed and the minimum permissible product of bandwidth and reverberation time is replaced with 4 [S. K. Lee, J, Sound Vib. 252, 141-153 (2002)]. In the present paper, it is demonstrated how the short reverberation times at low frequencies are successfully measured by using the wavelet filter bank. In order to present this job, two synthetic signals and one measured signal are used for impulse responses of an acoustic room.

• Journal of electromagnetic engineering and science
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• v.16 no.1
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• pp.52-56
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• 2016

This paper suggests a class-F power amplifier with split open-end stubs to provide a broadband high-efficiency operation. These stubs are designed to have wide bandwidth by splitting wide open-end stubs into narrower stubs connected in shunt in an output matching network for class-F operation. In contrast to conventional wideband class-F designs, which theoretically need a large number of matching lines, this method requires fewer transmission lines, resulting in a compact circuit implementation. In addition, the open-end stubs are designed with slant ends to achieve additional wide bandwidth. To verify the suggested design, a 10-W class-F power amplifier operating at 1.7 GHz was implemented using a commercial GaN transistor. The measurement results showed a peak drain efficiency of 82.1% and 750 MHz of bandwidth for an efficiency higher than 63%. Additionally, the maximum output power was 14.45 W at 1.7 GHz.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.3
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• pp.593-603
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• 1997

Connection admission decision in ATM networks requires decision made in real time using fast algorithm. It is difficult to construct a model of the multiplexed traffic and thus, approximation of the traffic load is necessary. In this paper, we propose a measurement-based dynamic CAC(Connection admission Control) in ATM(Asynchronous Transfer Mode) networks, which observes current traffic by the moving window and set the window size to the value which is computed from the measured cell loss amount. It is based on the measurements of the traffic load over an admission period that is load enough to reflect the current traffic behavior instead of analytic modeling. And, the dynamic reallocation of bandwidth for each class leads to effective bandwidth utilization. The performance of proposed method is analyzed through computer simulation. The performance of proposed method is analyzed by using SIMAN simulation package and FORTRAN language. As can be seen in the simulation result, cell loss performance and bandwidth utilization have been increased.

• Journal of electromagnetic engineering and science
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• v.19 no.1
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• pp.56-63
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• 2019

Typical negative group delay circuits (NGDC) are analyzed in terms of signal attenuation, group delay, and bandwidth using S-parameters. By inverting these formulations, we derive and present the design equations (for NGD circuit elements) for a desired specification of the two among the three parameters. The proposed design method is validated through simulation examples for narrow- and wide-band pulse inputs in the time and frequency domains. Moreover, an NGDC composed of lumped elements is fabricated at 1 GHz for measurement. As a function of frequency, the circuit-/EM-simulated and measured group delays are in good agreement. The provided simple NGDC design equations may be useful for many applications that require compensations of some signal delays.

• Journal of information and communication convergence engineering
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• v.21 no.1
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• pp.1-8
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• 2023

This paper presents a design method for a wideband bandpass filter (BPF) which compensates for frequency dependency based on the image admittance and image phase. In the proposed method, new compensation methods for the admittance and phase are integrated with the conventional method. The proposed method improves the frequency shift and reduces the unwanted bandwidth when designing more than 20% of the Fractional Bandwidth (FBW), whereas the conventional method exhibits frequency degradation at only 10% FBW. The proposed design theory was verified by applying it to both lumped elements and distributed lines through circuit simulation and measurements without an optimization process. The measurement results demonstrate improvements in the frequency shift and target bandwidth. In the future, an accurate design method based on frequency dependence can be implemented for the next-generation broadband communication system applications.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.24 no.7
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• pp.26-40
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• 2010

We propose a new routing protocol, MCQosR, that is based on bandwidth estimation, admission control, and a routing metric, MCCR - suitable for wireless ad-hoc networks with multiple radios and channels. To use the full capacity of a wireless link, we assume a node with multiple radios for full duplex operation, and a radio using multiple channels to exclude route-intra interference. This makes it possible to use the capacity of a wireless link. Then, to provide bandwidth and delay guarantee, we have a radio with a fixed channel for layer-3 data reception at each node, used to estimate the available bandwidth and expected delay of a wireless link. Based on the estimate of available bandwidth and delay, we apply the call admission control to a new call requiring bandwidth and delay guarantee. New calls with traffic that will overflow link or network capacity are rejected so the accepted calls can use the required bandwidth and delay. Finally, we propose a routing metric, MCCR, which considers the channel contentions and collisions of a wireless link operating in CSMA/CA. MCCR is useful for finding a route with less traffic and distributing traffic over the network to prevent network congestion as much as possible. The simulation of the MCQosR protocol and the MCCR metric shows traffic is distributed and guaranteed service is provided for accepted calls.

• Journal of KIISE
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• v.44 no.6
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• pp.628-636
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• 2017

HTTP adaptive streaming is a technique that improves the quality of experience by storing various quality videos on the server and requesting files of the appropriate quality based on network bandwidth. However, it is difficult to measure the actual bandwidth in wireless networks with frequent bandwidth changes and high loss rate. Frequent quality changes and playback interruptions due to bandwidth measurement errors degrade the quality of experience. We propose a technique to estimate the available bandwidth by measuring the jitter, which is the derivation of delay, on a packet basis and assigning a weight according to jitter. The proposed scheme reduces the number of quality changes and mitigates the buffer underflow by reflecting less bandwidth change when high jitter occurs due to rapid bandwidth change. The experimental results show that the proposed scheme improves the quality of experience by mitigating buffer underflow and reducing the number of quality changes in wireless networks.