• Journal of KIISE:Information Networking
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• v.31 no.2
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• pp.188-195
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• 2004

We have analyzed the performance of TCP in the CDMA mobile communications systems with the adaptive modulation and coding(AMC). The wireless channel using AMC is characterized with not high error rate but highly varying bandwidth. Due to time-varying bandwidth, timeout events of TCP occurs more frequently, which leads to the throughput degradation. The analysis model is composed of the two parts. In the first part, we divide TCP packet stream into ‘packet groups’and derive the probability distribution of the wireless transmission time of each Packet group that reflects the time varying characteristics of AMC. In the second part, we formulate embedded Markov chain by making use of the results of the first part to model TCP timer mechanism and wireless transmission. Since our system model is characterized by the forward link high speed data transmission using AMC, the results reported in this paper can be used as a guideline for the design and operation of HSDPA, 1xEV-DO, and 1xEV-DV.

• Journal of KIISE
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• v.43 no.7
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• pp.827-833
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• 2016

HTTP Adaptive Streaming (HAS) is a technique that adapts its video quality to network conditions for providing Quality of Experience. In the HAS approach, a video content is encoded at multiple bitrates and the encoded video content is divided into several video segments. A HAS player estimates the network bandwidth and adjusts the video bitrate based on estimated bandwidth. However, the segment scheduler in the conventional HAS player requests video segments periodically without considering TCP. If the waiting duration for the next segment request is quite long, the TCP connection can be initialized and it restarts slow-start. Slow-start causes the reduction in TCP throughput and consequentially leads to low-quality video streaming. In this study, we propose a TCP-aware segment scheduling scheme to improve performance of HAS service. The proposed scheme adjusts request time for the next video request to prevent initialization of TCP connection and also considers the point of scheduling time. The simulation proves that our scheme improves the Quality of Service of the HAS service without buffer underflow issue.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.6C
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• pp.581-593
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• 2002

In this paper, we proposed an Adaptive RIO scheme to solve the problem of RIO scheme that occurs when admission control is performed for QoS guarantee of Assured Service in Differentiated Services. To prevent an early random drop of the admitted In-profile packet, proposed Adaptive RIO scheme updates parameters of RIO scheme every time interval according to the estimated numbers of maximum packet arrivals of In-profile traffic and total traffic during the next time interval. The numbers of maximum packet arrivals during the next time interval are estimated based on the buffer size determined by the network topology and the ratio of bandwidth allocated to each subclass. We found from simulation results that, compared with RIO scheme, proposed Adaptive RIO scheme can improve performance of the throughput for In-profile traffic when admission control is performed or congestion occurs.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.21 no.10
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• pp.73-81
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• 2007

To improve the capability of instantaneous torque control, a design method of an MRAC-based adaptive current controller for a PMSM servo motor is proposed. In the synchronous frame current controller, a new control inputs can be obtained through the decoupling compensation. Using this, a desired controller bandwidth can be assigned However, the control performance may be degraded due to disturbances caused by the parameter variations or dead time of the switch. To improve these drawbacks, an adaptive current controller is proposed and the design method is obtained using the hyperstability theory. The asymptotic stability is proved and the effectiveness is verified through simulations and experiments using DSP TMS320C31.

• Journal of Communications and Networks
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• v.7 no.4
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• pp.489-498
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• 2005

For better performance over a noisy channel, mobile wireless networks transmit packets with forward error correction (FEC) code to recover corrupt bits without retransmission. The static determination of the FEC code size, however, degrades their performance since the evaluation of the underlying channel state is hardly accurate and even widely varied. Our measurements over a wireless sensor network, for example, show that the average bit error rate (BER) per second or per minute continuously changes from 0 up to $10^{-3}$. Under this environment, wireless networks waste their bandwidth since they can't deterministically select the appropriate size of FEC code matching to the fluctuating channel BER. This paper proposes an adaptive FEC technique called adaptive FEC code control (AFECCC), which dynamically tunes the amount of FEC code per packet based on the arrival of acknowl­edgement packets without any specific information such as signal to noise ratio (SNR) or BER from receivers. Our simulation experiments indicate that AFECCC performs better than any static FEC algorithm and some conventional dynamic hybrid FEC/ARQ algorithms when wireless channels are modeled with two-state Markov chain, chaotic map, and traces collected from real sensor networks. Finally, AFECCC implemented in sensor motes achieves better performance than any static FEC algorithm.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.8
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• pp.3804-3819
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• 2018

The increasing popularity of HTTP adaptive video streaming services has dramatically increased bandwidth requirements on operator networks, which attempt to shape their traffic through Deep Packet inspection (DPI). However, Google and certain content providers have started to encrypt their video services. As a result, operators often encounter difficulties in shaping their encrypted video traffic via DPI. This highlights the need for new traffic classification methods for encrypted HTTP adaptive video streaming to enable smart traffic shaping. These new methods will have to effectively estimate the quality representation layer and playout buffer. We present a new machine learning method and show for the first time that video quality representation classification for (YouTube) encrypted HTTP adaptive streaming is possible. The crawler codes and the datasets are provided in [43,44,51]. An extensive empirical evaluation shows that our method is able to independently classify every video segment into one of the quality representation layers with 97% accuracy if the browser is Safari with a Flash Player and 77% accuracy if the browser is Chrome, Explorer, Firefox or Safari with an HTML5 player.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.05a
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• pp.673-675
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• 2010

This paper proposes the adaptive class-E power amplifier with maintaining high power added efficiency (PAE) in 400MHz range. This amplifier is used a microprocessor to adapt a resonator circuits and to maintain high efficiency in case of input frequency variation. To validate the adaptive amplifier operation, which is a 450MHz operating frequency and a 100MHz bandwidth, the class E amplifier is implemented. As a result, the adaptive amplifier is maintained above 60% efficiency and has a 74.8% maximum efficiency.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.3B
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• pp.380-389
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• 2010

Since the internet is intend to be the best effort service, the system that stream a large amount of high quality medias need a techniques to overcome the network status for implementation. In this paper, we design and implement a method that estimate quickly whether network permits the needed bandwidth of media and a method that control QoS through that. Presented system uses Relative One-Way Delay(ROWD) trend in the case of the former, and leverages temporal encoding among Scalable Video Coding(SVC) that is apt to apply real time comparatively in the case of the latter. The streaming server classifies the medias by real time to several rates and begins transmission from top-level and is reported ROWD trend periodically from the client. In case of the server reported only 'Increase Trend', the sever decides that the current media exceeds the available bandwidth and downgrades the next media level. The system uses probe packet of difference quantity of the target level and the present level for upgrading the media level. In case of the server reported only 'No Increase Trend' by the ROWD trend response of the probe packet from client, the media level is upgraded. The experiment result in a fiber to the home(FTTH) environment shows progress that proposed system adapts faster in change of available bandwidth and shows that quality of service also improves.

• Journal of Korea Multimedia Society
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• v.5 no.4
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• pp.441-449
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• 2002

For multimedia traffics to be supported successfully in wireless network environment, it is necessary to provide Qualify-of-Service(QoS) guarantees among mobile hosts(clients). In order to guarantee the QoS, we have to keep the call blocking probability below a target value during hand-off session. However, the QoS negotiated between the client and the network may not be guaranteed due to lack of available channels for traffic of a new cell, since on service mobile clients should be able to continue their sessions. In this paper, we propose an efficient load-balancing algorithm based on the adaptive bandwidth reservation scheme for enlarging available channels in a cell. Proposed algorithm predicts the direction of clients in a cell and adjusts the amount of the channel to be reserved according to the load status of the cell. This method is used to reserve a part of bandwidths of a cell for hand-off calls to its adjacent cells and this reserved bandwidth can be used for hand-off call prior to new connection requests. If the number of free channels is also under a low threshold value, our scheme use a load-balancing algorithm with an adaptive bandwidth reservation. In order to evaluate the performance of our algorithm, we measure metrics such as the blocking probability of new calls and dropping probability of hand-off calls, and compare with those of existing schemes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.8
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• pp.2042-2051
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• 1998

VBR compressed, pre-recorded video is known to exhibit significant, multiple time-scale bit rate variability. To reduce the variability and burstiness of the aggregated transmission rate as low as possible, in this paper, we present an adaptive bandwidth smoothing algorithm, that can be effectively applicalble for VoD services over a shared-medium channel. For these environments where many clients are connected to a single server, by introducing the conventional MVBA(minimum variability bandwidth allocation) algorithm and controlling adaptively the conventional MVBA(minimum variability bandwidth asllocation) algorithm and controlling adaptively the aggregated transmission rate whenever a new clients request is arrived at the server side, the proposed algorithm effectively reduces the burstiness and variability of the aggregated transmission rate. Through computer experiments, it is shown that the proposed method perporms better than the convertional non-aggregated bandwidth smoothing schemes in terms of the peak rate, standard deviation, number of rate changes for the aggregated traffic.