• Journal of KIISE:Computing Practices and Letters
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• v.16 no.10
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• pp.975-979
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• 2010

Existing quality adaptation schemes are useful for improving network stability in a congested network, but these schemes do not guarantee the quality of experience on video streaming due to the frequent variations in playback quality for a streaming service. In this paper, we propose a network-adaptive quality adaptation scheme to guarantee the quality of experience on SVC(Scalable Video Coding) video streaming. The proposed scheme estimates the available bandwidth by bandwidth measurement scheme using media characteristic, and then smoothly adjusts the quality level of SVC video stream according to the estimated bandwidth. Through the simulation, we prove that our scheme guarantees the quality of experience for video streaming by minimizing quality variation.

• The Journal of the Korea Contents Association
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• v.10 no.9
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• pp.36-44
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• 2010

In this paper, we propose video quality control system(VQCS) which can control video service quality through the monitoring of end-to-end available bandwidth for video streaming service like IPTV in NGN convergence network. Various multimedia services such as video, voice and gaming service can be provided by IPTV, and these services require large amounts of bandwidth. At this time, video quality degradation like video jerkiness, block distortion and blurring is caused when network available bandwidth is insufficient. Available bandwidth monitoring method is need to stably control video streaming quality. So, we periodically calculate the amount of the packets in link and measure available bandwidth by using total length field in IP header at terminal. Scalability extractor in network selects suitable video streaming data rate based on available bandwidth and transports video streaming with adaptive data rate to prevent video quality deterioration.

• Journal of Internet Computing and Services
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• v.9 no.6
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• pp.179-188
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• 2008

Even though a lot of researches to improve the performance of TCP over wireless networks have been performed, the performance of TCP cannot satisfy users under high packet loss rate of wireless networks. This paper addresses the problem of TCP Westwood under high packet loss rate and reveals the reasons through the extensive simulation analysis. The ABE(Available Bandwidth Estimation) of TCP westwood becomes incorrect due to TCP dynamics of loss recovery. Based on this observation, we propose two algorithms for enhancing the correctness of ABE. We have evaluated the effects of the proposed algorithms through simulation. The proposed algorithms have achieved more correct ABE under the packet loss rate less than 1%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.12B
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• pp.2275-2282
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• 1999

In ATM networks, bandwidth allocation using available bandwidth is important to control ABR traffic efficiently. However, it is difficult to dynamically assign bandwidth on bursty traffic, because the tracking of the existing available bandwidth information uses statistical value. In this paper, we propose a output-bufferd ATM switch using Per-VC queueing scheme with timer and Controller. The controller determines whether each VC is in congestion state or not, and the timer can assign dynamically as determining the available bandwidth for the Present period by Backlogged time and total link capacity, and then transmitting the determined bandwidth to source. Instead of using statistical parameters, the proposed scheme can improve link efficiency as allocating the bandwidth of the next period, by use of measured bandwidth during the current period.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.19 no.1
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• pp.89-96
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• 2000

ABR service is defined by ATM Forum and ITU for the efficient use of link bandwidth, and 2-pass service policing is proposed for this service. 2-pass service policing is effective by the real-time measurement of the used bandwidth, and this scheme obtains multiplexing gam and control efficiently ABR traffic as dynamically allocating rate by residual bandwidth information. In this study, we propose the real-time bandwidth prediction scheme for ABR traffic control, as using dynamic rate allocation by available bandwidth information. This study can obtain the simple hardware structure by means of as predicting available bandwidth by the total link bandwidth and the sum of transmission rate on the backlogged state connections.

• Journal of Broadcast Engineering
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• v.23 no.6
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• pp.813-823
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• 2018

UHD content is characterized by higher bandwidth requirements to provide seamless playback. The existing HTTP adaptive streaming scheme periodically requests video segment of lower quality than the available bandwidth in the steady state. In the steady state, the On-Off pattern is repeated due to the difference of the available bandwidth and the encoding rate of the content. And the repeated Off interval is causing the problem that can not efficiently use the available bandwidth. In the wireless environment where the delay and loss rate are higher with large bandwidth fluctuations, this problem is exacerbated because the available bandwidth is estimated inaccurately. Therefore, in this paper, we propose a segment scheduling scheme for efficient bandwidth utilization of UHD contents streaming in wireless environment. Proposed scheme is a group segment that simultaneously reqeusts multiple segments in one request message in order to reduce frequency of OFF interval. Through simulation, Our scheme improves bandwidth utilization.

• Journal of Korea Multimedia Society
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• v.9 no.7
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• pp.844-853
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• 2006

The IEEE 802.17 Resilient Packet Ring (RPR) for future Local, Metropolitan, and Wide Area Networks was recently standardized, but it still suffer from delay jitter deterioration and even some bandwidth loss under unbalanced overload. In order to overcome these drawbacks, this paper proposes a bandwidth redistribution based fairness control method, compatible with the legacy one, in which each congested node measures the amount of available bandwidth of its bottleneck link resulted from regulating upstream nodes' shares of the link bandwidth, calculates optimal fair rate with the number of uptream nodes requiring more bandwidth, and then redistributes the available bandwidth to the upstream nodes by advertising the rate. The performance evaluation results show that the proposed method fairly redistributes 95% of the bottleneck link bandwidth with even only two redistributions.

• Proceedings of the Korean Information Science Society Conference
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• 2010.06d
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• pp.301-305
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• 2010

최근 무선 네트워크 서비스의 수요가 증가함에 따라 낮은 비용, 간편한 설치 그리고 높은 대역폭 특성을 가지는 무선랜 (WLAN)을 기반으로 한 사내 네트워크 구성이 학교 및 기업을 중심으로 증가하고 있다. 이로 인해 사내 네트워크를 통한 랩탑 (Laptop), 넷북 (Notebook), 스마트 폰 (Smart Phone), 그리고 UMPC (Ultra Mobile Personal Computer)등 휴대용 단말을 이용한 스트리밍 서비스의 수요가 증가하고 있다. 하지만 무선랜 환경은 사용자 이동에 의해 핸드오버가 발생할 경우, 이동할 대상 네트워크의 혼잡 상태를 고려하지 않는다. 따라서 이동할 네트워크에 빠르게 적응하지 못함으로써 패킷 손실에 의한 서비스의 불연속성 문제가 발생할 수 있다. 본 논문에서는 이러한 문제점을 해결하기 위해 핸드오버 이전에 이동 대상 네트워크의 혼잡 상태를 인지하여 전송률을 조절함으로써, 스트리밍 서비스의 불연속성 문제를 해결하는 적응적인 스트리밍 시스템을 설계 및 구현하였다. 제안하는 시스템은 핸드오버 이전에 이동 대상 네트워크의 가용대역폭을 측정하여 혼잡 상태를 인지하고, 측정한 가용대역폭 값을 기반으로 전송하는 프레임의 수를 우선순위에 따라 차등적으로 조절하여 전송률을 조절하였다.

• Journal of the Korea Society of Computer and Information
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• v.4 no.4
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• pp.155-162
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• 1999

Various applications of video conferencing are required real-time transmission in order to offer service of best effort in internet. Because the bandwidth of internet changes dynamically, appropriated QoS could not be guaranteed To resolve the problem. available bandwidth between sender and receiver is measured. And according to measured bandwidth, the transmission of multimedia data is controlled In this paper, we propose algorithm of efficient transmission for best QoS in internet According to a present status of network, we measure available bandwidth using feedback RTCP information and change a compression rate to reduce a producing CODEC data. And according to the priority that is measured by packet loss for received RTCP information, we abandon frames indicated as lower weight in transmission buffer of sender.