• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1996.06a
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• pp.153-158
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• 1996

Delay and jitter are critical factors in the real-time video services over ATM network. Mostly, delay and jitter problem are generated in input buffer when video are multiplexed. In this paper, we analyze delay and jitter of input buffer, and consider efficient control and flexible bandwidth allocation of video traffic. Also, we analyze decision of buffer size related maximum allowable delay.

• KIPS Transactions on Computer and Communication Systems
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• v.6 no.6
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• pp.255-260
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• 2017

The limitation of real-time video conferencing system is that the delay of network and buffering and the transmission of user information are not efficiently performed between systems, so real - time performance is not guaranteed completely. In order to overcome this problem, the study on the extension of the network infrastructure and the jitter delay is actively carried out, but the study on the buffering delay is insufficient. In this paper, we propose a frame-rate control buffer management (FRCB) scheme to solve the problem caused by buffering delay. The FRCB is used to prevent overflow and underflow of the buffer by adopting the two-stage buffer threshold of Fast-play THreshold (FTH) and Slow-play THreshold (STH). Therefore, it showed better performance than jitter buffer even under high CPU load, and showed that it is suitable for high quality real time video conferencing.

• Proceedings of the IEEK Conference
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• 2001.06c
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• pp.93-96
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• 2001

In this paper, we propose a delay-constrained retransmission method to control packet error or loss in common internet. The Delay Regulator(or Jitter Buffer) which is used to control errors caused by unreliable UDP connection, stores received data packets fDr a small amount of time to prevent network jitter from affecting display quality, which causes constant delay. In this paper, we propose a retransmission method to increase efficiency of ARQ(Automatic Repeat reQuest) by using characteristic of delay regulator.

• Transactions on Control, Automation and Systems Engineering
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• v.4 no.2
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• pp.164-168
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• 2002

In this article, we present the novel approach of avoiding temporal insufficiency of data blocks, jitter, which occurs due to the commencement of new session. We propose to make the sufficient amount of data blocks available on memory such that the ongoing session can survive the cycle extension. This technique is called ″pre-buffering″. We examine two different approaches in pre-buffering: (i) loads all required data blocks prior to starting playback and (ii) incrementally accumulates the data blocks in each cycle. We develop an elaborate model to determine the appropriate amount of data blocks necessary to survive the cycle extension and to compute startup latency involved in loading these data blocks. The simulation result shows that limiting the disk bandwidth utilization to 60% can greatly improve the startup latency as well as the buffer requirement for individual streams.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.93-99
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• 2011

This paper proposes a novel network jitter estimation algorithm for robust VoIP communication system. The proposed method computes the current network environment mode using the differences of arrival time and generation time from sequential received packets. According to the current network environment mode, the jitter variance weights is adjusted to minimize the error for estimating the network jitter. The jitter average and variance are calculated by the autoregressive estimated algorithm, and then the network jitter is estimated by applying the jitter variance weights.

• 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 Korea institute of electronic communication sciences
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• v.6 no.4
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• pp.487-492
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• 2011

This paper proposes a retransmission persistence control scheme for automatic retransmit request (ARQ) protocol to improve the reliability of a wireless link. Most existing ARQ protocols adopt a fixed retransmission persistence. If the ARQ protocol sets the retransmission persistence too low, there is a limitation in providing transmission reliability. On the other hand, if the ARQ protocol sets the retransmission persistence too high, it increases transmission delay and jitter. In order to figure out the problem, the proposed scheme considers the number of frames in the buffer in controlling the retransmission persistence; it improves the throughput of ARQ protocol by increasing the retransmission persistence when the number of frames is small and decreasing otherwise. Simulation results show that the proposed scheme decreases the transmission delay and jitter significantly comparing to the existing schemes.

• Journal of KIISE
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• v.41 no.11
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• pp.974-981
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• 2014

These days, use of multimedia streaming service and demand of QoS (Quality of Service) improvement have been increased because of development of network. QoS of streaming service is influenced by a jitter, delay, throughput, and loss rate. For guaranteeing these factors which are influencing QoS, the role of transport layer is very important. But existing TCP which is a typical transport layer protocol increases the size of congestion window slowly and decreases the size of a congestion window drastically. These TCP characteristic have a problem which cannot guarantee the QoS of UHD multimedia streaming service. In this paper, we propose a router buffer based congestion control method for improving the QoS of UHD streaming services. Our proposed scheme applies congestion window growth rate differentially according to a degree of congestion for preventing an excess of available bandwidth and maintaining a high bandwidth occupied. Also, our proposed scheme can control the size of congestion window according to a change of delay. After comparing with other congestion control protocols in the jitter, throughput, and loss rate, we found that our proposed scheme can offer a service which is suitable for the UDH streaming service.

• Proceedings of the Korea Contents Association Conference
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• 2006.11a
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• pp.75-79
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• 2006

In a VoIP(Voice over IP) system which support the realtime speech service, speech quality is deteriorated by the delay, the jitter, the loss, and the reversed packet order. In this thesis, APBC for receiver site is proposed, which compensate the jitter by the adaptive playout algorithm and conceal the packet loss, and align the packet order. Also, a VoIP application system is implemented, and the performance of APBC is verified on the implemented system by measuring the processing speed and the speech quality. From the result, processing speed is 257$\mu$sec, which is fast enough to deal with packet being received in realtime. Also, the speech quality by MOS(Mean Opinion Score) is improved as 18 percent compared with algorithm of fixed playout delay.

• The KIPS Transactions:PartC
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• v.9C no.6
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• pp.953-960
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• 2002

Rel-time application programs have constraints which need to be met between media-data. It is client-leading synchronization that is absorbing variable transmission delay time and that is synchronizing by feedback control and palyout control. It is the important factor for playback rate and QoS if the buffer level is normal or not. This paper, The method of maintenance buffer normal state transmits in multimedia server by appling feedback of filtering function. And synchronization method is processing adaptive playout time for smooth presentation without cut-off while media frame is skip. When audio frame which is master media is in upper threshold buffer level we decrease play out time gradually, low threshold buffer level increase it slowly.