• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.6
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• pp.1168-1176
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• 2003

Internet telephony service is very cheap and very easy to introduce the value-added service than the POTS, but is difficult to the QoS of telephone service. The existing internet typically offers "best effort" services only, which do not make any commitment about delay, packet loss and jitter. This paper compensates the low quality of the speech for packet loss or delay using FEAPT algorithm in Internet phone. In the FEAPT algorithm, the receiver expands the received packet under resonable threshold, and hence compensates the QoS of speech.of speech.

• 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.

• The Transactions of the Korea Information Processing Society
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• v.4 no.1
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• pp.224-236
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• 1997

In order to control the flow of traffic in ATM networks and optimize the usage of network resources, an appropriate control mechanism is necessary to cope with congestion or prevent degradation of network performance caused by congestion. While the conventional UPC algorithm provides only unstable preventive function unrelated with the state of networks and is unable to recover congestion, the proposed adaptive UPC algorithm supervises the situation of ATM networks using the information from OAM cell. Then the monitor of the proposed adaptive UPC algorithm controls leaky rate and buffer threshold value according to QOS. Therefore, the proposed algorithm can cope with congestion as well as prevent and react sensitively to buffer delay. In proportion to the diversity of traffic and the increase of transmission rate in networks, the adaptive UPC-BT algorithm proposed in this paper can be effectively used in ATM networks with wide applications. This paper shows that the proposed algorithm efficiently uses in ATM networks with application. This paper shows that the proposed algorithm efficiently use network resources and provides QOS to users for various kinds of traffics by comparing with conventional UPC algorithms.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.48 no.3
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• pp.28-35
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• 2011

Cooperative ARQ scheme is effective for better QoS guarantee for the next generation mobile communication systems where multimedia data transmission highly increases. In this paper, we propose a cooperative ARQ strategy in ad hoc cognitive relays for mobile multimedia communication for supporting instantaneous cooperation in MANET environment. In the proposed strategy, to support real time, delay-sensitive services, whenever a frame is transmitted from the source, each relay actively senses the SINR of the signal transmitted from the source, and determine whether to propose retransmission or not before the destination transmits feedback signal. To minimize the false retransmission decision or needless retransmission, we propose an adaptive sensing threshold optimization algorithm to maintain suboptimal sensing thresholds for each relay. By computer simulation, it is shown that the proposed cooperative ARQ retransmission scheme outperforms the conventional schemes with respect to frame transmission delay and frame loss probability in real time multimedia data transmission system.

• Journal of Information Processing Systems
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• v.10 no.2
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• pp.176-192
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• 2014

The realization of Wireless Multimedia Sensor Networks (WMSNs) has been fostered by the availability of low cost and low power CMOS devices. However, the transmission of bulk video data requires adequate bandwidth, which cannot be promised by single path communication on an intrinsically low resourced sensor network. Moreover, the distortion or artifacts in the video data and the adherence to delay threshold adds to the challenge. In this paper, we propose a two stage Quality of Service (QoS) guaranteeing scheme called Prioritized Multipath WMSN (PMW) for transmitting H.264 encoded video. Multipath selection based on QoS metrics is done in the first stage, while the second stage further prioritizes the paths for sending H.264 encoded video frames on the best available path. PMW uses two composite metrics that are comprised of hop-count, path energy, BER, and end-to-end delay. A color-coded assisted network maintenance and failure recovery scheme has also been proposed using (a) smart greedy mode, (b) walking back mode, and (c) path switchover. Moreover, feedback controlled adaptive video encoding can smartly tune the encoding parameters based on the perceived video quality. Computer simulation using OPNET validates that the proposed scheme significantly outperforms the conventional approaches on human eye perception and delay.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.12
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• pp.5800-5818
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• 2018

Interference may occur when several co-located wireless body area networks (WBANs) share the same channel simultaneously, which is compressed by resource scheduling generally. In this paper, a QoS-aware Adaptive Coloring (QAC) scheduling algorithm is proposed, which contains two components: interference sets determination and time slots assignment. The highlight of QAC is to determine the interference graph based on the relay scheme and adapted to the network QoS by multi-coloring approach. However, the frequent resource assignment brings in extra energy consumption and packet loss. Thus we come up with a launch condition for the QAC scheduling algorithm, that is if the interference duration is longer than a threshold predetermined, time slots rescheduling is activated. Furthermore, based on the relative distance and moving speed between WBANs, a prediction model for interference duration is proposed. The simulation results show that compared with the state-of-the-art approaches, the QAC scheduling algorithm has better performance in terms of network capacity, average delay and resource utility.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.12
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• pp.5835-5854
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• 2017

The implementation of IEEE 802.15.6 in Wireless Body Area Network (WBAN) is contention based. Meanwhile, IEEE 802.15.4 MAC provides limited 16 channels in the Superframe structure, making it unfit for N heterogeneous nature of patient's data. Also, the Beacon-enabled Carrier-Sense Multiple Access/Collision-Avoidance (CSMA/CA) scheduling access scheme in WBAN, allocates Contention-free Period (CAP) channels to emergency and non-emergency Biomedical Sensors (BMSs) using contention mechanism, increasing repetition in rounds. This reduces performance of the MAC protocol causing higher data collisions and delay, low data reliability, BMSs packet retransmissions and increased energy consumption. Moreover, it has no traffic differentiation method. This paper proposes a Low-delay Traffic-Aware Medium Access Control (LTA-MAC) protocol to provide sufficient channels with a higher bandwidth, and allocates them individually to non-emergency and emergency data. Also, a Contention Differentiated Adaptive Slot Allocation CSMA-CA (CDASA-CSMA/CA) for scheduling access scheme is proposed to reduce repetition in rounds, and assists in channels allocation to BMSs. Furthermore, an On-demand (OD) slot in the LTA-MAC to resolve the patient's data drops in the CSMA/CA scheme due to exceeding of threshold values in contentions is introduced. Simulation results demonstrate advantages of the proposed schemes over the IEEE 802.15.4 MAC and CSMA/CA scheme in terms of success rate, packet delivery delay, and energy consumption.

• Journal of Broadcast Engineering
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• v.17 no.1
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• pp.165-178
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• 2012

In most conventional video codecs, such as MPEG-2 and MPEG-4, inter coding is performed with the fixed motion vector resolution. When KTA software was developed, resolution for MVs can be selected in each slice. Although KTA codec uses a variety of resolutions for ME, the selected resolution is applied over the entire pixels in the slice and the statistical property of the local area is not considered. In this paper, we propose an adaptive decision scheme for motion vector resolution which depends on region, where MV search area is divided to multiple regions according to the distance from PMV. In each region, the assigned resolution is used to estimate MV. Each region supports different resolution for ME from other regions. The efficiency of the proposed scheme is affected from threshold values to divide the search area and the entropy coding method to encode the estimated MV. Simulation results with HM3.0 which is the reference software of HEVC show that the proposed scheme provides bit rate gains of 0.9%, 0.6%, and 2.9% in Random Access, Low Delay with B picture, and Low Delay with P picture structures, respectively.

• The Journal of the Acoustical Society of Korea
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• v.36 no.4
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• pp.285-291
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• 2017

Many methods have been studied to estimate the time delay between incoming signals to two receivers. In the case of the method based on the channel estimation technique, the relative delay between the input signals of the two receivers is estimated as an impulse response of the channel between the two signals. In this case, the characteristic of the channel has sparsity. Most of the existing methods do not take advantage of the channel sparseness. In this paper, we propose a time delay estimation method using BPD (Basis Pursuit Denoising) optimization technique, which is one of the sparse signal optimization methods, in order to utilize the channel sparseness. Compared with the existing GCC (Generalized Cross Correlation) method, adaptive eigen decomposition method and RZA-LMS (Reweighted Zero-Attracting Least Mean Square), the proposed method shows that it can mitigate the threshold phenomenon even under a white Gaussian source, a colored signal source and oceanic mammal sound source.

• Journal of Communications and Networks
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• v.5 no.3
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• pp.249-257
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• 2003

Ring networks are very commonly exploited among local area and metropolitan area networks (LAN/MAN), whereas cells or small fixed-size packets are widely used in synchronized ring networks. In this paper, we present an analytical method for evaluating the delay-throughput performance of a MAC protocol with multiple self-tokens in a slotted ring network under uniform traffic. In our analysis, we introduce the stationary probability, which indicates the number of packets in a node. Also, it is assumed that each node has a sufficiently large amount of self-tokens, and a slotted ring has the symmetry. The analytical results with respect to delay-throughput performance have similar values to computer simulated ones. Furthermore, in order to achieve fair access under non-uniform traffic, we propose an adaptive MAC protocol, where the number of self-tokens in a node dynamically varies, based on the number of packets transmitted within a specified period. In the proposed protocol, when the number of packets transmitted by a node within a specified period is larger than a specified threshold, the node decreases the number of self-tokens in a per-node distributed method. That results in creating free slots in the ring, thus all nodes can obtain an equal opportunity to transmit into the ring. Performance results obtained by computer simulation show that our proposed protocol can maintain throughput fairness under non-uniform traffic.