• ETRI Journal
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• v.33 no.1
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• pp.136-139
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• 2011

A modular upgrade design for packet transport switch nodes is presented where packet loss is dramatically reduced by intra-module and inter-module buffer sharing. This modular design offers significant cost and power reduction in a high-data-rate system where buffers are highly costly and power-greedy.

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2327-2330
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• 2003

This paper deals with face detection and recognition using ellipsodal information and wavelet packet analysis. We proposed two methods. First, Face detection method uses general ellipsodal information of human face contour and we find eye position on wavelet transformed face images A novel method for recognition of views of human faces under roughly constant illumination is presented. Second, The proposed Face recognition scheme is based on the analysis of a wavelet packet decomposition of the face images. Each face image is first located and then, described by a subset of band filtered images containing wavelet coefficients. From these wavelet coefficients, which characterize the face texture, the Euclidian distance can be used in order to classify the face feature vectors into person classes. Experimental results are presented using images from the FERET and the MIT FACES databases. The efficiency of the proposed approach is analyzed according to the FERET evaluation procedure and by comparing our results with those obtained using the well-known Eigenfaces method. The proposed system achieved an rate of 97％(MIT data), 95.8％(FERET databace)

• Proceedings of the KSME Conference
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• 2008.11a
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• pp.832-837
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• 2008

A modeling method for the modal analysis of a multi-packet blade system having tapered cross section undergoing rotational motion is presented in this paper. Blades are idealized as tapered cantilever beams that are fixed to a rotating disc. The stiffness coupling effects between blades due to the flexibilities of the disc and the shroud are modeled with discrete springs. Hybrid deformation variables are employed to derive the equations of motion. To obtain more general information, the equations of motion are transformed into dimensionless forms in which dimensionless parameters are identified. The effects of the dimensionless parameters including tapered ratio and the number of packets as well as blades on the modal characteristics of the system are investigated with some numerical examples.

• Journal of Broadcast Engineering
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• v.14 no.6
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• pp.796-806
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• 2009

SVC (Scalable Video Coding) is a new video coding standard to provide convergence media service in heterogeneous environments with different networks and diverse terminals through spatial-temporal-quality combined flexible scalabilities. This paper presents the performance analysis on packet loss in the delivery of SVC over IP networks and an efficient adaptation method to packet loss caused by buffer overflow. In particular, SVC with MGS (Medium Grained Scalability) as well as spatial and temporal scalabilities is addressed in the consideration of packet-based adaptation since finer adaptation is possible with a sufficient numbers of quality layers in MGS. The effect on spatio-temporal quality due to the packet loss of SVC with MGS is evaluated. In order to minimize quality degradation resulted by packet loss, the proposed adaptation of MGS based SVC first sets adaptation unit of AU (Access Unit) or GOP corresponding to allowed delay and then selectively discards packets in order of importance in terms of layer dependency. In the experiment, the effects of packet loss on quantitative qualities are analyzed and the effectiveness of the proposed adaptation to packet loss is shown.

• The Journal of the Korea institute of electronic communication sciences
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• v.8 no.1
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• pp.71-76
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• 2013

In this paper, end-to-end performances for some VoIP traffics are measured and analyzed over MANET(Mobile Ad-hoc Networks) with packet aggregation transmission under blackhole attacks. A VoIP simulator based on NS-2 is used for performance measure. In the simulation, MOS, call connection ratio, network delay and packet loss rate are measured for end-to-end transmission performance. With analysis of measured data, characteristics of each traffics for performance parameters of packet aggregation transmission is showed on MANET under blackhole attack, As a results of this paper, some considerations be required for packet aggregation transmission are suggested to implement VoIP services on MANET under blackhole attacks.

• Journal of Korea Multimedia Society
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• v.7 no.2
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• pp.204-210
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• 2004

This paper proposes a transmission probability control scheme and MAC protocol for packet data services in slotted CDMA system. In slotted CDMA system, multiple access interference is the major factor of unsuccessful packet transmissions. Therefore, in order to obtain the optimal system throughput, the number of simultaneously transmitted packets should be kept at a proper level. In the proposed protocol, the base station calculates the packet transmission probability of mobile stations according to the offered load and then broadcasts it. Mobile stations, which have a packet, attempt to transmit packet with the received probability. Numerical analysis and simulation results show that the proposed scheme can offer better system throughput than the conventional one, and guarantee a good fairness among all mobile stations regardless of the offered load.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1999.05a
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• pp.126-130
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• 1999

In packet CDMA networks, it is important to design a MAC protocol that meets the QoS requirements , of the different traffic types and allocates the radio channels efficiently. In this paper, a RRS$^2$-CDMA MAC protocol is proposed for integrating voice and data services in the microcellular packet CDMA networks. In PRS$^2$-CDMA, a voice terminal can resolve a spreading code to transmit voice packets during a talkspurt while a data terminal has to contend for a code for each packet transmission. The numerical results show that the proposed protocol can improve the system capacity, while guaranteeing the QoS of voice and data services.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.660-666
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• 2004

Overall TCP performance represented by end-to-end throughput is largely dependent upon its loss recovery performance. In particular non-congestion packet losses caused by transmission errors degrade TCP performance seriously. Using Markov process, we analyze TCP loss recovery performance for correlated packet losses caused by multipath fading. The results show that loss recovery performance can be severely affected by burstiness in packet losses, even if overall packet loss ratio is very low.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.8
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• pp.2090-2095
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• 1998

In this paper, we measured voice packet loss over the Korean Internet and analyzed percentage articulation by variation of the packet loss. To do this, we reviewed real-time transmission service based on RTP/UDP/IP and test method of the transmission quality. and implemented the real-time speech transmission system using GSM and UDP/IP. Monosyllable list has been chosen for the percentage articulation test, each voice packet has been coded and compressed by GSM and it has sequence number to measured packet loss and to recover out-of-order packets. In transmission results using seven router over the Korean Internet, we have show that loss rates reached 1.6% (unload), 22.5%(load) and loss rates after packet recovery by resequencing and FEC are from 9% to 35%. Finally, we have shown that percentage articulations by variation of the network traffic are Table 4.

• The Journal of the Korea institute of electronic communication sciences
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• v.10 no.2
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• pp.289-296
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• 2015

Wireless multimedia sensor network senses and transfers mass multimedia data. Also, it is sensitive to latency. This thesis proposes a routing technique based on traffic priority in order to improve the network efficiency by minimizing latency. In addition, it proposes a congestion control mechanism that uses packet service time, packet inter-arrival time, buffer usage, etc. In this thesis, we verified the reduction of packet latency in accordance with the quality level of packet as a result of the performance analysis through the simulation method. Also, we verified that the proposed mechanism maintained a reliable network state by preventing packet loss due to network overload.