• Journal of KIISE:Information Networking
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• v.30 no.3
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• pp.437-447
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• 2003

In order to reduce the increasing packet loss rates caused by an exponential increase in network traffic, the IETF(Internet Engineering Task Force) is considering the deployment of active queue management techniques such as RED(Random Early Detection) algorithm. However, RED algorithm simple but does not protect traffic from high-bandwidth flows, which include not only flows that fail to use end-to-end congestion control such as UDP flow, but also short round-trip time TCP flows. In this paper, in order to solve this problem, we propose a simple fairness queue management scheme, called AFQM(Approximate Fair Queue Management) algorithm, that discriminate against the flows which submit more packets/sec than is allowed by their fair share. By doing this, the scheme aims to approximate the fair queueing policy Since it is a small overhead and easy to implement, AFQM algorithm controls unresponsive or misbehaving flows with a minimum overhead.

• The KIPS Transactions:PartC
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• v.9C no.2
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• pp.297-306
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• 2002

Continuous multimedia applications require a guaranteed retricval and transfer rate of streaming data, which conventional file server mechanism generally does not provide. In this paper we describe a dynamic negotiated admission control and dick bandwidth scheduling framework for Continuous Media (CM : e.g., video) servers. The framework consists of two parts. One is a reserve-based admission control mechanism and the other part is a scheduler for continuous media streams with dynamic resource allocation to achieve higher utilization than non-dynamic scheduler by effectively sharing available resources among contending streams to improve overall QoS. Using our policy, we could increase the number of simultaneously running: clients that coo]d be supported and cot]d ensure a good response ratio and better resource utilization under heavy traffic requirements.

• Journal of Korea Multimedia Society
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• v.7 no.5
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• pp.737-743
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• 2004

In this paper, we propose the call control scheme that can improve the capacity of the wireless system for the non-uniform traffic load distribution and the multiple types of services in multiple cells CDMA system. The number of mobile stations that can be served simultaneously in a base station is limited by the amount of total interference received in CDMA system. Further, the average number of mobile stations in each cell may not be uniformly distributed. Considering this factors, the call admission control scheme using the effective bandwidth concept is adapted in this paper. Thus, the bandwidth for a new call can be varied dynamically for reducing the blocking rate of new calls and the dropping rate of handoff calls. The suggested call control scheme is experimented through a simulation by dynamically assigning the bandwidth to new and handoff calls. The simulation results show that the proposed call control scheme can accommodate more mobile stations than the other methods in multiple cells environment.

• The KIPS Transactions:PartA
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• v.10A no.6
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• pp.691-700
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• 2003

Due to the fast development of wire&wireless internet and computer hardware, more and more internet services are being developed, such as Internet broadcast, VoD (Video On Demand), etc. So QoS (Qualify of Service) is essentially needed to guarantee the quality of these services. Traditional Internet is Best-Effort service in which all packets are transported in FIFO (First In First Out) style. However, FIFO is not suitable to guarantee the quality of some services, so more research in QoS router and QoS protocol are needed. Researched QoS router and protocol are high cost and inefficient because the existing infra is not used. To solve this problem, a new QoS control method, named Network Adaptive QoS, is introduced and applied to client/server-based streaming systems. Based on network bandwidth monitoring mechanism, network adaptive QoS control method can be used in wire&wireless networks to support QoS in real-time streaming system. In order to reduce application cost, the existing streaming service is used in NAQoS. A new module is integrated into the existing server and client. So the router and network line are not changed. By simulation in heavy traffic network conditions, we proved that stream cannot be seamless without network adaptive QoS method.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.23 no.12
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• pp.1343-1350
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• 2012

This paper presents a new method for designing a dual-band WIFI antenna using the third-order harmonic mode of a monopole antenna whose first-order mode operates at the low frequency band of WIFI. As analysing the current distribution of the third-order mode of this monopole antenna, the strongest point of electric field can be found. Then by attaching a stub at this point, the resonant frequency of the stub radiator can be adjusted from the third-order mode of the monopole antenna into the high frequency band of WIFI and the input impedance at this resonant frequency can be controlled with the width of the branch, without affecting the low frequency band of WIFI (the first-order mode of the monopole antenna). The compact dual-band antenna is designed at the size of an USB(universal serial bus) dongle and the bandwidth covers 600 MHz(2.3~3 GHz) at 2 GHz and 1 GHz(4.9~5.9 GHz) at 5 GHz under -10 dB which is satisfied with WLAN frequency. Efficiency of proposed antenna achieves over 50 % at WLAN frequency.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.11
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• pp.51-62
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• 2003

This paper presents a novel survivor memeory management and decoding techniques with sequential backward state transition control in the trace back Viterbi decoder. The Viterbi algorithm is an maximum likelihood decoding scheme to estimate the likelihood of encoder state for channel error detection and correction. This scheme is applied to a broad range of digital communication such as intersymbol interference removing and channel equalization. In order to achieve the area-efficiency VLSI chip design with high throughput in the Viterbi decoder in which recursive operation is implied, more research is required to obtain a simple systematic parallel ACS architecture and surviver memory management. As a method of solution to the problem, this paper addresses a progressive decoding algorithm with sequential backward state transition control in the trace back Viterbi decoder. Compared to the conventional trace back decoding techniques, the required total memory can be greatly reduced in the proposed method. Furthermore, the proposed method can be implemented with a simple pipelined structure with systolic array type architecture. The implementation of the peripheral logic circuit for the control of memory access is not required, and memory access bandwidth can be reduced Therefore, the proposed method has characteristics of high area-efficiency and low power consumption with high throughput. Finally, the examples of decoding results for the received data with channel noise and application result are provided to evaluate the efficiency of the proposed method.

• The KIPS Transactions:PartC
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• v.14C no.4
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• pp.349-358
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• 2007

Ethernet Passive Optical Network(EPON), which is one of PON technologies for realizing FTTx(Fiber-To-The-Curb/Home/Office), can cost-effectively construct optical access networks. In addition, EPON can provide high transmission rate up to 10Gbps and it is compatible with existing customer devices equipped with Ethernet card. To effectively control frame transmission from ONUs to OLT EPON can use Multi-Point Control Protocol(MPCP) with additional control functions in addition to Media Access Control(MAC) protocol function. For EPON, many researches on intra- and inter-ONU scheduling algorithms have been performed. Among the inter-ONU scheduling algorithms, IPS(Interleaved Polling with Stop) based on polling scheme is efficient because OLT assigns available time portion to each ONU given the request information from all ONUs. Since the IPS needs an idle time period on uplink between two consecutive frame transmission periods, it wastes time without frame transmissions. In this paper, we propose a dynamic bandwidth allocation algorithm to increase the channel utilization on uplink and evaluate its performance using simulations. The simulation results show that the proposed Two-phase Cycle Danamic Bandwidth Allocation(TCDBA) algorithm improves the throughput about 15%, compared with the IPS and Fast Gate Dynamic Bandwidth Allocation(FGDBA). Also, the average transmission time of the proposed algorithm is lower than those of other schemes.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.43 no.1 s.307
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• pp.27-38
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• 2006

The main purpose of rate control is to achieve the highest video quality when bandwidth or storage capacity is limited. For this purpose, we need a rate control algorithm which is adaptively controlled by the motion information of sequences, scene change, buffer capacity and time-varing bandwitdh channels. A rate-control method in the encoder requires the accurate estimation of target bit for each frame and the low end-to-end delay for transmitting video data by intelligent selection of encoding parameters. In this paper, we suggest three kinds of linear relation in the encoder to satisfy the characteristics of rate control. The first relation is that between the percentage of zero quantized transformed coefficients(p) and coded bits. Second relation is that between the PSNR of encoded frame and its Quantization parameter(QP). Finally, we can find out a linear approximation between QP and p. According to the experimental analysis, the proposed method results in an efficient rate control in terms of the bit estimation, the buffer capacity, and PSNR compared with the existing rate control in the H.264 JM 9.3.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.1
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• pp.26-39
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• 2007

In recent years, rate control is an important technique in real time video communication applications using H.264/AVC. Many existing rate control algorithms employ the quadratic rate-distortion model, which is determine the target bits for each P frame. In this paper, a new rate control algorithm for transmission of H.264/AVC video bit stream through CBR(Constant Bit Rate) channel is proposed. The proposed algorithm predicts an adaptive QP(Quantization Parameter) for improving video distortion, due to high motion and abruptly scene change, which target bit rate and MAD(Mean of Absolute Difference) for current frame considering image complexity variance between previous and current frames. Additionally, it uses frame skip technique to maintain bit stream within a manageable range and protect buffer from overflow or underflow. Experimental results show that the proposed method gives a quality improvement of about 0.5dB when compared to previous rate control algorithm. Also our proposed algorithm encodes the video sequences with less frame skipping compared to the existing rate control for H.264/AVC.

• The KIPS Transactions:PartC
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• v.13C no.2 s.105
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• pp.241-248
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• 2006

Mobile computing has such characteristics as portability, wireless network, mobility, etc. These characteristics cause various problems to mobile terminals like frequent disconnection, high error rate, and varying network status. These problems motivate us to develop an adaptive error control mechanism for supporting multimedia service in mobile computing environment. In this paper, we propose the Adaptive Error Control(AEC) scheme using client's buffer size and current error rate. After categorizing the status into four groups according to client's buffer size and current error rate, this scheme applies an appropriate error control scheme to each status. In this scheme, thresholds of buffer size and error rate are determined by the data transmission time, play rate and average VOP size, and by the probability of error for a sequence of packets. The performance of proposed scheme is evaluated by flaying MPEG-4 files on an experimental client/server environment, respectively. The results show that error correcting rate is similar to other schemes while the time for correcting error reduce a little. In addition, the size of data for correcting error is decreased by 23% compared with FEC and Hybrid FEC, respectively. Theses results demonstrate that the proposed scheme is more suitable in mobile computing environment with small bandwidth and varying environment than existing schemes.