• Journal of information and communication convergence engineering
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• v.8 no.6
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• pp.721-724
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• 2010

In a network router, packet loss may occur when it overflows due to sudden burst traffic. This paper studies how much large buffers are required to eliminate the packet losses. There are buffers on which packet arrive and one output port to which a packet is transmitted at a time. The buffer management algorithm should determine one of the buffers whose packet is transmitted to the output port at each time. The front packet belonging to the buffer determined by the algorithm is transmitted. The goal is to minimize the sum of the lengths of buffers to transmit all the packets. We provide an optimal off-line algorithm and also we show the lower bounds of on-line algorithms. The on-line algorithm has no prior information of the packets having arrived in the future. Its performance is compared to that of the optimal off-line algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.7
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• pp.1403-1410
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• 1997

Because streaming mode connection server suitable to wide area ATM networks performs transmission, reception and lookup with time restriction for the transmission time of a cell, it has demerits of large cell loss incase that burst traffic occurs. Therefore, in this paper to decrease cell loss we propose a high speed CAM (Content Addressable Memory) which is capable of processing data of streaming mode connections server at a high speed. the proposed CAM is applied to forwarding table VPC map which performs function to output connection numbers about input VPI(Virtual Path Identifier)/VCI(Virtual Channel Identifier). The designed high speed CAM consist of DBL(Dual Bit Line) CAM structure performed independently write operation and match operation and two-port SRAM structure. Also, its simulation verification and full-custom layout is performed by Hspice and Composs tools in 0.8 .$\mu$m design rule.

• Journal of Korean Institute of Industrial Engineers
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• v.37 no.3
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• pp.229-233
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• 2011

The existing approaches for analyzing the data service capacity of CDMA2000 1X EV-DO (Evolution-Data Only) systems have focused on deriving the maximum number of active customers assuming that the data rates provided to the customers can be determined beforehand. However, we have seen that through real experiment, the data rates actually provided to the customers adaptively change according to the number of active customers. In this paper, we propose new capacity analysis model considering the data rate adaptation. Numerical examples are also presented to obtain the maximum input traffic rates for various values of the length of data burst and the required mean delay bound.

• The Journal of the Korea institute of electronic communication sciences
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• v.2 no.1
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• pp.10-18
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• 2007

In this paper, we extend the multiple time scale control framework to window-based congestion control, in particular, TCP. This is performed by interfacing TCP with a large time scale control module which adjusts the aggressiveness of bandwidth consumption behavior exhibited by TCP as a function of "large time scale" network state. i.e., conformation that exceeds the horizon of the feedback loop as determined by RTT. Performance evaluation of multiple time scale TCP is facilitated by a simulation bench-mark environment which is based on physical modeling of self-similar traffic. If source traffic is not extended exceeding, when RTT is 450ms, in self similar burst environment, performance gain of TCP-SSC is up to 45% for ${\alpha}$=1.05. However, its is acquired only 20% performance gain for ${\alpha}$=1.95 relatively. Therefore we showed that by TCP-MTS at large time scale into a rate-based feedback congestion control, we are able to improve two times performance significantly.

• Journal of Internet of Things and Convergence
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• v.8 no.1
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• pp.53-58
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• 2022

Smoothing is a transmission plan that converts video data stored at a variable bit rate into a constant bit rate. In the study of [6-7], when a data rate increase is required, the frame with the smallest increase is set as the start frame of the next transmission rate section, when a data tate decrease is required. the frame with the largest decrease is set as the start frame of the next transmission rate section, And the smoothing algorithm was proposed and performance was evaluated in an environment where network traffic is not considered. In this paper, the smoothing algorithm of [6-7] evaluates the adaptive CBA algorithm and performance with minimum frame rate, average frame rate, and frame rate variation from 512KB to 32MB with E.T 90 video data in an environment that considers network traffic. As a result of comparison, the smoothing algorithm of [6-7] showed superiority in the comparison of the minimum refresh rate.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.6
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• pp.205-212
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• 2003

Because a wireless ad hoc network does not have fixed infrastructure, a call admission control approach researched in a wireless network is not feasible to this network. In this paper, we propose call admission control scheme to support this problem and the burst handoff traffic due to group mobility in a wireless ad hoc network. This scheme is an adaptive guard channel scheme which adapt the number of guard channels in each MBS(mobile base station) according to the current estimate of the potential handoff call rate derived from the number of ongoing calls within the coverage area of an MBS that initiate group handoff in a wireless ad hoc network. Our simulation studies are performed for comparisons of the proposed scheme with the other channel allocation schemes. Simulation results show that the proposed scheme efficiently reduces handoff call blocking probability in wireless ad hoc networks.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.26 no.4
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• pp.562-570
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• 2022

The supply expansion of 5G services and personal smart devices has caused the sharp increase of data traffic and the demand of various services. Again, these facts have resulted in the huge demand of network bandwidth. However, existing network technologies using electronic signal have reached the limit to accommodate the demand. Therefore, in order to accept this request, optical internet has been studied actively. However, optical internet still has a lot of problems to solve, and among these barriers a very urgent issue is to develop QoS technologies. Hence, in order to achieve service differentiation between classes in optical internet, especially in OBS network, a new QoS method automatically tuning the size of data bursts is proposed in this article. Especially, the algorithm suggested in this article is based on fiber delay line.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.38 no.12
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• pp.1-9
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• 2001

In the APON(ATM Passive Optical Network), the transmission of the upstream traffic is based on a TDMA(Time Division Multiple Access) method that an OLT(Optical Line Termination) permits ONUs(Optical Network Units) sending cells by allocating time slots. Because the upstream is not a streaming mode, the cell synchronizer has to be operated in the burst mode. Also, the cell phase monitor is required to prevent collisions between cells which are transmitted by multiple ONUs through a single optical fiber. In this paper, a TDMA burst cell synchroniser is implemented with the FPGA(Field Programmable Gate Array) being used in the APON based on G.983.1 for transmitting upstream cells. It has two main functions which are the upstream data recovery and the phase monitoring. The former is to recover the upstream data and clock in the OLT by seeking the preamble which is the overhead of the upstream time slot and by aligning the phase of the bit and cell with the system clock. The latter is to provide the information to the ONU to compensate for the equalization delay by monitoring continuously the phase difference between adjacent cells to avoid the cell collision on the upstream.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.12
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• pp.46-58
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• 2003

The recent Internet is changing from its best-effort service and needs the network structure and traffic engineering, which can support various applications requiring differentiated traffic processing and high quality of service. The extension of the Internet from wired to wireless system generating location-dependent and burst errors make more difficult to support a good service with the existing packet scheduling algorithm. In this paper, we propose a wireless diffserv packet scheduling (WDSPS) algorithm that can provide reliable service in the differentiated wireless internet service network. The proposed scheduling algorithm solves the HOL(head of line) blocking problems of class queue occurred in wireless network, supports the differentiated service for each class which is defined in differentiated service networks and makes possible the gradual and efficient service compensation not only among classes but also among flows to prevent the monopoly of one class or one flow. Through simulations, we show that our proposed WDSPS scheduling algorithm can provide the required QoS differentiation between classes and enhance the service throughput in various wireless network conditions.

• Journal of Communications and Networks
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• v.10 no.4
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• pp.412-421
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• 2008

We observe simultaneous transmission of relay stations (RSs) allowed in current IEEE 802.16j draft standard for multi-hop relay networks may involve severe interference among the RSs, hence leading to throughput degradation. Allowing only 1/3 of the RSs to simultaneously transmit instead of 1/2 RSs as in the current draft standard reduces the interference but results in reduced throughput. To remedy this problem, we devise schemes to incorporate network coding at link-layer level (decode-and-forward) into the simultaneous transmission of RSs. Data movement is rearranged to maximize coding gain. Formula is derived to dictate exact movement of packets traveling between base station (BS) and mobile stations (MSs) via intermediate RSs. The frame structure in the current IEEE 802.16j draft standard does not allow broadcast needed for network coding. We devise a new frame structure which supports the broadcast. A new R-MAP (pointers to the burst data) is introduced to implement the broadcast. Since our new frame structure is used only for BS to RS or RS to RS communication, our schemes retain backward compatibility with legacy MSs based on IEEE 802.16e standard. Simulation based on simple configuration of RSs shows considerable improvement in terms of system throughput and round trip delay. For a 4-hop relay network with 1 BS and 4 RSs with symmetric traffic in uplink (UL) and downlink (DL), throughput is improved by 49% in DL and by 84% in UL traffic compared with IEEE 802.16j draft standard under the assumption that omni-directional antennae are used in BS and RSs.