• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.11B
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• pp.946-955
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• 2004

It is shown that the do Bruijn graph (dBG) can be used as an architecture for interconnection network and a suitable structure for parallel computation. Recent works have classified dBG based routing algorithms into shortest path routing and fault tolerant routing but investigation into fault free shortest path (FFSP) on dBG has been non-existent. In addition, as the size of the network increase, more faults are to be expected and therefore shortest path dBG algorithms in fault free mode may not be suitable routing algorithms for real interconnection networks, which contain several failures. Furthermore, long fault free path may lead to high traffic, high delay time and low throughput. In this paper we investigate routing algorithms in the condition of existing failure, based on the Bidirectional do Bruijn graph (BdBG). Two FFSP routing algorithms are proposed. Then, the performances of the two algorithms are analyzed in terms of mean path lengths and discrete set mean sizes. Our study shows that the proposed algorithms can be one of the candidates for routing in real interconnection networks based on dBG.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.9B
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• pp.751-759
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• 2008

Carrier Ethernet backbone network integrates distributed layer-2 based metro networks. In this networks, Multiple Spanning Tree Protocol (MSTP) has been uscd as a main routing protocol that allows multiple spanning trees in a network. A better routing protocol called IEEE802.1aq - Shortest Path Bridging (SPB) is recently proposed, that generates the shortest spanning tree per a destination node. As SPB provides a routing path per a destination node, there is no way to adapt network traffic at normal condition. If we are free from the principle of "a spanning tree per a destination node", we can achieve adaptive routing. Based on this philosophy, we propose a new spanning tree based protocol - Edge Node Divided Spanning Tree (ENDIST). ENDIST divides an edge node into sub-nodes as many as connecting links from the node and each sub-node generates a single shortest path tree based on SPB. Depending on network or nodal status, ENDIST chooses a better routing path by flow-basis. This added traffic engineering ability contributes to enhanced throughput and reduced delay in backbone networks. The simulation informs us that ENDIST's throughput under heavy load performs about 3.4-5.8 and 1.5-2.0 times compared with STP's and SPB's one respectively. Also, we verified that ENDIST's throughput corresponds to the theoretical upper bound at half of cases we investigated. This means that the proposed ENDIST is a dramatically enhanced and the close-to-perfect spanning tree based routing schemes.

• Journal of Korean Society of Transportation
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• v.31 no.4
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• pp.32-46
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• 2013

Because of short length between interchanges, traffic congestion has been caused by a lot of short trip traffic in the urban section of interregional expressway. Also, in case of expressway with Bus Only Lane, bus causes dangerous situation by conflicting between vehicles and buses in order to access from Bus Only Lane to IC. Therefore, this study aims to propose Express Lane applied by Bus Only Lane as a new lane operation in order to efficiently operate and to analyze the effectiveness on constructing Express Lane to the Pangyo-Hannam section in Gyeongbu Expressway. The study analyzes effectiveness by scenario using the micro traffic simulation tool, VISSIM 5.4. For this, the indicators which evaluate construction effect of Express Lane, are used by average delay time, average speed and total travel time. The results of the study showed the best effectiveness when Express Lane installed one, and it represented an ideal traffic condition when long distance trip traffic of the whole traffic is 25% in the study section. Therefore, the construction of Express Lane using Bus Only Lane achieves positive effectiveness on safety and expense aspect as well as uncongested flow.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.1
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• pp.105-113
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• 2013

RAID systems have been widely used by connecting several disks in parallel structure. to resolve the delay and bottleneck of data I/O. Recently, SSD based RAID systems are emerging since SSDs have better I/O performance than HDD. However, endurance and power consumption problems due to frequent write operation in SSD based RAID system should be resolved. In this paper, we propose a de-duplication method of parity disk in SSD based RAID system with expensive update cost. The proposed method segments chunk of parity data into small pieces and removes duplicate data, therefore, it can reduce wear-leveling and power consumption by decreasing write operation for duplicated parity data. Experimental results show that bit update rate of the proposed method is 16% in total disk, 31% in parity disk less than that of existing method in RAID-6 system using EVENODD erasure code, and the power consumption of the proposed method is 30% less than that of existing method. Besides the proposed method is 12% in total disk, 32% in parity disk less than that of existing method in RAID-5 system, and the power consumption of the proposed method is 36% less than that of existing method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9B
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• pp.1742-1754
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• 1999

In this paper, we propose a rate control scheme, using histogram based rate-distortion (R-D) estimation, which produces a consistent picture quality between consecutive frames. The histogram based R-D estimation used in our rate control scheme offers a closed-form mathematical model that enable us to predict the bits and the distortion generated from an encoded frame at a given quantization parameter (QP) and vice versa. The most attractive feature of the R-D estimation is low complexity of computing the R-D data because its major operation is just to obtain a histogram or weighted histogram of DCT coefficients from an input picture. Furthermore, it is accurate enough to be applied to the practical video coding. Therefore, the proposed rate control scheme using this R-D estimation model is appropriate for the applications requiring low delay and low complexity, and controls the output bit-rate ad quality accurately. Our rate control scheme ensures that the video buffer do not underflow and overflow by satisfying the buffer constraint and, additionally, prevents quality difference between consecutive frames from exceeding certain level by adopting the distortion constraint. In addition, a consistent considering the maximum tolerance BER of the voice service. Also in Rician fading channel of K=6 and K=10, considering CLP=$10^{-3}$ as a criterion, it is observed that the performance improment of about 3.5 dB and 1.5 dB is obtained, respectively, in terms of $E_b$/$N_o$ by employing the concatenated FEC code with pilot symbols.

• The Transactions of the Korea Information Processing Society
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• v.7 no.8
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• pp.2484-2496
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• 2000

High speed switches have been developing to interconnect a large number of nodes. It is important to analyze the switch performance under various conditions to satisfy the requirements. Queueing analysis, in general, has the intrinsic problem of large state space dimension and complex computation. In fact, The petri net is a graphical and mathematical model. It is suitable for various applications, in particular, manufacturing systems. It can deal with parallelism, concurrence, deadlock avoidance, and asynchronism. Currently it has been applied to the performance of computer networks and protocol verifications. This paper presents a framework for modeling and analyzing ATM switch using stochastic activity networks (SANs). In this paper, we provide the ATM switch model using SANs to extend easily and an approximate analysis method to apply A TM switch models, which significantly reduce the complexity of the model solution. Cell arrival process in output-buffered Queueing A TM switch with finite buffer is modeled as Markov Modulated Poisson Process (MMPP), which is able to accurately represent real traffic and capture the characteristics of bursty traffic. We analyze the performance of the switch in terms of cell-loss ratio (CLR), mean Queue length and mean delay time. We show that the SAN model is very useful in A TM switch model in that the gates have the capability of implementing of scheduling algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.11 no.3
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• pp.179-196
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• 1986

In this paper, the peformance of a statistical packet voice/data multiplexer is studied. In ths study we assume that in the packet voice/data multiplexer two separate finite queues are used for voice and data traffics, and that voice traffic gets priority over data. For the performance analysis we divide the output link of the multiplexer into a sequence of time slots. The voice signal is modeled as an (M+1) - state Markov process, M being the packet generation period in slots. As for the data traffic, it is modeled by a simple Poisson process. In our discrete time domain analysis, the queueing behavior of voice traffic is little affected by the data traffic since voice signal has priority over data. Therefore, we first analyze the queueing behavior of voice traffic, and then using the result, we study the queueing behavior of data traffic. For the packet voice multiplexer, both inpur state and voice buffer occupancy are formulated by a two-dimensional Markov chain. For the integrated voice/data multiplexer we use a three-dimensional Markov chain that represents the input voice state and the buffer occupancies of voice and data. With these models, the numerical results for the performance have been obtained by the Gauss-Seidel iteration method. The analytical results have been verified by computer simylation. From the results we have found that there exist tradeoffs among the number of voice users, output link capacity, voic queue size and overflow probability for the voice traffic, and also exist tradeoffs among traffic load, data queue size and oveflow probability for the data traffic. Also, there exists a tradeoff between the performance of voice and data traffics for given inpur traffics and link capacity. In addition, it has been found that the average queueing delay of data traffic is longer than the maximum buffer size, when the gain of time assignment speech interpolation(TASI) is more than two and the number of voice users is small.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.6A
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• pp.533-539
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• 2002

It is well known the fact that turbo lodes has better performance as the number of iteration and the interleaver size increases in the AWGN channel environment. However, as the number of iteration and the interleaver size are increased, it is required much delay and computation for iterative decoding, and caused the flattening effect phenomenon which is very litter BER performance improvement at the arbitrary SNR. In this paper, We proposed the new asymmetric turbo codes, which consist of parallel concatenated turbo codes that use mixed types of component codes with different not only constraint length but also generate polynomial and analyzed its BER performance for log-MAP decoding algorithm with frame size of 128, 256, 512 and 1024 bits, and coding rate of 1/3. As a results of simulation, proposed asymmetric turbo codes verify that its BER performance is superior to conventional symmetric turbo codes. It can be also observed that the flattening effect phenomenon is very reduced by applying the proposed asymmetric turbo codes. It gains respectively 1.7dB ~2.5dB and 2.0dB~2.5dB SNR improvements in the case of short frame(128, 256) and large frame(512, 1024) size for the BER $10_{-4}$>/TEX> region.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2B
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• pp.279-288
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• 2004

Rapidly emerging P2P(Peer to Peer) applications generate very bursty traffic, which gives a lot of burden to network, and the amount of such traffic is increasing rapidly. Thus it is becoming more important to understand the characteristics of such traffic and reflect it when we design and analyze the network. To do that we measured the traffic in a campus network and present flow statistics and traffic models of the measured traffic, and compare them with those of the web traffic. The results indicate that P2P traffic is much burstier than web traffic and as a result it negatively affects network performance. We modeled P2P traffic using self-similar traffic model to predict packet delay and loss occurred in network which are very important to evaluate network performance. We also predict queue length distribution and loss probability in SSQ(Single Sewer Queue). To assess accuracy of traffic model, we compare the SSQ statistics of traffic models with that of the traffic trace. The results show that self-similar traffic models we use can predict P2P traffic behavior in network precisely. It is expected that the traffic models we derived can be used when we design network capacity and predict network performance and QoS of the P2P applications.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38A no.1
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• pp.68-78
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• 2013

In this paper, we propose an early spectrum sensing(ESS) as an advance preparation for radio-access trial, which enables multi-mode terminals to access the most appropriate radio-access system in a cloud-conceptual base station system where multiple radio access technologies(RATs) coexist. Prior to a random access to one of RATs, a multi-mode terminal conducts a spectrum sensing over entire frequency bands of whole RATs, then select the RAT with the lowest sensing power, that is likely to have the most available spectrum. Thus, an access failure caused by that the selected RAT has no available radio spectrum could be avoidable in advance. In computer simulation, we consider as various RATs as possible. First, circuit and packet systems are taken into consideration. In addition, the packet systems are classified according to the feasibility of carrier aggregation(CA). In case of terminal, three modes are considered with circuit-only, packet-only, and multi-mode. Subsequently, packet traffic is classified into real-time and non-real-time traffic with three different tolerable delay levels. The simulation includes a call process starting with a call generation and ending up with a resource allocation reflecting individual user's QoS requirements and evaluates the proposed scheme in terms of the successful access probability, system access time, system balancing factor and packet loss probability.