• The Journal of Korean Institute of Communications and Information Sciences
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• v.17 no.11
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• pp.1206-1218
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• 1992

Token ring systems which control to switch the data stream of networks by passing the token have been widely used to medium access controls in many ring or bus topology LANs. The system could be modeled for analysis as single-server-multi-queue system of the cyclic service method. These concepts could be expanded to multi-token ring systems interconnected with single ring consisting of bridges implemented simply to be stored and transmitted. In the proposal for the performance analysis of the interconnected token ring system, in has been assumed M/G/1 queueing model that frame arrivals are the Poisson process at each station queue and frame sizes are independently and identically distributed. And the average time delays were analyzed mathematically for arbitrary frame transferred from source station to destination area. The time delay of the frame transmission could be explained as the sum of the average time which the token passed from arbitrary position to source station, such as the waiting time in the source station transferring the previous arrival frames, and the propagation time from source station to interdestinated point. These delays were given as the sum of the duration from inner and outer bridge queues, the time delays from inner and outer bridge queues, and the time from outer bridge queue to destination station. These results were investigated by varing parameters effected to total time delays. In the results, those factors to be effected to dominant the total time delays were increased were in the cases of the high arrival rates and the high ration of destination of the other outerring. The system were shown the time delays increased exponentially in spite of the priority service policy. In order to decreasing the number of outerrings and increasing the number of nodes in backbone relatively, so the systems could be decreased the total delay in the interconnected token ring system.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5B
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• pp.457-465
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• 2003

In this paper, we analyze the performances of variable MAC (Medium Access Control) protocols and present an efficient MAC protocol for Ethernet PON (Passive Optical Network). We consider three MAC protocols: static TDMA, dynamic TDMA, and Interleaved polling. Static TDMA assigns an equal amount of bandwidth to all ONUs regardless of the request information but Dynamic TDMA dynamically allocates the bandwidth to each ONU considering its request. Interleaved Polling operates a cycle with variable time period and a polling method for informing a uplink transmission chance to each ONU. This paper theoretically analyzes the available bandwidth for each of three protocols. We also implement the simulation models for them by using OPNET and evaluates the performances under various bursty traffic environments. The results are compared and analyzed in terms of channel utilization and queueing delay.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.5
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• pp.892-900
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• 1994

Recently, ITU-TSS recommends a service specific connection oriented protocol (SSCOP) which assures the successful transmission of signalling information for B-ISDN. In the SSCOP, there is a mechanism to exchange periodically status informations for both error and flow controls. In this paper, we analyze the SSCOP with a discrete-time queueing model. We consider two cases for link-by-link and end-to-end controls. As the performance measures, we investigate delay, throughput, and buffer size of the receiver. The average values of these measures are analyzed with respect to the loss probability of messages, the traffic load, and the exchange period of status information. We also verify the accuracy of this model with simulations.

• Journal of Korean Institute of Industrial Engineers
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• v.25 no.2
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• pp.233-239
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• 1999

With the increase of Internet protocol (IP) packets the performance of routers became an important issue in internetworking. In this paper we examined the matching algorithm in gigabit router which has input queue with virtual output queueing. Port partitioning concept is employed to reduce the computational burden of the scheduler within a switch. The input and output ports are divided into two groups such that the matching algorithm is implemented within each input-output pair group in parallel. The matching is performed by exchanging input and output port groups at every time slot to handle all incoming traffics. Two algorithms, maximal weight matching by port partitioning (MPP) and modified maximal weight matching by port partitioning (MMPP) are presented. MMPP has the lowest delay for every packet arrival rate. The buffer size on a port is approximately 20-60 packets depending on the packet arrival rates. The throughput is illustrated to be linear to the packet arrival rate, which can be achieved under highly efficient matching algorithm.

• Journal of Korean Port Research
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• v.11 no.1
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• pp.45-64
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• 1997

Ship's delay caused by port congestion has drawn attention to the necessity for analysis on overall operation in port. But, in order to analyze the complicated port operation which contains large number of variable factors, queueing theory is needed to be adopted, which is applicable to a large scale transportation including ship's arrival in a large port. In this paper, a simulation model for Inchon Port was developed by the software SIMAN V and presented congestion rate under a certain scenario regarding the arrival ship's number and service levels. To develop the simulation model, types of ships and cargoes during the 1995 in Inchon Port was analyzed in advance. The results of the simulation can be summarized as follows : In order to maintain present levels of congestion rate and time with the increasing number of arrival vessels, service rate should be increased at an exponential rate. To improve the current congestion effectively, part of cargoes are needed to be transferred to a newly developing port. Results obtained from simulation can be used properly to prepare improved service levels and to plan appropriate investment strategies.

• The Journal of the Korea Contents Association
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• v.18 no.3
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• pp.34-40
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• 2018

In this study, the optimal buffer size is calculated for seamless video playback on a mobile device. Buffer means the memory space for multimedia packet which arrives in mobile device for video play such as VOD service. If the buffer size is too large, latency time before video playback can be longer. However, if it is too short, playback service can be paused because of shortage of packets arrived. Hence, the optimal buffer size insures QoS of video playback on mobile devices. We model the process of buffering into a discret-time queueing model. Mean busy period length and mean waiting time of Geo/G/1 queue with N-policy is analyzed. After then, we uses the main performance measures to present numerical examples to decide the optimal buffer size on mobile devices. Our results enhance the user satisfaction by insuring the seamless playback and minimizing the initial delay time in VOD streaming process.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.6
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• pp.1065-1071
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• 2009

Satellite application services can be divided into realtime services like voice communications and non-real time services like traditional data communications. To support both types of services on the same On-Board Switching(OBS), a scheduler which depends on their service classes is required. A fixed priority scheduling policy has a starvation problem. In this paper we propose a scheduling scheme based on premium and age. Premium is a fixed value which is given to a certain class of services. Age is another parameter of the scheduling policy and it will be increased by one for every scheduling cycle. The scheme we propose chooses a packet which has the largest sum of its age and premium. Simulation results indicate that the proposed approach shows better performance in both average cell delay and std-dev of cell delay for the lower class of service. There is no staying in infinite starvation state.

• Journal of Digital Contents Society
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• v.10 no.2
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• pp.357-365
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• 2009

To effectively exploit the underlying network bandwidth while maximizing user perceivable QoS, mandatory to make proper estimation on packet loss and queuing delay of the underling network. This issue is further emphasized in wireless network environment where network bandwidth is scarce resource. In this work, we focus our effort on developing performance model for wireless network. We collect packet trace from actually wireless network environment. We find that packet count process and bandwidth process in wireless environment exhibits long range property. We extract key performance parameters of the underlying network traffic. We develop an analytical model for buffer overflow probability and waiting time. We obtain the tail probability of the queueing system using Fractional Brown Motion (FBM). We represent average queuing delay from queue length model. Through our study based upon empirical data, it is found that our performance model well represent the physical characteristics of the IEEE 802.11b network traffic.

• Journal of Korea Multimedia Society
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• v.2 no.3
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• pp.329-338
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• 1999

In this paper we propose a WATM MAC protocol called DTDD/PS (Dynamic Time Division Duplex using Poll/Select scheme). The idea of the DTDD/PS protocol is based that a radio cell is considered as queueing system. The MAC controller in base station is charged in coordinating terminals for the QoS requirement from each terminal. For that reason, terminals send it's own dynamic parameter represented its status, and controller controls the access of terminals and assigns time slots dynamically using received parameters. Therefore this system can satisfy the QoS of real-time service in wireless environments. To evaluate DTDD/PS protocol simulations are executed. In 25Mbps channel capacity, the voice and data coexist traffic situation, as the total number of terminal is increased, the delay of data terminals is increased rapidly. But the case of only increasing data terminals, the number of supported voice terminals is not varying. Therefore we can expect that DTDD/PS protocol can applied to multimedia traffic effectively.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.6
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• pp.17-29
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• 1999

In this paper, the prioritized queueing handoff scheme in CDMA (Code Division Multiple Access) cellular system is proposed. Also, the analytical survey for the proposed scheme is carried out, and the performance of this scheme is compared with that of non prioritized scheme and FIFO (First In First Out) queue scheme by computer simulation. The handoff region is defined as the time between the handoff treshold and the receiver threshold, and it is used for the maximum queue waiting time in the proposed scheme. The handoff and the receiver thresholds are defined as rewpectively: 1) the time that the Pilot Strength Measurement Message in the neighbor in the neighbor cell is received to the BS (Base Station) under the T_ADD threshold; and 2) the time that the T_DROP timer is expired and the Pilot Strength Measurement Message in the current cell is received to the BS under the T_DROP threshold. The performance metrics for analyzing the proposed scheme are : 1) probability of forced termination; 2) probability of call blocking; 3) ratio of carried traffic to total offered load; 4) average queue size; 5) average handoff delay time in queue. The simulation results show that the proposed scheme maintains high performance for handoff requests at a small penalty in total system capacity.