• Journal of the Korea Society for Simulation
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• v.24 no.3
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• pp.27-35
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• 2015

We analyze an M/G/1/K queueing system with queue-length dependent service and arrival rates. There are a single server and a buffer with finite capacity K including a customer in service. The customers are served by a first-come-first-service basis. We put two thresholds $L_1$ and $L_2$($${\geq_-}L_1$$ ) on the buffer. If the queue length at the service initiation epoch is less than the threshold $L_1$, the service time of customers follows $S_1$ with a mean of ${\mu}_1$ and the arrival of customers follows a Poisson process with a rate of ${\lambda}_1$. When the queue length at the service initiation epoch is equal to or greater than $L_1$ and less than $L_2$, the service time is changed to $S_2$ with a mean of $${\mu}_2{\geq_-}{\mu}_1$$. The arrival rate is still ${\lambda}_1$. Finally, if the queue length at the service initiation epoch is greater than $L_2$, the arrival rate of customers are also changed to a value of $${\lambda}_2({\leq_-}{\lambda}_1)$$ and the mean of the service times is ${\mu}_2$. By using the embedded Markov chain method, we derive queue length distribution at departure epochs. We also obtain the queue length distribution at an arbitrary time by the supplementary variable method. Finally, performance measures such as loss probability and mean waiting time are presented.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.39 no.1
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• pp.20-33
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• 2002

Usage of ATM AAL2 packets becomes dominant to increase transmission efficiency of voice traffic in the backbone network. In case of voice service that uses AAL2 mechanism, if resources of network are enough, connection of new call is accepted. However, due to packets generated by the new call, transmission delay of packets from old calls can increase sharply. To control this behavior, in this paper we present an AAL2 buffer management scheme that allocates a virtual buffer to each call and after calculating its propagation delay variation(PDV), decides to drop packets coming from each call according to the PDV value. We show that this packet dropping algorithm can effectively prevent abrupt QoS degradation of old calls. To do this, we analyze AAL2 packet composition process to find a critical factor in the process that influences the end-to-end delay behavior and model the process by K-policy M/D/1 queueing system and MIN(K, Tc)-policy M/D/1 queueing system. From the mathematical model, we derive the probability generating function of AAL2 packets in the buffer and mean waiting time of packets in the AAL2 buffer. Analytical results show that the AAL2 packet dropping algorithm can provide stable AAL2 packetization delay and ATM cell generation time even if the number of voice sources increases dramatically. Finally we compare the analytical result to simulation data obtained by using the COMNET Ⅲ package.

• The Transactions of the Korea Information Processing Society
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• v.6 no.4
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• pp.1058-1070
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• 1999

In this paper, we propose a control model which can control the burst input messages of the BSC(Base Station controller) in mobile communication systems more efficiently and reliably, by dividing the input messages characteristically and using multiprocessor system. Using M/M/c/K queueing model, we briefly analyze proposed model to get characteristic parameters which are required to performance improvement. On the base of the results, we compare our proposed model with the conventional one by using SLAM II with regard to the following factors : the call blocking rate of the input message, the distribution of average queue length, the utilization of process controller(server), and the distribution of average waiting time in queue. In addition, we modified our model which has overload control function for burst input messages, and analyzed its performance.

• 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.

• 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.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.5B
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• pp.942-955
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• 2000

This paper proposes enhanced designs of global service logic and information flow for the mass-call based IN service, which increase call completion rates and optimize the resource in joint wired and wireless networks. In order to hanve this logic implemented, we design a Dynamic Queue Manager(DQM) applied to the call queuing service feature in the Service Control Point(SCP). In order to apply this logic to wireless service subscribers as well as wired service subscribers, the service registration flags between the Home Location Register(HLR) and the SCP are managed to notify the DQM of the corresponding service subscribers’ mobility. Hence, we present a dynamic queue management mechanism, which dynamically manages the service group and the queue size based on M/M/c/K queueing model as the wireless subscribers roam the service groups due to their mobility characteristics. In order to determine the queue size allocated by the DQM, we simulator and analyze the relationship between the number of the subscriber’s terminals and the drop rate by considering the service increment rate. The appropriate waiting time in the queue as required is simulated according to the above relationship. Moreover, we design and implement the DQM that includes internal service logic interacting with SIBs(Service Independent building Blocks) and its data structure.