• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.8A
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• pp.613-619
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• 2003

Bluetooth is a communication technology enabling short-range devices to be wirelessly connected. A master and one or more slave devices are connected to form a piconet, and piconets are joined to form a scatternet. The units participating in two or more piconets in a scatternet, is called bridge or gateway nodes. In order to operate the scatternet efficiently, both piconet scheduling for the master and slaves of a piconet, and scatternet scheduling for the bridge nodes are playing important roles. In this paper, we propose a traffic-adaptive dynamic scatternet scheduling algorithm based on rendezvous points and rendezvous windows. The performance of the proposed algorithm is compared and analyzed with that of a static scheduling algorithm via simulations. Simulation results show that our algorithm can distribute wireless resources efficiently to bridge nodes depending on the traffic characteristics.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.6A
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• pp.558-564
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• 2007

3G Long Term Evolution, which aims for various mobile multimedia service provision by enhanced wireless interface, proposes VoIP-based voice service through a Packet Switching (PS) domain. As delay and loss-sensitive VoIP traffic flows through the PS domain, more challenging technical difficulties are expected than in Circuit Switching (CS) domain based VoIP services. Moreover, since 3G LTE, which adopts the OFDM as its physical layer, introduces Physical Resource Block (PRB) as a unit for transmission resources, new types of resource management schemes are needed. This paper proposes a PRB scheduling algorithm of MAC layer for VoIP service in 3G LTE and shows the simulation results. The proposed algorithm has two key parts; dynamic activation of VoIP priority mode to satisfy VoIP QoS requirements and adaptive adjustment of the priority mode duration in order to minimize the degradation of resource utilization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.11
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• pp.1701-1713
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• 1993

This paper proposes a simple and highly effective RACMC(Real-time ATM Cell Monitoring and Control) algorithm and the resulting bandwidth gain effects art considered. RACMC algorithm performs usage parameter controls according to the monitoring informations of input data cells generated from the accepted connections and the controlling informations set by the M/P(Management Plane) for that connection. The results of monitoring and controlling actions for ATM data cells are transmitted to the M/P and the control parameters in lookup table are updated according to the condition of currently used bandwith. Therefore, the proposed algorithm can allocate network resource optimally and solve the several tantalizing problems that the existing cell control algorithm have, that is, the difficulty in controlling as monitoring very bursty traffics, unavoidable processing delay, and limited input buffer size when implemented. By the performance analysis using computer simulation, RACMC algorithm proves to be very effective especially in ATM network as implemented simply.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.7B
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• pp.868-877
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• 2011

In this paper, the feasibility of ORBCOMM satellite-based automatic identification system (SAT-AIS) is evaluated in the context of ship AIS slot collision probability depending on reporting rate, We evaluate detection probability evaluation of ORBCOMM satellite-based AIS considering link budget, SOTDMA protocol and satellite's swath width. The simulation determines the total number of vessels served by ORBCOMM satellite according to satellite's swath width, AIS slot allocation and reporting rate. By simulation results, the ORBCOMM satellite-based AIS slot collision is increased directly proportional to the total number of vessels and the more detection probability evaluation of ORBCOMM satellite-based AIS degrader, the more sip AIS reporting rate shorter.

• Journal of Korea Multimedia Society
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• v.7 no.5
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• pp.698-712
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• 2004

DiffServ network architecture does not define a call admission control procedure explicitly. In this paper, a new DiffServ QoS control mechanism is suggested which, after the call admission control, can execute packet process according to the class while, at the same time, executing on the flow based call admission control in the DiffServ network. Routers on the path from the source to the destination estimate the aggregated class traffic of the existing flows and the new incoming flow and then, perform a call admission control in accordance with the type of classes efficiently based on the required bandwidth per each class that can meet the user's QoS requirements. In order to facilitate the packet process according to the class after the flow based call admission control, a mechanism is suggested that can adjust the network resources to classes dynamically. The performance analysis on this mechanism is carried out through a simulation.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.11 s.353
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• pp.68-77
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• 2006

In accommodating the asymmetric traffic for future wireless multimedia services, the dynamic time division duplexing (D-TDD) scheme is considered as one of the key solutions. With the D-TDD mode, however, the inter-BS and inter-MS interference is inevitable during the cross time slot (CTS) period, and this interference seriously degrades the system performance. To mitigate such interference, we propose a region and time partitioning D-TDD architecture for OFDM systems. Each time slot in the CTS period is split into several minislots, and then each cell is divided into as many regions as the number of minislots per time slot. We then assign the minislots only to the users in its predefined corresponding region. On top of such architecture which inherently separates the interfering entities farther from each other, we design a robust time slot allocation scheme so that the inter-cell interference can be minimized. By the computer simulation, it has been verified that the proposed scheme outperforms the conventional time slot allocation methods in both the outage probability and the bandwidth efficiency.

• Journal of Korea Multimedia Society
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• v.15 no.5
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• pp.587-594
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• 2012

IEEE 802.15.6 standard technology is proposed for low-power wireless communication in, on and around body, where vital signs such as pulse, blood pressure, ECG, and EEG signals are transmitted as a type of data packet. Especially, these vital signs should be delivered in real time, so that the latency from slave node to hub node can be one of the pivotal performance requirements. However, in the case of IEEE 802.15.6 technology data retransmission caused by transmission failure can be done in the next superframe. In order to overcome this limitation, we propose an adaptive polling algorithm for IEEE 802.15.6 technology. The proposing algorithm makes the hub to look for an appropriate time period in order to make data retransmission within the superframe. Through the performance evaluation, the proposing algorithm achieves a 61% and a 73% latency reduction compared to those of IEEE 802.15.6 technology in the environment of 70% traffic offered load with 10ms and 100ms superframe period. In addition, the proposing algorithm prevents bursty traffic transmission condition caused by mixing retransmission traffic with the traffic reserved for transmission. Through the proposing adaptive polling algorithm, it will be possible to transmit time-sensitive vital signs without severe traffic delay.

• 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.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2001.06a
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• pp.21-24
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• 2001

For very large multimedia traffic to be supported successfully in wireless network environment, it is necessary to provide Quality-of-Service(QoS) guarantees between mobile hosts(clients). In order to guarantee the Qos, we have to keep the call blocking probability below target value during handoff session. However, the QoS negotiated between the client and the network may not be guaranteed due to lack of available channels for traffic in the new cell, since mobile clients should be able to continue their on-going sessions. In this paper we propose a efficient load-balancing algorithm based on the adaptive bandwidth reservation scheme for enlarging available channels in a cell. We design a new method to predict the mobility of clients using MPT(mobility profile table). This method is then used to reserve a part of bandwidths for handoff calls to its adjacent cells and this reserved bandwidth can be used for handoff call prior to new connection requests. If the number of free channels is also under a low threshold value, our scheme use a load-balancing algorithm with a adaptive bandwidth reservation. In order to evaluate the performance of our algorithm, we measure the metrics such as the blocking probability of new calls and dropping probability of handoff calls, and compare with other existing schemes.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.9-16
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• 2005

A new online routing algerian is proposed in this paper, which use the interference-prediction to solve the network congestion originated from extension of Internet scope and increasing amount of traffic. The end-to-end QoS has to be guaranteed in order to satisfy service level agreements (SLAs) in the integrated networks of next generation. For this purpose, bandwidth is allocated dynamically and effectively, moreover the path selection algorithm is required while considering the network performance. The proposed algorithm predicts the level of how much the amount of current demand interferes the future potential traffic, and then minimizes it. The proposed algorithm considers the bandwidth on demand, link state, and the information about ingress-egress pairs to maximize the network performance and to prevent the waste of the limited resources. In addition, the interference-prediction supports the bandwidth guarantee in dynamic network to accept more requests. In the result, the proposed algorithm performs the effective admission control and QoS routing. In this paper, we analyze the required conditions of routing algorithms, the aspect of recent research, and the representative algorithms to propose the optimized path selection algorithm adequate to Internet franc engineering. Based on these results, we analyze the problems of existing algorithms and propose our algorithm. The simulation shows improved performance by comparing with other algorithms and analyzing them.