• Proceedings of the Korea Contents Association Conference
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• 2006.11a
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• pp.716-720
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• 2006

In this paper, we designed and proposed new routing algorithm that can support a variety of vital-sign traffic characteristic and could be applicable to USN for telemedicine by using adaptive transmission power level and increase frequency of routing request message. In proposed routing algorithm, when an emergency vital-sign traffic is applied, we use large transmission power to reduce route query response time and make the priority order in route process. On the other hand, for non emergency vital-sign traffic, we use low transmission power and adaptive decrease frequency of routing request message. which is insensitive to delay. The proposed scheme should be better QoS performance in complex USN than conventional method, which is performed based on uniform transmission power level.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.12B
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• pp.811-817
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• 2005

Efficient bandwidth management is necessary in order to provide high quality service to users in a multimedia wireless/mobile network. In this paper, I propose an on-line load balancing algorithm with preemption. This technique is able to balance the traffic load among cells accommodating heterogeneous multimedia services while ensuring efficient bandwidth utilization. The most important features of my algorithm are its adaptability, flexibility and responsiveness to current network conditions. In addition, my online scheme to control bandwidth adaptively is a cell-oriented approach. This approach has low complexity making it practical for real cellular networks. Simulation results indicate the superior performance of my algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.7A
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• pp.629-634
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• 2002

The next generation of wireless communication system is expected to provide users with wireless multimedia services such as high speed internet access and wireless mobile computing. Depending on the Quality of Service(QoS) requirements and different applications per user, many wireless communication systems have been proposed. The simple decoding algorithm of space-time block coding is based on the assumption of flat fading channel, where no intersymbol interference exists. In this paper, we propose to combine space-time transmit diversity with adaptive equalizer. The proposed system effectively eliminates interference caused by multipath environment. Simulation results show that the proposed system provides improved bit error rate performance.

• The KIPS Transactions:PartC
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• v.9C no.6
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• pp.953-960
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• 2002

Rel-time application programs have constraints which need to be met between media-data. It is client-leading synchronization that is absorbing variable transmission delay time and that is synchronizing by feedback control and palyout control. It is the important factor for playback rate and QoS if the buffer level is normal or not. This paper, The method of maintenance buffer normal state transmits in multimedia server by appling feedback of filtering function. And synchronization method is processing adaptive playout time for smooth presentation without cut-off while media frame is skip. When audio frame which is master media is in upper threshold buffer level we decrease play out time gradually, low threshold buffer level increase it slowly.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.48 no.6
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• pp.140-148
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• 2011

Multimedia streaming service is susceptible to loss and delay of data as it requires high bandwidth and real time processing. Therefore QoS cannot be guaranteed due to data loss caused by heavy network traffic. To deal with these problems, a study on algorithm that can provide high quality multimedia service by considering both network congestion and characteristics of multimedia data is required. In this paper, we propose APQM algorithm which probabilistically removes packet by the congestion level of the queue in wireless station. The comparison with other existing scheduling algorithms shows tht congestion in the network is reduced and multimedia service quality of the proposed algorithm is improved.

• Proceedings of the Korean Operations and Management Science Society Conference
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• 2005.05a
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• pp.230-237
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• 2005

WCDMA방식에서 채택한 AMR(Adaptive Multirate) 음성 코덱은 4.75kbps에서 12.2kbps까지 8개의 가변 출력률을 가진다. 기지국제어시스템(Radio Network Controller)은 무선망 상황에 따라 AMR 출력 모드를 가변적으로 조정해 줌으로써 높은 사용자 QoS와 효율적인 시스템 성능을 얻을 수 있다. 본 연구에서는 순방향 WCDMA 채널에서 세 가지의 AMR 출력 모드 할당 방식을 제안하고, 음성 사용자가 경험하는 QoS 만족도를 시스템 성능 척도로하여 제안된 방식들을비교 할 수 있는 분석적 방법을 제시한다. 실험 예제를 통해서 시스템 부하에 따른 시스템 성능 척도의 변화를 도시함으로써 최적의 AMR 모드 할당 방식을 결정하는 기준을 제시한다.

• Journal of Korea Multimedia Society
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• v.5 no.4
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• pp.441-449
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• 2002

For multimedia traffics to be supported successfully in wireless network environment, it is necessary to provide Qualify-of-Service(QoS) guarantees among mobile hosts(clients). In order to guarantee the QoS, we have to keep the call blocking probability below a target value during hand-off session. However, the QoS negotiated between the client and the network may not be guaranteed due to lack of available channels for traffic of a new cell, since on service mobile clients should be able to continue their sessions. In this paper, we propose an efficient load-balancing algorithm based on the adaptive bandwidth reservation scheme for enlarging available channels in a cell. Proposed algorithm predicts the direction of clients in a cell and adjusts the amount of the channel to be reserved according to the load status of the cell. This method is used to reserve a part of bandwidths of a cell for hand-off calls to its adjacent cells and this reserved bandwidth can be used for hand-off 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 an adaptive bandwidth reservation. In order to evaluate the performance of our algorithm, we measure metrics such as the blocking probability of new calls and dropping probability of hand-off calls, and compare with those of existing schemes.

• Journal of Advanced Navigation Technology
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• v.10 no.1
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• pp.70-78
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• 2006

This paper has addressed the scheme to design single rate Rate Adaptive Shaper (srRAS) proposed in RFC2963. srRAS is the shaper used in conjugation with downstream single rate Three Color Marker (srTCM) described in RFC269. it is tail-drop First Input First Out (FIFO) queue that is drained at a variable rate. srTCM meters IP packet streams from srRAS and marks its packets to be either green, yellow, or red. This shaper has been proposed to use at the ingress of differentiated services networks providing AF PHB. And then srRAS can reduce the burstiness of the upstream traffic of srTCM. This paper addresses algorithm, architecture of srRAS, and the scheme to implement srRAS using Field-Programmable Gate Arrays (FPGA) and the related technology.

• Journal of KIISE:Information Networking
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• v.34 no.4
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• pp.239-245
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• 2007

New multimedia services over the cellular/WLAN overlay networks require different Quality of Service (QoS). In the cellular/WLAN overlay structure, the main issue of load balancing is to balance the available bandwidth among different networks. In this paper, we propose a new online network management scheme. Our proposed scheme is designed to improve bandwidth utilization by balancing the traffic load between the cellular network and WLANs. Simulation results indicate the superior performance of our proposed framework to strike the appropriate performance balance between contradictory QoS requirements under widely varying diverse traffic loads.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.4B
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• pp.435-443
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• 2004

Although RSVP is a strong signaling protocol in providing QoS support, it is not scalable. Thus, the IETF introduced the concept of aggregation into RSVP However, the specification for the aggregation of RSVP does not specify how to compute the size of aggregate reservation for a path, but only describes how the resources are reserved and released with a given size. This paper proposes the method of properly determining the size of aggregate reservation on the path condition, when aggregate reservation is performed through RSVP for guaranteeing QoS of multimedia traffic. Simulation results show that the proposed method improves efficiency of the aggregation of RSVP.