• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.3B
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• pp.288-303
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• 2009

In the near future it is foreseen that a genuine multimedia service over the WiMAX system is provided in a worldwide manner by exploiting the QoS technologies introduced in the wireless and wired broadband network. In this work we propose a pricing scheme for the multimedia service over the generic WiMAX system that supports a full QoS functionality. We assume real-time services such as the voice and video as well as the nonreal-time service such as the conventional high-speed data, and we propose a pricing and charging scheme for those services by investigating the inherent characteristics of those services and the multiple-class of QoS-service provided to them. After that we propose a method to compute expected revenue that is obtained from the WiMAX system by using an analytic method to estimate the usage of the bandwidth resources for the different class of services. Via numerical experiment, we verify the implication of the work.

• Proceedings of the IEEK Conference
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• 2002.06a
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• pp.97-100
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• 2002

In this paper, we analyze All IP network architecture and VoIP Service procedure using U Multimedia domain. And, we propose QoS Resource management scheme using with statistical measurement based admission control 8l rate based borrow bandwidth to improve real-time traffic QoS guarantee.

• The Transactions of the Korea Information Processing Society
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• v.7 no.2
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• pp.446-453
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• 2000

Mobile computing refers to an emerging new computing environment incorporating both wireless and wired high-speed networking technologies. In the near future, it is expected that mobile users will have access to a wide variety of services that will be made available over highspeed networks. The quality of these services in the high-speed networks can be specified in terms of several QoS paameters. The important QoS parameter in mobile computings is the guarantee for seamless communication which is to provide disruption free service to mobile users. A disruption in service could occur due to active handoffs. This paper proposes an extended staggened muticast approach which provides a probabilistic guarantee for disruption free service. The extended staggerd multicast approach estimates mobility direction and mobility velocity for a user. It is possible that data packets for a mobile host are multicasted to not all neighbor cells but a part of neighbor cells on the basis of these information. Therefore, the extended staggered multicast significantly reduces the static network bandwidth usage also provides a probabilistic guarantee for disruption free service.

• Journal of Digital Contents Society
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• v.18 no.2
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• pp.411-422
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• 2017

Current policies for sharing bandwidth increase average throughput and improve utilization of the bandwidth in the local network. However, with these policies, a central administer, which is responsible for allocating bandwidth to each network flow, cannot allocate resources based on user characteristics. Thus, it leads to unfair bandwidth allocation because it does not guarantee services based on user characteristics. Therefore, we propose a novel negotiation method to share the bandwidth in a limited bandwidth network, in which, a user negotiates with other users to gain more resource. Ideally, we use a fuzzy system to infer and determine whether a device will request bandwidth or not based on the current usage of the given device. We conduct two experiments consisting of a video streaming simulation in OPNET and a real-time video streaming in WebRTC. The results of the experiment indicate that the proposed method can flexibly share the bandwidth utilization based on user's requirement in the network.

• Journal of the Institute of Convergence Signal Processing
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• v.7 no.1
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• pp.24-32
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• 2006

In this paper, we proposed the multimedia call admission control algorithm with the bandwidth reservation based on the prediction of wireless terminal's location to guarantee quality of service for multimedia applications in cellular networks. This algorithm aims at minimizing possible errors In predicting the moving direction of terminals using a mobility prediction scheme. This prediction reduces the size of bandwidth reserved redundantly. In order to evaluate the performance of the algorithm, the blocking rate of new calls and the forced termination rate of hand-off calls are measured and compared the results with those of existing schemes. The results of the experiment revealed that the algorithm presented in this paper achieved better performance with lower call blocking rates and forced-termination rates than those of other methods.

• Proceedings of the IEEK Conference
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• 1998.06a
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• pp.86-89
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• 1998

Multiple types of information in a multimedia service are delivered though multiple virtual connections on ATM network, while each virtual connection may be controlled independently. A multimedia service requires an associative relationship among multiple information streams to provide required harmonization. There may be required additional traffic descriptors to guarantee the required harmonization among multiple information streams in a multimedia service. For buffering of large bandwidth information stream(e.g., video), extremely large buffer size is necessary, but this approach should not be efficient way to compensate a severely delayed cells/blocks experienced at network. The best way to solve this problem will be minimization of relative-delayed-transfer of cells/blocks to application processes through ATM network control. To minimize a delayed transfer the mapping between relative delay parameter(i.e., associative Group QoS parameters) and per-VC traffic descriptor will be necessary. This paper is present additional functions and parameters to interpret the mapping between relative delay parameters(i.e., associative Group QoS parameters) and per-VC traffic descriptors in ATM API for multimedia application services.

• Current Optics and Photonics
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• v.2 no.1
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• pp.39-46
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• 2018

Visible-light communication (VLC) combined with advanced illumination can be expected to become an integral part of next-generation communication networks. One of the major concerns in VLC implementation is developing resource-allocation schemes in a multi-user scenario. However, the scheduling for heterogeneous quality of service (QoS) traffic has not been studied so far, for the indoor VLC downlink system. In this paper, we creatively introduce effective-bandwidth and effective-capacity theory into the multi-user scheduling (MUS) problem, to guarantee the user's statistical delay QoS. We also take account of the aggregate interference (AI) in the indoor VLC downlink system, and analyze its impact on the user-centric MUS problem for the first time. Simulations show that the AI has a nonnegligible influence on the scheduling result, and that the proposed scheduling scheme could guarantee the user's QoS requirement under the premise of ensuring sum capacity.

• JSTS:Journal of Semiconductor Technology and Science
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• v.11 no.1
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• pp.33-39
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• 2011

H.264 decoders usually have pipeline architecture by a macroblock or a 4 ${\times}$ 4 sub-block. The period of the pipeline is usually fixed to guarantee the operation in the worst case which results in many idle cycles and higher data bandwidth. Adaptive pipeline architecture for H.264 decoders has been proposed for efficient decoding and lower the requirement of the bandwidth for the memory bus. However, it requires a controller for the adaptive priority control to utilize the advantage. We propose a smart bus arbiter that replaces the controller. It is introduced to adjust the priority adaptively the QoS (Quality of Service) control of the decoding process. The smart arbiter can be integrated the arbiter of bus systems and it works when certain conditions are met so that it does not affect the original functions of the arbiter. An H.264 decoder using the proposed architecture is designed and implemented to verify the operation using an FPGA.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.8B
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• pp.683-690
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• 2003

In this paper we present a theoretical framework for the network design with delay QoS guarantee to a voice at the packet level. Especially, we propose a method for estimating the bandwidth at the ingress edge routers accommodating the voice connections and data sessions in the next-generation If network. First, we describe network architecture for VoIP (Voice over IP) services in the NGN (Next Generation Network). After that, we propose a procedure for dimensioning the bandwidth at the output port of a router that accommodates voice and data traffic using the non-preemptive queuing system with strict priority service scheme. Via numerical experiments we illustrate the implication of the proposition.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.12B
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• pp.2320-2334
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• 1999

In this paper, we propose multiple IS-MMBS service schemes for very low power and micro-size RMIMS (radio-interfaced micro information monitoring system) terminals. In MMBS service area, when new arrival RMIMS terminals have real-time traffic characteristic or large traffic bandwidth, only single IS-MMBS service scheme can not guarantee RMIMS terminal's QoS(quality of service) such as buffer overflow or packet loss. In this case, the proposed multiple IS-MMBS service schemes can be effectively used for QoS service of RMIMS terminal. According to clustering method of RMIMS terminals and MMBS segment method, the proposed schemes can be divided into terminal segment method, region segment method, application based segment method, traffic type based segment method, overlapping segment method and hybrid segment method