• Journal of the Korea Society of Computer and Information
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• v.21 no.4
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• pp.17-24
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• 2016

In this paper, we propose a AH LSU(Advanced Hybrid QoS Routing Link State Update) Algorithm that improves the performance of Hybrid LSU(Hybrid QoS Link State State Update) Algorithm with statistical information of flow holding time in network. AH LSU algorithm has had both advantages of LSU message control in periodic QoS routing LSU algorithm and QoS routing performance in adaptive LSU algorithm. It has the mechanism that calculate LSU message transmission priority using the flow of statistical request bandwidth and available bandwidth and include MLMR(Meaningless LSU Message Removal) mechanism. MLMR mechanism can remove the meaningless LSU message generating repeatedly in short time. We have evaluated the performance of the MLMR mechanism, the proposed algorithm and the existing algorithms on MCI simulation network. We use the performance metric as the QoS routing blocking rate and the mean update rate per link, it thus appears that we have verified the performance of this algorithm.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2012.11a
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• pp.52-53
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• 2012

Belief propagation (BP) is a commonly used global energy minimization algorithm for solving stereo matching problem in 3D reconstruction. However, it requires large memory bandwidth and data size. In this paper, we propose a novel memory-efficient algorithm of BP in stereo matching on the Graphics Processing Units (GPU). The data size and transfer bandwidth are significantly reduced by storing only a part of the whole message. In order to maintain the accuracy of the matching result, the local messages are reconstructed using shared memory available in GPU. Experimental result shows that there is almost an order of reduction in the global memory consumption, and 21 to 46% saving in memory bandwidth when compared to the conventional algorithm. The implementation result on a recent GPU shows that we can obtain 22.8 times speedup in execution time compared to the execution on CPU.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.117-120
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• 2000

As the use of real-time multimedia applications increases, a considerable amount of “greedy” UDP traffic would easily dominate network bandwidth and packet loss. As a result, bandwidth available to TCP connections is oppressed and their performance extremely deteriorates. In or-der that both TCP and UDP sessions fairly co-exist in the Internet, it is vital that we consider the fairness among both protocols. In this work, we implement a “TCP-friendly” rate control mechanism suitable to video applications and con-sider its applicability to a real system through observation of the video quality at the receiver and the connection state. It is shown that we can achieve high-quality and stable video transfer fairly sharing the network bandwidth with TCP by applying our rate control at a control interval of 32 times as long as RTT.

• Speech Sciences
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• v.11 no.4
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• pp.67-73
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• 2004

Packet loss, jitter, and delay in the Internet are caused mainly by the shortage of network bandwidth. It is due to queuing and routing process in the intermediate nodes of the packet network. In the Internet whose bandwidth is changing very rapidly in time depending on the number of users and data traffic, controlling the peak transmission bit-rate of a VoIP. system depending on the channel condition could be very helpful for making use of the available network bandwidth. Adapting packet size to the channel condition can reduce packet loss to improve the speech quality. It has been shown in [1] that a VoIP system with an AMR speech codec provides better speech quality than VoIP systems with fixed rate speech codecs. With the adaptive codec mode assignment. algorithm proposed in [1], in this paper, we performed the voice transmission experiments using the wireless LAN through the real Internet environment. Experimental results are analyzed and discussed with our findings.

• Journal of Advanced Navigation Technology
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• v.18 no.5
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• pp.415-420
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• 2014

Another crucial issue is a providing secondary user(SU) with the its guaranteed quality of service(QoS) in cognitive radio systems, from the SU view to be allowed to opportunistically utilize the primary user(PU) spectrum on non-interfering. In this paper, we propose a bandwidth reallocation scheme for reducing SU dropping rate through renegotiation of requested channel numbers when available bandwidth is not enough for accepting the spectrum handoff SUs. We categorize SU calls into two types : the first priority and the second priority SU, and the first SU' service is supported by bandwidth reservation based on ARMA prediction model for PU arrivals, while the second SU's bandwidth demands for spectrum handoff is to be reallocated through their renegotiation. Simulation results show that our scheme can improve SU dropping rate and system resource utilization efficiency by bandwidth reallocation.

• Journal of KIISE
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• v.44 no.6
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• pp.628-636
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• 2017

HTTP adaptive streaming is a technique that improves the quality of experience by storing various quality videos on the server and requesting files of the appropriate quality based on network bandwidth. However, it is difficult to measure the actual bandwidth in wireless networks with frequent bandwidth changes and high loss rate. Frequent quality changes and playback interruptions due to bandwidth measurement errors degrade the quality of experience. We propose a technique to estimate the available bandwidth by measuring the jitter, which is the derivation of delay, on a packet basis and assigning a weight according to jitter. The proposed scheme reduces the number of quality changes and mitigates the buffer underflow by reflecting less bandwidth change when high jitter occurs due to rapid bandwidth change. The experimental results show that the proposed scheme improves the quality of experience by mitigating buffer underflow and reducing the number of quality changes in wireless networks.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.6
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• pp.203-210
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• 2013

Majority of the wireless ad hoc routing protocols are proposed to find feasible routes without considering the network load, end-to-end link quality and bandwidth requirements of the application. Therefore, protocol may not provide sufficient quality of service (QoS) to a high speed traffic such as multimedia. In this paper, we propose a path-quality and bandwidth-estimation based routing protocol (PBBR) for the high quality multimedia stream that can meet the application's bandwidth requirements and find the best reliable route. The novelty of this protocol is to select a reliable path to respond the application's requirements based on available bandwidth at each intermediate node and end-to-end path loss ratio. Obtained results from the simulation demonstrates that our protocol can achieve sufficient performance in terms of throughput and end-to-end delay.

• Journal of Satellite, Information and Communications
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• v.1 no.1
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• pp.12-19
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• 2006

In this paper, we propose the bandwidth borrowing scheme which improves the performance of the cal admission control of the integrated GEO satellite networks. In general, target transmission rates of communications and streaming services are fixed, but data services do not have the target transmission rates. Therefore, we can control the transmission rates for data services flexibly according to the system loading situation. When the available bandwidth of the system is insufficient, the bandwidth borrowing scheme gives the bandwidth to request real time services by the transmission rates control of data services through packet scheduler. We make the queueing model for our system model and demonstrate the results through simulations. The simulation results show that there is a 8.7-35.2 dB gain at the total blocking probability according to the use of bandwidth borrowing scheme.

• Journal of KIISE:Information Networking
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• v.35 no.5
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• pp.384-394
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• 2008

A wireless router is a device which allows several wireless clients to share an internet line using NAT (Network Address Translation). In a school or a small office environment where many clients use multiple wireless routers, a client may select anyone of wireless routers so that most clients can be clustered to a small set of the wireless routers. In such a case, there exists load unbalancing problem between clients and wireless routers. One of its result is that clients using the busiest router get poor service. The other is that the resource utilization of the whole wireless routers becomes very low. In order to resolve the problems, we propose a load sharing scheme to maximize network bandwidth utilization based on multiple SSID. In a time internal, the proposed scheme keeps to show the available bandwidth information of all the possible wireless routers to clients through multiple SSID. A new client can select the most available band with router. This scheme allows to achieve a good load balancing between clients and routers in terms of bandwidth utilization. We implemented the proposed scheme with ASUS WL 500G wireless router and performed experiments. Experimental results show the bandwidth utilization improvement compared to the existing method.

• Proceedings of the Korean Information Science Society Conference
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• 2005.11a
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• pp.652-654
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• 2005

인터넷에서의 효율적인 스트리밍 서비스(Streaming Service)를 위해서는 안정된 전송률의 보장이 중요하며, 네트워크의 대역폭이 제한적이기 때문에 네트워크를 공유하는 경쟁 트래픽의 형평성에 대한 고려도 필요하다. 이러한 필요성에 따라 멀티미디어 데이터 전송의 주요 프로토콜인 UDP(User Datagram Protocol)에 혼잡제어 메커니즘 적용에 대한 연구가 활발히 진행되고 있으며, 대표적으로 가용대역 폭 측정을 통한 혼잡제어 기법이 있다. 이 기법은 혼잡제어를 하기 위해서 네트워크의 상태에 따라 가변하는 가용대역폭(Available Bandwidth)을 측정하고 이것을 기반으로 전송률을 조절하는 방식을 말한다. 본 논문에서는 네트워크 상태에 따라 혼잡제어를 하기 위해서 가용대역폭을 보다 빠르고 정확하게 측정하고, 이것을 기반으로 스트리밍 서비스에 맞게 전송률을 조절하는 방법을 제안하였다. 실험을 통해 본 논문에서 제안한 방법이 기존의 가용대역폭을 측정하는 스트리밍 프로토콜 보다 성능이 개선이 되었음을 확인할 수 있었다.