• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.6B
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• pp.495-507
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• 2006

In wireless local area networks (WLANs), the successful design of scheduling algorithm is a key factor in guaranteeing the various quality of service (QoS) requirements for the stringent real-time constraints of multimedia services. In this paper we propose a multipolling-based dynamic scheduling algorithm for providing delay guarantees to multimedia traffics such as MPEG streams. The dynamic algorithm exploits the characteristics of MPEG stream, and uses mini frames for feedback control in order to deliver dynamic parameters for channel requests from stations to the point coordinator (PC) operating at the access point (AP). In this scheme, the duration of channel time allocated to a station during a superframe is changed dynamically depending on the MPEG frame type, traffic load and delay bound of the frame, etc. Performance of the proposed scheme is investigated by simulation. Our results show that compared to conventional scheme, the proposed scheme is very effective and has high performance while guaranteeing the delay bound.

• Journal of Korea Multimedia Society
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• v.12 no.3
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• pp.419-426
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• 2009

The WiMedia Alliance has specified a Distributed Medium Access Control (D-MAC) protocol based on UWB for high speed wireless home networks and WPANs. The WiMedia D-MAC fundamentally removes the problems of the centralized MAC revealed at IEEE 802.15.3 MAC, by adopting a distributed architecture. However, there is no completely distributed method for allocating fair data rates to all traffic streams by considering each stream's QoS parameter. In this paper, a novel fair distributed QoS admission control method is proposed. The proposed method is a fair, adaptive QoS provisioning method, by allocating time slots to devices according to the current traffic load condition, through executing only a proposed single satisfaction of QoS (SoQ) algorithm at each device.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.1B
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• pp.39-46
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• 2005

As the multimedia application traffic takes more portion in the internet traffic, it is necessary to control the network congestion through the congestion control protocol. In addition, the QoS-enabled networks such as DiffServ become an indispensable technology when running the multimedia applications. However, the previously proposed end-to-end congestion control algorithms take the round trip time to react the network congestion. Thus, as the RTT becomes larger, the reaction against the congestion gets delayed further, while the network congestion gets worse. In addition the performance of end-to-end congestion control algorithm is degraded if the QoS-enabled network runs the congestion control mechanism in the network level without any coordination between them. In this paper, we propose the early rate adaptation protocol for the DiffServ network which effectively linke the congestion control algorithm at the host and the congestion mechanism in the network together. By taking advantage of early congestion notification from the network it is possible to react the network congestion more quickly and effectively.

• Journal of Korea Multimedia Society
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• v.11 no.2
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• pp.197-205
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• 2008

TCP packets occupy over 90% of current Internet traffic thus understanding of TCP throughput is crucial to understand Internet. Under the TCP congestion regime, heterogeneous flows, i.e., flows with different round-trip times (RTTs), that share the same bottleneck link will not attain equal portions of the available bandwidth. In fact, according to the TCP friendly formula, the throughput ratio of two flows is inversely proportional to the ratio of their RTTs. It has also been shown that TCP's unfairness to flows with longer RTTs is accentuated under loss synchronization. In this paper, we show that, injecting bursty background traffic may actually lead to new type of synchronization and result in unfairness to foreground TCP flows with longer RTTs. We propose three different metrics to characterize traffic burstiness and show that these metrics are reliable predictors of TCP unfairness.

• Journal of Korea Multimedia Society
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• v.11 no.3
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• pp.331-340
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• 2008

In order to make load balancing, workload information of nodes should be informed to the network. In a load balancing algorithm[13] based on the SBIBD(Symmetric Balanced Incomplete Block Design), each node receives global workload information by only two round message exchange with O (${\upsilon}{\sqrt{\upsilon}}$) traffic overhead, where ${\upsilon}$ is the number of nodes. It is very efficient but works well only when ${\upsilon}=p^2+p+1$ for a prime number p. In this paper, we generate a special incidence structure in order for the algorithm works well for an arbitrary number of nodes. In the experiment with w,($5{\leq}w{\leq}5,000$), nodes and more than 80% of receiving workload information, traffic overhead was less than O ($w{\sqrt{w}})$ and the result for standard deviation of traffic overhead showed that each node has largely balanced amount of traffic overhead.

• Journal of Korea Multimedia Society
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• v.10 no.4
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• pp.450-461
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• 2007

In this paper, we propose the method to use power-saving mode (PSM) applicable to non real-time traffic and PSM applicable to real-time traffic simultaneously, for VoIP traffic with silence suppression. The proposed method uses PSC II during talk-spurt interval of parties A and/or B and uses PSC I or probabilistic sleep interval decision (PSID) method during mutual silence interval, respectively. To evaluate the performance of hybrid PSM (HPSM) based on PSC II or PSID method, we present average buffering delay, energy consumption of mobile station and VoIP packet drop probability with simulation runs. Results shows that proposed HPSM decreases energy consumption of mobile station up to 25 % while satisfying the packet drop probability within QoS requirement in case of end-to-end VoIP connection.

• Journal of Korea Multimedia Society
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• v.8 no.2
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• pp.192-200
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• 2005

In this paper, an improved User Parameter Control (UPC) algorithm is proposed for controlling traffic in ATM networks. The present algorithm consists of Cell Buffer, Red Token Pool, Green Token Pool, and Spacer. A token is generated at a certain time period, and the tokens are consumed one each time a cell arrives. At this time, if the delay element, named Spacer, is empty, the cell comes into the network. On the other hand, if Spacer is not empty, the cell cannot come into the network. In addition, in the case that tokens do not exist in Token Pool, the corresponding cell is thrown away The arbitrary function of the token used in this paper is removed. Instead, the token is used only to control network traffic. This research proposes new UPC Algorithm to improve the current cell delay rate and cell loss rate by dynamically adopting the delay element, named Spacer, with respect to the traffic condition, which is different from controlling that the cell comes into the network after a certain delay time period.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.10A
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• pp.1532-1538
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• 1999

In general, because error rate of wireless link is higher than that of wired link, DLC layer protocol for wired network with low error rate is not proper for wireless environments. In addition, the conventional DLC layer protocol for wireless network is optimized for the low-speed data service, so it is difficult to use conventional DLC protocol in the current mobile communication environments handing high-speed and multimedia services. Therefore, a DLC layer protocol that is suitable to current wireless communication environments is required. In this paper, we propose a novel error control scheme that supports a variety of traffic attribute and is applicable to high-speed and multimedia data service in WATM. The proposed scheme provides enhanced throughput performance for real-time traffic by using modified ASR ARQ without ACK and reduces loss rate by using FEC in the case of high error condition. Also, for non real-time traffic, the use of ASR ARQ without ACK enhances throughput performance and delay time is decreased by using FEC in the case of high error rate channel. As a result of simulation, the proposed scheme has better performance than conventional ASR ARQ protocol in view of delay and throughput.

• Journal of Korea Multimedia Society
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• v.11 no.7
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• pp.966-976
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• 2008

Providing Quality-of-Service(QoS) guarantee requires the reservation of resource needed to accommodate the traffic flow of service for which resource reservation protocol(RSVP) was introduced. However, if any RSVP-capable router on the path fails to accommodate the traffic flow due to lack of resource, the flow is not allowed no matter how much surplus resources other routers on the path have. In order to solve this problem, this paper proposes a resource reservation method that can adjust adaptively the delay budget of traffic flow allocated to each RSVP-capable router in which the router uses the recently developed general-time fair queuing scheduler. The results of the simulation applying the proposed method to an evaluation network show that it may yield the gain of up to 90% compared to that in the original one in terms of the number of admitted flows.

• Journal of the Korea Society of Computer and Information
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• v.16 no.1
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• pp.183-189
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• 2011

This paper suggests a user based OST(Open Street Traffic) system that solves TPEG's one-way communication problem, UTIS's limited usage and DSRC's small traffic bandwidth. In current commercial TPEG service, only some service providers collect traffic information. Thus, it can't cover traffic status in local lanes And UTIS, which local governments and police supports, requires additional equipments. Currently, only taxi and official vehicles use this system. Therefore, new traffic service by mobile device and user's participation can provide very detail traffic information coupling with previous traffic systems. But in this new system, real-time high volume data can be a problem. So, in this paper, new data storage manager design(TDSM :Traffic Data Storage Manager) is suggested and its performance is measured against commercial DBMS.