• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.12B
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• pp.805-810
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• 2005

According to the user requirements for continuous and real-time multimedia information, the concept of Quality of Service (QoS) has emerged as a main issue in mobile ad-hoc networks. QoS routing is to find the route that meets the QoS requirements of the users. In this paper, we propose a routing protocol that is based on AODV over TDMA, one of the typical routing protocols for mobile ad-hoc networks, by making a bandwidth reservation for QoS guarantee. While the existing schemes calculate the maximum available bandwidth for each candidate path, our scheme is to check only if the bandwidth of a given path satisfies the end-to-end QoS requirement and then to make a reservation by allocating time slots efficiently in the wireless environment. In order to evaluate the performance of the proposed QoS routing protocol, some simulations are carried out in the ad-hoc environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.10B
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• pp.831-840
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• 2004

Providing quality of service (QoS) guarantees over wireless link requires thorough understanding and quantification of the interactions among the traffic source, the wireless channel, and the underlying error control mechanisms. In this paper, we account for such interactions in a network-layer model that we use to investigate the delay performance of a wireless channel. We consider a single ON/OFF traffic stream transported over a wireless link. The capacity of this link fluctuates according to a fluid version of Gilbert-Elliot's model. We derive the packet delay distribution via two different approaches: uniformization and Laplace transform. Numerical aspects of both approaches are compared. The delay distribution is further used to quantify the wireless effective bandwidth under a given delay guarantee. Numerical results and simulations are used to verify the adequacy of our analysis and to study the impact of error control and bandwidth allocation on the packet delay performance. Wireless networks, QoS, delay distribution, fluid analysis.

• Journal of KIISE:Computer Systems and Theory
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• v.36 no.1
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• pp.9-20
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• 2009

In a real-time system with both hard real-time periodic tasks and soft real-time aperiodic tasks, it is important to guarantee the deadlines of each periodic task as well as obtain fast response time for each aperiodic task. This paper proposes Enhanced Total Bandwidth Server (ETBS) with possibly shorter response time than Total Bandwidth Server (TBS), which is efficient and widely used for servicing aperiodic tasks. For uniprocessor system using Earliest Deadline First (EDF) scheduling algorithm, ETBS assigns an on-line deadline to each aperiodic task considering a surplus slack time which gained for every unit execution time of periodic job. The proposed method can fully utilize the processor while meeting all the deadlines of periodic tasks. We show that the proposed ETBS provides better response time of aperiodic tasks than TBS theoretically, but has the same computational complexity as TBS, O(1). Simulation results show that the response time of aperiodic tasks with ETBS are shorter than one with TBS.

• Journal of KIISE:Computer Systems and Theory
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• v.34 no.10
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• pp.529-538
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• 2007

Fair Queueing algorithms based on Generalized Processor Sharing (GPS) not only guarantee sessions with service rate and delay, but also provide sessions with instantaneous fair sharing. This fair sharing distributes server capacity to currently backlogged sessions in proportion to their weights without regard to the amount of service that the sessions received in the past. From a long-term perspective, the instantaneous fair sharing leads to a different quality of service in terms of delay and bandwidth to sessions with the same weight depending on their traffic pattern. To minimize such long-term unfairness, we propose a delay-bandwidth normalization model that defines the concept of value of service (VoS) from the aspect of both delay and bandwidth. A model and a packet-by-packet scheduling algorithm are proposed to realize the VoS concept. Performance comparisons between the proposed algorithm and algorithms based on fair queueing and service curve show that the proposed algorithm provides better long-term fairness among sessions and that is more adaptive to dynamic traffic characteristics without compromising its service rate and delay guarantees.

• Journal of KIISE:Software and Applications
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• v.35 no.4
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• pp.207-218
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• 2008

With the growth of the Internet, multimedia streaming becomes an important means to deliver video contents over the Internet and the amount of the streaming multimedia contents is also getting increased. However, it becomes difficult to guarantee the quality of service in real-time over the IP network environment with instantaneously varying bandwidth. In this paper, we propose an optimal adaptation framework for streaming contents over the Internet in the sense that the perceptual quality of the multi-angie content with multiple visual objects is maximized given the constraints such as available bandwidth and transcoding cost. In the multi-angle video service framework, the user can select his/her preferred alternate views among the given multiple video streams captured at different view angles for a same event. This enhanced experience often entails streaming problems in real-time over the network, such as instantaneous bandwidth changes in the Internet. In order to cope with this problem, we assume that multi-angle video contents are encoded at different bitrates and the appropriate video streams are then selected or transcoded for delivery to meet such bandwidth constraints. For the user selective consumption of the various bitstreams in the multi-angle video service, the bitstream in each angle can be encoded in various bitrate, and the user can select a sub-bitrstream in the given bitrstreams or transcode the corresponding content in order to deliver the optimally adapted video contents to the instantaneously changing network condition. Therefore, we define the transcoding cost which means the time taken for transcoding the video stream and formulate a unified optimization framework which maximizes the perceptual quality of the multiple video objects in the given constraints such as the transcoding cost and the network bandwidth. Finally, we present plenty of the experimental results to show the effectiveness of the proposed method.

• 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.

• The Transactions of the Korea Information Processing Society
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• v.4 no.7
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• pp.1770-1780
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• 1997

Our problem is improvement of image quality because it is inevitable cell loss of image data when traffic congestion occurs. If cells are discarded indiscriminately in transmission of MPEG video data, it occurs severe degradation in quality of service(QOS). In this paper, to solve this problem, we propose two method. The first, we analyze the traffic characteristics of an MPEG encoder and generate high priority and low priority data stream. During network congestion, only the least low priority cells are dropped, and this ensures that the high priority cells are successfully transmitted, which, in turn, guarantees satisfactory QoS. In this case, the prioritization scheme for the encoder assigns components of the data stream to each priority level based on the value of a parameter ${\beta}$. The second, Number of high priority cells are increased when value of ${\beta}$ is large. It occurs the loss of high priority cell in the congestion. To prevent it, this paper is regulated to data stream rate as buffer occupancy with UPC controller. Therefore, encoder's bandwidth can be calculated renegotiation of the encoder and networks. In this paper, the encoder's bandwidth requirements are characterized by a usage parameter control (UPC) set consisting of peak rate, burstness, and sustained rate. An adaptive encoder rate control algorithm at the Networks Interface Card(NIC) computes the necessary UPC parameter to maintain the user specified quality of service. Simulation results are given for a rate-controlled VBR video encoder operating through an ATM network interface which supports dynamic UPC. These results show that dynamic bandwidth renegotiation of prioritized data stream could provided bandwidth saving and significant quality gains which guarantee high priority data stream.

• 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 Communications and Networks
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• v.11 no.2
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• pp.175-186
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• 2009

A new medium access control (MAC) scheme embedding physical channels into multiband carrier sense multiple access/collision avoidance (CSMA/CA) networks is proposed to provide strict quality of service (QoS) guarantee to high priority users. In the proposed scheme, two priority classes of users, primary and secondary users, are supported. For primary users physical channels are provided to ensure strict QoS, whereas secondary users are provided with best-effort service using CSMA/CA modified for multiband operation. The performance of the proposed MAC scheme is investigated using a new multiband CSMA/CA Markov chain model capturing the primary user activity and the operation of secondary users in multiple bands. The throughput of secondary users is obtained as a function of the primary user activity and other CSMA/CA parameters. It is shown that the new MAC scheme yields larger throughput than the conventional single-band CSMA/CA when both schemes use the same bandwidth.

• Proceedings of the IEEK Conference
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• 2000.06a
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• pp.229-231
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• 2000

Nowadays, the telecommunication service provider has witnessed an unprecedented growth in data traffic and the need for networking. optical fiber can provide some THz of huge bandwidth WDM technology has been an emerging issue for the efficient use of optical links. WDM uses a number of different wavelength that are assigned to each channel. The minimal number of optical transceivers and receivers should be used in a node to build an economic WDM transmission system without degrading system performance. Hence, the analysis of performance parameters such as throughput and delay is important to guarantee the WDM system performance. in this paper, the performance of a MAC protocol on a slotted WDM system that has a tunable transmitter(Txt), a tunable receiver(Rxt), and a fixed receiver(Rxf), respectively, on each node, was statistically analyzed The computer simulation validates the performance analysis.