• Journal of KIISE:Information Networking
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• v.28 no.4
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• pp.527-539
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• 2001

The wireless asynchronous transfer mode (ATM) system has the advantage of providing the broadband services with various quality-of-service requirements to the mobile terminal efficiently by utilizing the ATM technology developed for the wired ATM system. The available bit rate (ABR) service among various ATM services utilizes the available bandwidth remaining in the ATM link, which allows the efficient bandwidth usage. During the handoff of the mobile terminal, however, the queue length in the access point (AP) which resides in the boundary of the wired ATM network and the wireless ATM network may increase abruptly. In this paper, we propose a scheme which prevents the buffer-overflow and buffer-underflow in the AP during the handoff of the wireless ABR connection in the wireless ATM system using binary feedback rate-based ABR traffic control. This scheme controls the source's cell generation rate during both handoff period and some time interval after the completion of the handoff procedure. The simulation results show that the proposed scheme prevents the buffer-overflow and buffer-underflow. The proposed scheme can contribute to increasing the throughput of the wireless ABR service during handoff by preventing the buffer overflow and underflow during handoff period.

• The KIPS Transactions:PartC
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• v.12C no.1 s.97
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• pp.121-128
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• 2005

Tho existing TCP(Transmission Control Protocol) is known to be unsuitable for a network with the characteristics of high RDP(Bandwidth-Delay Product) because of the fixed small or large buffer size at the TCP sender and receiver. Thus, some trial cases of adjusting the buffer sizes automatically with respect to network condition have been proposed to improve the end-to-end TCP throughput. ATBT(Automatic TCP fluffer Tuning) attempts to assure the buffer size of TCP sender according to its current congestion window size but the ATBT assumes that the buffer size of TCP receiver is maximum value that operating system defines. In DRS(Dynamic Right Sizing), by estimating the TCP arrival data of two times the amount TCP data received previously, the TCP receiver simply reserves the buffer size for the next arrival, accordingly. However, we do not need to reserve exactly two times of buffer size because of the possibility of TCP segment loss. We propose an efficient TCP buffer tuning technique(called TBT-PLR: TCP buffer tuning algorithm based on packet loss ratio) since we adopt the ATBT mechanism and the TBT-PLR mechanism for the TCP sender and the TCP receiver, respectively. For the purpose of testing the actual TCP performance, we implemented our TBT-PLR by modifying the linux kernel version 2.4.18 and evaluated the TCP performance by comparing TBT-PLR with the TCP schemes of the fixed buffer size. As a result, more balanced usage among TCP connections was obtained.

• The KIPS Transactions:PartA
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• v.15A no.4
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• pp.227-238
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• 2008

Recently, as network environment has broadened, the demands on VOD have been increased. The VOD services can be categorized into two types, RVOD and NVOD. Practical VOD services adopt one of them exclusively. Since a method using only one of RVOD and NVOD is not able to deal with frequently variable demand of clients, it leads to a result of overload on a server and a waste of server bandwidth. The efficiency of the network resource usage becomes lower. Hence this paper presents a study on the protocol for efficient VOD services. We propose a new protocol appliable for the existing VOD service algorithm, analyze its performance through simulation, and developed server/client systems applying the new protocol. We propose a mode switching protocol combined with protocols used in RVOD and NVOD. The proposed protocol is not able only to control both RVOD and NVOD but also to change the mode between RVOD and NVOD. As a result of using the proposed protocol to meet frequently variable demand, server bandwidth can be used efficiently. Especially, it can be applied to the existing VOD service algorithms. Therefore, we expect that the proposed protocol in this paper will be widely used in emerging VOD markets.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.2A
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• pp.114-126
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• 2011

Video streaming over wireless multi-hop networks(WMNs) contains the following challenges from channel fading and variable bandwidth of wireless channel, and it cause degradation of video streaming performance. To overcome the challenges, currently, WMNs can use Forward Error Correction (FEC) mechanism. In WMNs, traditional FEC schemes, E2E-FEC and HbH-FEC, for video streaming are applied, but it has long transmission delay, high computational complexity and inefficient usage of resource. Also, to distinguish network status in streaming path, it has limitation. In this paper, we propose monitoring-based coordination of network-adaptive hop-to-end(H2E) FEC scheme. To enable proposed scheme, we apply a centralized coordinator. The coordinator has observing overall monitoring information and coordinating H2E-FEC mechanism. Main points of H2E-FEC is distinguishing operation range as well as selecting FEC starting node and redundancy from monitored results in coordination. To verify the proposed scheme, we perform extensive experiment over the OMF(Orbit Measurement Framework) and IEEE 802.1la-based multi-hop WMN testbed, and we carry out performance improvement, 17%, from performance comparison by existing FEC scheme.

• Journal of KIISE:Computing Practices and Letters
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• v.14 no.7
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• pp.657-668
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• 2008

Traditional off-the-shelf file server has several potential drawbacks to store data blocks. A first drawback is a lack of practical de-duplication consideration for storing data blocks, which leads to worse storage capacity waste. Second drawback is the requirement for high performance computer system for processing large data blocks. To address these problems, this paper proposes a clustering backup system that exploits file fingerprinting mechanism for block-level de-duplication. Our approach differs from the traditional file server systems in two ways. First, we avoid the data redundancy by multi-level file fingerprints technology which enables us to use storage capacity efficiently. Second, we applied a cluster technology to I/O subsystem, which effectively reduces data I/O time and network bandwidth usage. Experimental results show that the requirement for storage capacity and the I/O performance is noticeably improved.

• The Transactions of the Korea Information Processing Society
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• v.6 no.6
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• pp.1481-1492
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• 1999

As Internet is popularized by the advent of Web concept having characteristics such as open network, user-friendly, and easy-usage, there are many changes in Information systems providing various information. Web is rapidly transferred traditional Information systems to Web-based Information systems, because it provides not only text information but also multimedia information including image, audio, video, and etc. Also, as information contents were changed from text-based simple abstract information to full-text information, there was appeared various document formats processing Full-text information. But, as they naturally demand large systems memory, long processing time, broader transmission bandwidth, and etc, estimating of these factors is necessary when constructing information systems. This paper focuses on how to design and construct information system processing full-text information and providing function of an integrated document. Primarily, we should review standard document format which is used or developed, and any document format is appropriate to process full-text information in review with viewpoint of information system. Also, practically we should construct information system providing full-text information based on PDF document.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.12
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• pp.1870-1883
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• 1993

In this paper, in order to build up the User Network Interface based on ATM, a study on traffic control techniques which should be performed by main function groups-B 75,5 NT2, LEX-is discussed. The structure of B-NT2 which is the most important function group In the User Network Interface is defined in quite a simple manner in addition, the functional blocks of LEX are defined in a similar manner as those of B NT2. It is possible to distribute total traffic control functions by using the similarities between B-NT2 and LEX and by allocating virtual path identifiers fixedly according to the characteristics of the traffics. For the traffic control techniques of ATM, relations among Connection Admtsslon Control, Usage Parameter Control and Bandwidth Allocation Control are defined and Multi Rule Base structure to realize optimal control functions according to the characteristics of the source traffics is proposed. And the Real-time Variable Window algorithmsimply designed to be suitable for the Multi Rule Base architecture is also proposed. The performances of the proposed algorithm are analyzed through the computer simulation by generating on-off source traffic in a virtual system that has the form of the proposed hardware. The analyzed results show that the distributed control is possible and that the implementation of the proposed architecture and algorithm is possible.

• The KIPS Transactions:PartC
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• v.10C no.7
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• pp.891-898
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• 2003

This paper analyzes and proposes a scheme that improves the performance of the RTP that is developed to support the end-to-end transmission function and QoS monitor function for real-time multimedia data transmission. Although the existing RTP module supports real-time transmission, it has some problems in guaranteeing QoS parameters. To solve this problem, we propose a new Selective Repeat Adaptive Rate Control (SRARC). The SRARC can support QoS by referring to the data transmission status from the client and then classifying the network status into three levels. It selectively transmits multimedia data and dynamically controls transmission rates based on such information as bandwidth, packet loss rate, and latency that can be calculated in data transfer phase. To verify the SRARC, we implement it in real local area networks and compare the QoS parameters of the SRARC with those of the SR and RTP By the experimental results, the SRARC shows better performance in the aspects of bandwidth usage rate, packet loss rates, and transmission delays than the existing RTP schemes.

• Journal of KIISE:Information Networking
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• v.30 no.3
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• pp.407-415
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• 2003

As services requesting data transfer from a sender to a multiple number of receivers become popular, efficient group communication mechanisms like multicast get much attention. Since multicast is more efficient than unicast in terms of bandwidth usage and group management for group communication, many multicast protocols providing scalability and reliability have been proposed. Recently, router-supported reliable multicast protocols have been proposed because routers have the knowledge of the physical multicast tree structure and, in this scheme, repliers which retransmit lost packets are selected by routers. Repliers are selected dynamically based on the network situation, therefore, any receiver within a multicast group can become a replier Hence, all receivers within a group maintains a buffer for loss recovery within which received packets are stored. It is an overhead for all group receivers to store unnecessary packets. Therefore, in this paper, we propose a new scheme which reduces resource usage by discarding packets unnecessary for loss recovery from the receiver buffer. Our scheme performs the replier selection and the loss recovery of lost packets based on the LSM [1] model, and discards unnecessary packets determined by ACKs from erasers which represent local groups.