• Journal of the Korea Society of Computer and Information
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• v.15 no.12
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• pp.75-84
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• 2010

As broadcasting in Mobile Ad hoc NETwork (MANET) is the process that a node sends a packet to all other nodes in the network. it is used for routing protocols such as Ad hoc On demand Distance Vector (AODV) to disseminate control information for establishing the routes. In this paper, we propose Probabilistic Broadcasting mechanism based on Distance Ratio between sender and receive node in MANETs. The proposed approach is based on the combination of probability and distance based approach. A mobile node receiving broadcast packets determines the probability of rebroadcasting considering distance ratio from sender. The distance ratio of a node is calculated by the distance from sender and the length of radio field strength. As a node with high distance ratio is located far away from sender, rebroadcast probability is set to high value. On contrary, the low rebroadcast probability is set for a node with low distance ratio which is close to sender. So it reduces packets transmission caused by the early die-out of rebroadcast packets. Compared with the simple flooding and fixed probabilistic flooding by simulation, our approach shows better performances results. Proposed algorithm can reduce the rebroadcast packet delivery more than 30% without scanting reachability, where as it shows up to 96% reachability compared with flooding.

• Journal of KIISE:Databases
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• v.28 no.3
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• pp.484-493
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• 2001

As real-time database systems are extended to the distributed computing environment, the need to apply the existing real-time concurrency control schemes to the distributed computing environment has been made. In this paper we propose an efficient concurrency control scheme for distributed real-time database system. Our proposed scheme guarantees a transaction to commit at its maximum, reduces the restart of a transaction that is on the prepared commit phase, and minimizes the time of the lock holding. This is because it raises the priority of the transaction that is on the prepared commit phase in the distributed real-time computing environment. In addition, it reduces the waiting time of a transaction that owns borrowed data and improves the performance of the system, as a result of lending the data that the transaction with the raised priority holds. We compare the proposed scheme with DO2PL_PA(Distributed Optimistic Two-Phase Locking) and MIRROR(Managing Isolation in Replicated Real-time Object Repositories) protocol in terms of the arrival rate of transactions, the size of transactions, the write probability of transactions, and the replication degree of data in a firm-deadline real-time database system based on two-phase commit protocol. It is shown through the performance evaluation that our scheme outperforms the existing schemes.

• Journal of KIISE:Information Networking
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• v.30 no.1
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• pp.134-142
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• 2003

We present an ATM switch architectures based on the multistage interconnection network(MIN) for the efficient multicast traffic control. Many of these applications require multicast connections as well as point-to-point connections. Muiticast connection in which the same message is delivered from a source to arbitrary number of destinations is fundamental in the areas such as teleconferencing, VOD(video on demand), distributed data processing, etc. In designing the multicast ATM switches to support those services, we should consider the fairness(impartiality) and priority control, in addition to the overflow problem, cell processing with large number of copies, and the blocking problem. In particular, the fairness problem which is to distribute the incoming cells to input ports smoothly is occurred when a cell with the large copy number enters upper input port. In this case, the upper input port sends before the lower input port because of the calculating method of running sum, and therefore cell arrived into lower input port Is delayed to next cycle to be sent and transmission delay time becomes longer. In this paper, we propose the cell splitting and group splitting algorithm, and also the fairness scheme on the basis of the nonblocking characteristics for issuing appropriate copy number depending on the number of Input cell in demand. We evaluate the performance of the proposed schemes in terms of the throughput, cell loss rate and cell delay.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.8B
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• pp.661-673
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• 2003

To prevent performance degradation of TCP due to packet losses in the smooth handoff by the route optimization extension of Mobile IP protocol, a buffering of packets at a base station is needed. A buffering of packets at a base station recovers those packets dropped during handoff by forwarding buffered packets at the old base station to the mobile user. But, when the mobile user moves to a congested base station in a new foreign subnetwork, those buffered packets forwarded by the old base station are dropped and TCP transmission performance of a mobile user in the congested base station degrades due to increased congestion by those forwarded burst packets. In this paper, considering the general case that a mobile user moves to a congested base station, we propose a Priority Packet Forwarding to improve TCP performance in mobile networks. In the proposed scheme, without modification to Mobile IP protocol, the old base station marks a buffered packet as a priority packet during handoff. And priority queue at the new congested base station schedules the priority packet firstly. Simulation results show that proposed Priority Packet Forwarding can improve TCP transmission performance more than Implicit Priority Packet Forwarding and RED (Random Early Detection) schemes.

• Applied Biological Chemistry
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• v.38 no.3
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• pp.239-247
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• 1995

The improvement of distribution system and freshness prolongation of crisphead lettuce were carried out through vacuum cooling and distribution under the low temperature. Lettuce that vacuum-cooled and transported by cold storage car was shown better freshness than that distributed by conventional method when they arrived at cunsuming area. And it took $10{\sim}17$ hours until their temperatures arrived at same temperatures when they were stored at $0{\sim}15^{\circ}C$ cold storage room. It was cooled to $1^{\circ}C$ after 27 minutes with vacuum cooling apparatus. The weight loss of lettuce that vacuum cooled and transported by cold storage car was below 5% after 30 days cold storage. And ascorbic acid and chlorophyll retentions were 86% and 52%, respectively. The shelf-life of crisphead lettuce, distributed by vacuum cooling and cold storage car transportation, was 5 days at $15^{\circ}C$ and over 40 days at $0^{\circ}C$, respectively. However, when it was distributed by conventional method, it was only 3 days at $15^{\circ}C$ and 20 days at $0^{\circ}C$, respectively.

• Journal of KIISE:Computing Practices and Letters
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• v.10 no.6
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• pp.460-471
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• 2004

Since the RTP is suitable for real-time data transmission in multimedia services like VoD, AoD, and VoIP, it has been adopted as a real-time transport protocol by RTSP, H.323, and SIP. Even though the RTP protocol stack for embedded systems has been in great need for efficient support of multimedia services, such a stack has not been developed yet. In this paper, we explain embeddedRTP which supports the RTP protocol stack at the kernel level so that it is suitable for embedded systems. Since embeddedRTP is designed to reside in the UBP module, existing applications which rely ell TCP/IP services can proceed the same as before, while applications which rely on the RTP protocol stack can request HTP services through embeddedRTp API. EmbeddedRTP stores transmitted RTP packets into per session packet buffer, using the packet's port number and multimedia session information. Communications between applications and embeddedRTP is performed through system calls and signal mechanisms. Additionally, embeddedRTP API makes it possible to develop applications more conveniently. Our performance test shows that packet-processing speed of embeddedRTP is about 7.5 times faster than that oi VCL RTP for multimedia streaming services on PDA in spite that its object code size is reduced about by 58% with respect to UCL RTP's.

• Journal of Korean Society of Transportation
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• v.30 no.2
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• pp.29-42
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• 2012

Recently, traffic congestion has become serious due to increase of private car usages. Carsharing or other innovative public transportation systems were developed to alleviate traffic congestion and carbon emissions. These measures can make the traffic environment more comfortable, and efficient. Cloud Transportation System (CTS) is a recent carsharing model. User can rent an electronic vehicles with various traffic information through the CTS. In this study, a concept, vision and scenarios of CTS are introduced. And, authors analyzed the location of CTS rental stations and estimated CTS demands. Firstly, we analyze the number of the population, employees, students and traffic volume in study areas. Secondly, the frequency and utilization time are examined. Demand for CTS in each traffic zone was estimated. Lastly, the CTS rental station location is determined based on the analyzed data of the study areas. Evaluation standard of the determined location includes accessibility and density of population. And, the number of vehicles and that of parking zone at the rental station are estimated. The result suggests that Haewoondae Square parking lot would be assigned 11 vehicles and 14.23 parking spaces and that Dongbac parking lot be assigned 7.9 vehicles and 10.29 parking spaces. Further study requires additional real-time data for CTS to increase accuracy of the demand estimation. And network design would be developed for redistribution of vehicles.

• Journal of KIISE:Computer Systems and Theory
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• v.28 no.9
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• pp.442-460
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• 2001

In video-on-demand(VOD) systems it is important to minimize initial latency and memory requirements. The minimization of initial latency enables the system to provide services with short response time, and the minimization of memory requirements enables the system to service more concurrent user requests with the same amount of memory. In VOD systems, since initial latency and memory requirement increase according to the increment of buffer size allocated to user requests, the buffer size allocated to user requests must be minimized. The existing static buffer allocation scheme, however, determines the buffer size based on the assumption that thy system is in fully loaded state. Thus, when the system is in partially loaded state, the scheme allocates user requests unnecessarily large buffers. This paper proposes a dynamics buffer allocation scheme that allocates user requests the minimum buffer size in fully loaded state as well as a partially loaded state. This scheme dynamically determines the buffer size based on the number of user requests in service and the number of user requests arriving while servicing current requests. In addition, through analyses and simulations, this paper validates that the dynamics buffer allocation outperforms the statics buffer allocation in initial latency and the number of concurrent user requests that can be supported. Our simulation results show that, in proportion to the static buffer allocation scheme, the dynamic buffer allocation scheme reduces the average initial latency by 29%~65%, and in a systems having several disks. increases the average number of concurrent user requests by 48%~68%. Our results show that the dynamic buffer allocation scheme significantly improves the performance and reduce the capacity requirements of VOD systems.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.7
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• pp.63-73
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• 2015

In the end to end data transfer protocols, it is very important to correctly estimate available bandwidth. In UDT (UDP based Data Transfer), receiver estimates the MTR (Maximum Transfer Rate) of the current link using pair packets transmitted periodically from sender and, then sender finally decides the MTR through EWMA (Exponential Weighted Moving Average) algorithm. Here, MTR has to be exactly estimated because available bandwidth is calculated with difference of MTR and current transfer rate. However, when network is congested due to traffic load and where competing flows are coexisted, it bring about a severe fairness problem. This paper proposes a congestion degree based MTR estimation algorithm. Here, the congestion degree stands a relative index for current congestion status on bottleneck link, which is calculated with arriving intervals of a pair packets. The algorithm try to more classify depending on the congestion degree to estimate more actual available bandwidth. With the network simulation results, our proposed method showed that the fairness problem among the competing flows is significantly resolved in comparison with that of UDT.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.2
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• pp.293-300
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• 2007

As the development of the various IP-based services, it is expected that Internet telephony service will gradually replace the traditional PSTN-based telephony service. But there are many issues resolved to spread the Internet telephony service. One of them is to support the emergency services in the Internet telephony. In the case of USA, it has been regulated that 911 services should be supported in the Internet telephony services using VoIP on the similar performance level to PSTN 911 service. According to the regulation, basic VoIP 911 calls should be routed to the general access line of LEA without the location information or the callback number, but the enhanced VoIP 911 calls with the location information and callback number should be routed on the dedicated 911 network and destined to the local 911 distribution center such as PSAP. But, in the current VoIP-based Internet telephony network, the emergency call service has not been handled as one of the special services as well at has a worse performance in comparison to it on PSTN. Moreover, the service has a critical problem that it can not be destined to the nearest PSAP because of the insufficient information about the location information and the call back number. In this paper, we suggest the SIP-based emergency call service mechanism in order to resolve the problems above mentioned. This suggested mechanism is implemented to show its effectiveness and efficiency.