• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.6 no.2
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• pp.26-33
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• 2007

Social cost has been increased by traffic accident and congestion since early 1990s. The construction of roadways and railways has been suggested as countermeasures. However, ITS has finally introduced as a logical solution because the expenses of infrastructures are costly. The data collection field has developed through numerous researches and pilot projects. However the information provision field does need a lot of study. The traffic information broadcasting whether simple traffic information or the value-added information has been available via radio, television and internet which does not require tremendous investment compared with data collection stage. Therefore, this study reviews the suitability of the gadget service usually offered by window vista users which is the result of the development of technology and the changes of internet environment. It also suggests to using the RSS(Really Simple Syndication) manner as a basic method to provide the traffic information based on the needs of user in order to enhance the usability of traffic information. For this, this study analyzes the current methods and techniques of traffic information service which is widely available by local governments and companies and suggest possible changes and methods in order to provide Gadget-based service to the public.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.06a
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• pp.360-363
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• 2007

In sensor network, the importance of transporting data with reliability is growing gradually to support communications. Data flow from sink to nodes needs reliability for the control or management, that is very sensitive and intolerable, however relatively, data flow from nodes to sink is tolerable. In this paper, with emphasis of the data flow from sink to nodes, we proposed the mechanism that establishes confidence interval for transport. Establishing confidence interval hop-by-hop, not end to end, if errors happen or there's missing data, this mechanism recovers them with selective acknowledgement using fixed window. In addition, this mechanism supports traffic congestion control depending on the buffer condition. Through the simulation, we showed that this mechanism has an excellent performance for error recovery in sensor network.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.4
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• pp.35-44
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• 2004

Recently announced TCP Vegas predicts the degree of congestion in the network and then control the congestion window size. Thus it shows better performance than TCP Reno. however, TCP vegas does not assume any network model, its congestion window control is very limited. Because or this limitation, TCP vegas still can not adapt to fast changing available bandwidth. In this paper, we introduce a new TCP algorithm which adapts to fast changing available bandwidth well. To devise such a TCP, we model the end to end network of TCP connection as a queueing system and finds congestion window size which can utilize the available bandwidth sufficiently but not make the network congested. The simulation results show that our algorithm adapts to the avaliable bandwidth faster than TCP vegas and as a results, when the available bandwidth is changing rapidly, our algorithm not only operates more stably than TCP Vegas, but also it shows higher thruput than TCP Vegas.

• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.3
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• pp.303-308
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• 2016

Recently, as the rapid development of network technology and the increase of services required high bandwidth such as multimedia service, the network traffic dramatically increases. This massive increase of network traffic causes some problems such as the degradation of QoS and the lack of network resources and, to solve these problems, various research to guarantee QoS have been performing. Currently, The most representative method to guarantee the QoS is the DiffServ(: Differentiated Service). The DiffServ defines the AF(: Assured Forwarding) PHB(: Per Hop Behavior) and statistically ensures the throughput over the certain level of data rate. However, the TCP congestion control method that make up the majority of the Internet traffic is not fundamentally suitable to the DiffServ that guarantees the throughput without managing the individual flow. Therefore, in this paper, we present this mismatch through the simulation as an example and propose the solution by controlling the TCP of the terminal in the network. The proposed scheme utilizes the information of the reception window size included in the ACK frame and does not require any modification of the TCP algorithms currently in use.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.6B
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• pp.387-395
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• 2005

In this paper, we propose an end-to-end mobility management and TCP flow control scheme which considers different link characteristics for vertical handover environments. The end-to-end mobility management is performed by using SIP protocol. When a mobile node moves to a new network, it informs its movement of the correspondent node by sending SIP INFO message containing a new IP address which will be used in the new network. And then the corresponding node encapsulates all packets with the new IP address and sends them to the mobile node. in general, RTT of WLAN is shorter than RTT of cdma2000. when the MN moves from WLAN network to cdma2000 network, TCP retransmission timeout will be occurred in spite of non congestion situations. Thus, TCP congestion window size will be decreased and TCP throughput will be also decreased. To prevent this phenomenon, we propose a method using probe packets after handover to estimate a link delay of the new network. We also propose a method using bandwidth ratio of each network to update RTT. It is shown through NS-2 simulations that the proposed schemes can have better performance than the previous works.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.4B
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• pp.288-297
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• 2003

The fast retransmit and fast recovery algorithm of TCP Reno, when multiple packets in the same window are lost, cannot recover them without RTO (Retransmission Timeout). TCP New-Reno can recover multiple lost packets by extending fast recovery using partial acknowledgement. If the retransmitted packet is lost again during fast recovery, however, RTO cannot be avoided. In this paper, we propose an algorithm called "Duplicate Acknowledgement Counting(DAC)" to alleviate this problem. DAC can detect the retransmitted packet loss by counting duplicate ACKs. Conditions that a lost packet can be recovered by loss recovery of TCP Reno, TCP New-Reno and TCP New-Reno using DAC are derived by modeling loss recovery behavior of each TCP. We calculate the loss recovery probability for random packet loss probability numerically, and show that DAC can improve loss recovery behavior of TCP New-Reno.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.6
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• pp.1190-1194
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• 2007

In sensor network, the importance of transporting data with reliability is growing gradually to support communications. Data flow from sink to nodes needs reliability for the control or management, that is very sensitive and intolerable, however relatively, data flow from nodes to sink is tolerable. In this paper, with emphasis of the data flow from sink to nodes, we proposed the mechanism that establishes confidence interval for transport. Establishing confidence interval herby-hop, not end to end, if errors happen or there's missing data, this mechanism recovers them with selective acknowledgement using fixed window. In addition, this mechanism supports franc congestion control depending on the buffer condition. Through the simulation, we showed that this mechanism has an excellent performance for error recovery in sensor network.

• 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.