• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.782-787
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• 2004

Internet Engineering Task Force (IETF) has been considering the deployment of the Random Early Detection (RED) in order to avoid the increasing of packet loss rates which caused by an exponential increase in network traffic and buffer overflow. Although RED mechanism can prevent buffer overflow and hence reduce an average values of packet loss rates, but this technique is ineffective in preventing the consecutive drop in the high traffic condition. Moreover, it increases a probability and average number of consecutive dropped packet in the low traffic condition (named as "uncritical condition"). RED mechanism effects to TCP congestion control that build up the consecutive of the unnecessary transmission rate reducing; lead to low utilization on the link and consequently degrade the network performance. To overcome these problems, we have proposed a new mechanism, named as Extended Drop slope RED (ExRED) mechanism, by modifying the traditional RED. The numerical and simulation results show that our proposed mechanism reduces a drop probability in the uncritical condition.

• The KIPS Transactions:PartC
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• v.10C no.4
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• pp.479-484
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• 2003

In the current Internet, it is difficult for an Internet Phone to guarantee the QoS due to variable network conditions such as packet loss rate, delay and bandwidth. In addition, the QoS of an Internet Video Phone is more hard to guarantee because of video data. In this paper, we investigate application-level QoS control schemes that can adapt to variable network conditions, and describe an error control scheme and a congestion control scheme. Based on these QoS control schemes, we have designed and implemented an Internet Video Phone System that supports adaptive audio and video delivery. Through experiments, we found that the Internet Video Phone can reduce the packet loss rate considerably as well as adjust the transmission rate considering other TCP flows.

• Proceedings of the KIEE Conference
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• 2003.11b
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• pp.207-210
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• 2003

In this paper, we have implemented the DVR system which is controlled far away, and added a function of TCP/IP Network for image data and control signal transmission, the DVR system has the advantage of easy to search and of no loss in stored quality. The continuously declining price of the hard drive presents the opportunity for the DVR system to displace the analog system. Also, with spread of the internet the needs of PC based the DVR system increase. Therefore, we have implemented DVR system within a function of network. When obtained image through the PTZ camera is transmitted to digital form, very large space of storage is required, hence image compression is essential. We use JPEG2000 for compression of image. JPEG2000 adopt DWT by means of transform. DWT concentrates important information of image on subband and has feature of multi-resolution. It is effective in order to express image. Thus JPEG2000 is suitable for image compression in DVR system. The significance of this paper is to design the DVR system which is controlled through TCP/IP network and to implement transmission of image compression using JPEG2000.

• Proceedings of the Korean Information Science Society Conference
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• 2010.06d
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• pp.343-347
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• 2010

TFMCC(TCP-Friendly Multicast Congestion Control)방식은 equation 기반의 멀티캐스트 혼잡 제어 메커니즘으로 TFRC(TCP-Friendly Rate Control) 프로토콜을 유니캐스트에서 멀티캐스트 도메인으로 확장한 방식이다. TFMCC 방식은 현재 무선 환경에 적용 시 유선 환경에서의 혼잡에 의한 패킷 손실뿐만 아니라, 무선 환경에서 무선 링크 에러를 네트워크의 혼잡으로 인식하며, single-rate 멀티캐스트 혼잡제어의 특성인 가장 낮은 수신단의 성능으로 전체 네트워크 전송률이 급격히 저하된다. 이에 본 논문에서는 무선 환경에서의 TFMCC의 성능 향상을 위해 네트워크의 무선 환경의 손실률과 유선 환경 손실률을 모델링하여 구분한 ARC(Analytical Rate Control)의 TCP 전송률 equation 을 TFMCC에 맞게 적용하였으며, 멀티캐스트 도메인에서 전송률 제어 시 무선 손실률을 별도로 고려하는 방식(M-ARC)을 제안하였다. 또한 성능 평가를 위해서 시뮬레이션 한 결과 무선 환경을 고려한 M-ARC(Multicast-Analytical Rate Control)가 기존의 TFMCC에 비해 더 높은 전송률을 유지함을 볼 수 있었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.9B
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• pp.829-837
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• 2006

Due to the internet network congestion, packets may be dropped or delayed at routers. This phenomenon degrades the quality of streaming applications that require high QoS requirements. The proposed algorithm in this paper, called DBRC(Delay-Based Rate Control), tries to cause router queue occupancy to reach a steady state or equilibrium by throttling the transmission rate of the multimedia traffics when network delays tend to increase and also probing for more bandwidth when network delays tend to decrease. Simulation results show that the proposed algorithm provides smooth transmission rate, nearly constant delay and low packet loss rates, compared with TFRC(TCP Friendly Rate Control) that is one of dominant multimedia congestion control algorithms.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.668-672
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• 2009

In wireless ad-hoc network, knowing the available bandwidth of the time varying channel is imperative for live video streaming applications. This is because the available bandwidth is varying all the time and strictly limited against the large data size of video streaming. Additionally, adapting the encoding rate to the suitable bit-rate for the network, where an overlarge encoding rate induces congestion loss and playback delay, decreases the loss and delay. While some effective rate controlling methods have been proposed and simulated well like VTP (Video Transport Protocol) [1], implementing to cooperate with the encoder and tuning the parameters are still challenging works. In this paper, we show our result of the implementation experiment of VTP based encoding rate controlling method and then introduce some techniques of our parameter tuning for a video streaming application over wireless environment.

• Journal of the Korean Society of Food Science and Nutrition
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• v.46 no.2
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• pp.273-278
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• 2017

This study was conducted to investigate the quality characteristics and antioxidant properties of cookies supplemented with 2~8% (w/w) Taraxacum coreanum powder (TCP). The pH and moisture content of cookie dough decreased significantly (P<0.05) while density was not influenced significantly by increasing levels of TCP. The spread ratio and loss rate of cookies increased significantly with increasing levels of TCP (P<0.05). Lightness, redness, and yellowness decreased significantly with higher amount of TCP (P<0.05). The use of TCP significantly increased hardness of cookies while 2,2-diphenyl-1-picrylhydrazyl and 2,2'-azino-bis(3-ethylbenzothiazoline-6-sulfonic acid) radical scavenging activities were significantly elevated (P<0.05). The consumer acceptance test indicated that addition of 2% TCP had a favorable effect on consumer preferences in all attributes. Based on overall observations, cookies with 2% TCP can take advantage of the functional properties of TCP without sacrificing consumer acceptability.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.149-158
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• 2005

This paper proposes a congestion avoidance algorithm which provides stable throughput and transmission rate regardless of buffer size by limiting the TCP congestion window in fixed bandwidth networks. Additive Increase, Multiplicative Decrease (AIMD) is the most commonly used congestion control algorithm. But, the AIMD-based TCP congestion control method causes unnecessary packet losses and retransmissions from the congestion window increment for available bandwidth verification when used in fixed bandwidth networks. In addition, the saw tooth variation of TCP throughput is inappropriate to be adopted for the applications that require low bandwidth variation. We present an algorithm in which congestion window can be limited under appropriate circumstances to avoid congestion losses while still addressing fairness issues. The maximum congestion window is determined from delay information to avoid queueing at the bottleneck node, hence stabilizes the throughput and the transmission rate of the connection without buffer and window control process. Simulations have performed to verify compatibility, steady state throughput, steady state packet loss count, and the variance of congestion window. The proposed algorithm can be easily adopted to the sender and is easy to deploy avoiding changes in network routers and user programs. The proposed algorithm can be applied to enhance the performance of the high-speed access network which is one of the fixed bandwidth networks.

• Journal of Korea Multimedia Society
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• v.12 no.11
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• pp.1593-1608
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• 2009

If a certain relay node in multi-hop wireless networks might become a malicious node that does not cooperate with other nodes or a selfish node, network throughput will be dramatically decreased. Most of existing ad hoc routing protocols assuming that the nodes will fully cooperate with other nodes do not resolve the problem of network performance degradation due to malicious and selfish nodes. This paper presents the CARE (Cooperative Ad hoc routing protocol based REputation) scheme incorporating the reputation management that can achieve a multi-hop wireless network with high throughput performance. The proposed scheme provides the horizontal cross-layer approach which can identify misbehaving malicious, selfish nodes dropped out of the hop-by-hop based packet processing in the network and then set up an optimal packet routing path that will detour misbehaving nodes. And the vertical cross-layer approach contained in the CARE scheme attempts to improve the quality of routing paths by exploiting the quality of link information received from the MAC layer. Besides, it provides high TCP throughput by exploiting the reputation values of nodes acquired from the network layer into the transport layer. A case study on experiments and simulations shows that the CARE scheme incorporating vertical and horizontal cross-layer approaches yields better performance in terms of the low rate of packet loss, fast average packet delivery time, and high TCP throughput between end-to-end nodes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.4B
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• pp.227-233
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• 2007

According to the recent development of network technology, the demands of users are diversified and the needs of multimedia traffic are increasing. In general, UDP(User Datagram Protocol) traffic is used to transport multimedia data, which satisfied the real-time and isochronous characteristics. UDP traffic competes with TCP traffic and incur the network congestion. However, TCP traffic performs network congestion control but does not consider the receiver's status. Thus, it is not appropriate in case of streaming services. In this paper, we solve a fairness problems and proposed a network algorithm based on RTP/RTCP(Real-time Transport Protocol/Realtime Transport Control Protocol) in view of receiver status. The POBA(Priority Ordered Buffer Algorithm), which applies priorities in the receiver's buffer and networks, shows that it provides the appropriate environment for streaming services in view of packet loss ratio and buffer utilization of receiver's buffer compared with the previous method.