• Proceedings of the Korean Information Science Society Conference
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• 2012.06d
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• pp.241-242
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• 2012

본 논문에서는 인터넷 혼잡 상황에서 멀티미디어 스트리밍 서비스의 품질을 향상시키기 위해, 지연을 기반으로 멀티미디어 스트리밍의 전송률을 조절하는 기법을 제안한다. 제안하는 기법에서는 패킷 손실이 발생하기 전에 RTT를 기반으로 네트워크의 혼잡을 인지하고 전송률을 조절함으로써 화질 열화를 줄여 서비스 품질을 향상한다. 실험을 통해 제안기법이 적은 패킷 손실을 발생시켜 스트리밍 품질을 향상시키는 것을 확인하였다.

• Proceedings of the Korean Information Science Society Conference
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• 2006.10d
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• pp.193-198
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• 2006

멀티미디어 데이터를 이종의 수신자에게 안정적으로 전달하고자 하는 것은 인터넷에서 중요한 주제 중의 하나이다. 특히, 최근에 소개된 계층형 MDC(Layered Multiple Description Coding) 기법은 오버레이 멀티캐스트에서 이 문제를 효율적으로 해결하는데 중요한 접근방법이다. 그러나, 이 과정에서 저속 수렴(slow convergence)과 참가 실험(join experiment) 동안에의 손실과 같은 문제가 새롭게 발생하게 되었다. 본 논문에서는 비디오 스트리밍 서비스에서 위의 문제를 해결할 수 있는 효율적인 계층형 멀티캐스트 혼잡 제어 기법을 소개한다. 여기서 제시하는 기법의 가장 특징적인 점은 사용자가 수신할 계층(layer)의 수를 결정하기 위하여 패킷페어(packet-pair)방식에 기반한 수신률 조절 메커니즘을 사용하는 것이다. 결과적으로 본 논문에서는 수신자가 최적의 전송률에 빠르게 수렴하면서도 손실을 최소화할 수 있는 종단-대-종단간 혼잡 제어 기법을 제시한다.

• Proceedings of the Korean Information Science Society Conference
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• 2004.04a
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• pp.526-528
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• 2004

본 논문은 유무선 환경에서 TCP를 이용한 데이터 전송 시 에이젼트를 이용하여 패킷 손실의 원인을 분석, 무선 링크에서 발생한 패킷 손실에 대해서는 혼잡 윈도우 크기를 유지하고, 유선 링크에서 발생한 패킷 손실에 대해서는 지역 재전송을 수행하는 저 전력 전송기법을 제안한다. 제안하는 저 전력 전송기법은 전송 후 WNIC를 저 전력 모드로 전환시킴으로써 WNIC 전력소비를 최소화한다. NS2 시뮬레이션 결과 기존 TCP 보다 무선 링크에서 에러 발생시 67~177(%) 성능향상과 22~44(%) 에너지 감소효과를 보였고, 유선 링크에서 에러 발생시 3~22(%)의 성능 향상과 2~13(%) 에너지 감소 효과를 나타냈다.

• 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.

• 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.

• Journal of Advanced Navigation Technology
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• v.14 no.3
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• pp.358-366
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• 2010

In the wireless network the congestion and delay occurs mainly when there are too many packets for the network to process or the sender transmits more packets than the receiver can accept. The congestion and delay is the reason of packet loss which degrades the performance of multimedia streaming. This paper proposes a novel transmission rate monitoring-based optimization mechanism to optimize packet loss and to improve QoS. The proposed scheme is based on the trade-off relationship between transmission rate monitoring and overhead monitoring. For this purpose this paper processes a source rate control-based optimization which optimizes congestion and delay. Performance evaluated RED, TFRC, and the proposed mechanism. The simulation results show that the proposed mechanism is more efficient than REC(Random Early Detection) mechanism and TFRC(TCP-friendly Rate Control) mechanism in packet loss rate, throughput rate, and average response rate.

• Journal of KIISE:Information Networking
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• v.41 no.4
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• pp.177-185
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• 2014

It has been occurred many times that wireless sensor networks (WSNs) had congested areas because all the sensing data collected by multiple sensor nodes are delivered to one sink node. Typically, in order to control congested areas, it used to reduce the traffic by increasing the sensing period or discarding packets. However, those schemes have a disadvantage that it loses the reliability when delivering sensing data. Moreover, there are no schemes considering case of having poor quality of links between nodes in practical terms. In this paper, we suggest a scheme not to reduce the traffic but to distribute the traffic by changing routing paths depends on the traffic and the quality of links. Also, it can be seen that the reliability of delivering of the sensing data is improved with the experiments improving collection rates and shortening end-to-end delay.

• Journal of KIISE
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• v.43 no.8
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• pp.919-926
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• 2016

In a wireless network, TCP causes the performance degradation because of mistaking packet loss, which is caused by characteristics of wireless link and throughput oscillation due to change of devices connected on a limited bandwidth. Delay based TCP is not affected by packet loss because it controls window size by using the RTT. Therefore, it can solve the problem of unnecessary degradation of the rate caused by misunderstanding reason of packet loss. In this paper, we propose an algorithm for improving the remaining problems by using delay based TCP. The proposed scheme can change throughput adaptively by adding the RTT, which rapidly reflects the network conditions to BaseRTT. It changes the weight of RTT and the increases and decreases window size based on the remaining amount of the buffer. The simulation indicated that proposed scheme can alleviate the throughput oscillation problem, as compared to the legacy TCP Vegas.

• Journal of KIISE:Information Networking
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• v.31 no.3
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• pp.269-279
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• 2004

The most important factor influencing the robustness of the Internet Is the end-to-end TCP congestion control. However, the congestion control scheme of TCP Reno, the most popular TCP version on the Internet, employs passive congestion indication. It makes worse the network congestion. Recently, Brakmo and Peterson have proposed a new version of TCP, which is named TCP Vegas, with a fundamentally different congestion control scheme from that of the Reno. Many studies indicate that the Vegas is able to achieve better throughput and higher stability than the Reno. But there are two unfairness problems in Vegas. These problems hinder the spread of the Vegas in current Internet. In this paper, in order to solve these unfairness problems, we propose a new congestion control algorithm called TCP PowerVegas. The existing Vegas depends mainly only on the rtt(round trip time), but the proposed PowerVegas use the new congestion control scheme combined the Information on the rtt with the information on the packet loss. Therefore the PowerVegas performs the congestion control more competitively than the Vegas. Thus, the PowerVegas is able to solve effectively these unfairness problems which the Vegas has experienced. To evaluate the proposed approach, we compare the performance among PowerVegas, Reno and Vegas under same network environment. Using simulation, the PowerVegas is able to achieve better throughput and higher stability than the Reno and is shown to achieve much better fairness than the existing Vegas.

• Proceedings of the Korean Information Science Society Conference
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• 2006.06d
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• pp.13-15
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• 2006

유선망에 최적화되도록 진화해온 TCP는 무선망이 가지는 링크의 불안정함으로 인한 손실을 네트워크의 혼잡으로 인한 손실로 오해한다. 그 결과 혼잡 제어 메커니즘이 수행되어 불필요하게 전송율을 줄이므로써 전송 성능을 저하시키는 문제점을 초래한다. 이러한 이유로 최근 유 무선 혼합망에서 TCP의 성능을 향상시키기 위한 연구가 활발히 이루어지고 있다. 본 논문에서는 기존에 제안된 성능향상 기법들 중 상대적으로 뛰어난 성능을 보이는 Snoop 프로토콜이 유 무선 혼합망 특히 IEEE 802.11 MAC 프로토콜을 사용하는 무선망에서 가지는 문제점을 분석하고, Cross-layering 기법을 통하여 이를 보완하는 기법을 제안한다.