• Proceedings of the Korean Information Science Society Conference
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• 2000.04a
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• pp.271-273
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• 2000

본 논문에서는 TCP 수신자가 지연 ACK 옵션(Delayed ACK Option)을 사용할 경우에 TCP 송신자에게 발생하는 성능 저하요인들을 분석하고 다음과 같은 해결책을 제시하였다. 먼저, 느린 시작 구간(Slow Start phase) 처음에 생기는 ACK 타임아웃은 큰 초기 윈도우(large initial window)또는 1-bit 마킹 기법을 통해 해결할 수 있다. 그리고, 느린 시작 구간과 혼잡 회피 구간(Congestion Avoidance phase)에서 혼잡 윈도우(cwnd)가 천천히 증가하는 문제는 적절히 바이트 카운팅 기법을 사용함으로써 해결할 수 있다. 마지막으로, 송신자가 버스트(burst)한 데이터를 네트웍에 발생시키는 문제는 트래픽을 평활(pacing)함으로써 해결할 수 있다. 또한 본 연구에서는 분석적 모델링을 통하여 TCP가 보내는 평균 전송률을 구하였으며 이 결과는 TCP에 친화한 전송률 기반 전송방법(TCP Friendly Rate Based Control)에 응용될 수 있을 것이다. 그리고 시뮬레이션을 통해서 제시한 방법의 성능이 향상됨을 확인하였다.

• Proceedings of the Korean Information Science Society Conference
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• 2002.04a
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• pp.508-510
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• 2002

인터넷 수요의 증대에 따라 오디오나 비디오 같은 실시간 전송을 요구하는 멀티미디어 데이터 응용들도 증가하게 되었다. 이런, 멀티미디어 데이터들의 전송은 패킷 손실과 지연을 최소화 하고, 대역폭 상태에 대해 TCP보다 순화된 전송률 조절이 가능한 혼잡제어 기법이 필요하다. Receiver-based 혼잡 제어방식은 network traffic의 변화에 신속히 대응할 수 있으며, 이 방식은 Sender-based 방식에 비해 멀티캐스팅 환경에 적당하다는 장점도 가진다. 본 논문에서는 인터넷 상에서 audio/video 데이터들의 종 단간 Layered Quality Adaptation을 제공하는 receiver-based 혼잡 제어 기법을 소개한다. 이를 통해 사용자들은 보다 좋은 QoS를 제공받을 수 있다.

• Journal of KIISE:Information Networking
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• v.29 no.5
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• pp.495-500
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• 2002

We propose a new estimation method for rate adjustment in the face of a packet loss in the TFRC protocol, a TCP-Friendly congestion control protocol for UDP flows. Previous methods respond in a sensitive way to a single packet loss, resulting in oscillatory transmission behavior. This is an undesirable for multimedia services demanding constant bandwidth. The proposed TFRC provides more smooth and fair (against TCP flows) transmission through collective response based on multiple packets loss events. We show our "Exponential smoothing method" performs better than known "Weight smoothing method" in terms of smoothness and fairness.

• Journal of KIISE:Information Networking
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• v.37 no.5
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• pp.363-373
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• 2010

The existing QoS mechanisms for video streaming are short of the consideration for various user environments and the characteristic of streaming applying programs. In order to overwhelm this problem, studies on the video streaming protocols exploiting scalable video coding (SVC), which provide spatial, temporal, and qualitative scalability in video coding, are progressing actively. However, these protocols also have the problem to deepen network congestion situation, and to lower fairness between other traffics, as they are not equipped with congestion control mechanisms. SVC based streaming protocols also have the problem to overlook the property of videos encoded in SVC, as the protocols transmit the streaming simply by extracting the bitstream which has the maximum bit rate within available bandwidth of a network. To solve these problems, this study suggests TCP-friendly network adaptive SVC streaming(T-NASS) protocol which considers both network status and SVC bitstream property. T-NASS protocol extracts the optimal SVC bitstream by calculating TCP-friendly transmission rate, and by perceiving the network status on the basis of packet loss rate and explicit congestion notification(ECN). Through the performance estimation using an ns-2 network simulator, this study identified T-NASS protocol extracts the optimal bitstream as it uses TCP-friendly transmission property and perceives the network status, and also identified the video image quality transmitted through T-NASS protocol is improved.

• Journal of Korea Multimedia Society
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• v.11 no.2
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• pp.197-205
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• 2008

TCP packets occupy over 90% of current Internet traffic thus understanding of TCP throughput is crucial to understand Internet. Under the TCP congestion regime, heterogeneous flows, i.e., flows with different round-trip times (RTTs), that share the same bottleneck link will not attain equal portions of the available bandwidth. In fact, according to the TCP friendly formula, the throughput ratio of two flows is inversely proportional to the ratio of their RTTs. It has also been shown that TCP's unfairness to flows with longer RTTs is accentuated under loss synchronization. In this paper, we show that, injecting bursty background traffic may actually lead to new type of synchronization and result in unfairness to foreground TCP flows with longer RTTs. We propose three different metrics to characterize traffic burstiness and show that these metrics are reliable predictors of TCP unfairness.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.652-657
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• 2004

As the real-time multimedia applications through Internet increase, the bandwidth available to TCP connections is oppressed by the UDP traffic, result in the performance of overall system is extremely deteriorated. Therefore, developing a new transmission protocol is necessary. The TCP-friendly algorithm is an example meeting this necessity. The TCP-friendly (TFRC) is an UDP-based protocol that controls the transmission rate based on the available round transmission time (RTT) and the packet loss rate (PLR). In the data transmission processing, transmission rate is determined based on the conditions of the previous transmission period. If the one-step ahead predicted values of the control factors are available, the performance will be improved significantly. This paper proposes a prediction model of transmission rate control factors that will be used for the transmission rate control, which improves the performance of the networks. The model developed through this research is predicting one-step ahead variables of RTT and PLR. A multiplayer perceptron neural network is used as the prediction model and Levenberg-Marquardt algorithm is used for the training. The values of RTT and PLR were collected using TFRC protocol in the real system. The obtained prediction model is validated using new data set and the results show that the obtained model predicts the factors accurately.

• Journal of Communications and Networks
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• v.16 no.3
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• pp.344-354
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• 2014

Transmission control protocol friendly rate control (TFRC) is designed to mainly provide optimal service for unicast applications, such as multimedia streaming in the best-effort Internet environment. However, high bandwidth networks with large delays present an environment where TFRC may have a problem in utilizing the full bandwidth. TFRC inherits the slow-start mechanism of TCP Reno, but this is a time-consuming process that may require many round-trip-times (RTTs), until an appropriate sending rate is reached. Another disadvantage inherited from TCP Reno is the RTT-unfairness problem, which severely affects the performance of long-RTT flows. In this paper, we suggest enhanced TFRC for high quality video streaming over high bandwidth delay product networks. First, we propose a fast startup scheme that increases the data rate more aggressively than the slow-start, while mitigating the overshooting problem. Second, we propose a bandwidth estimation method to achieve more equitable bandwidth allocations among streaming flows that compete for the same narrow link with different RTTs. Finally, we improve the responsiveness of TFRC in the presence of severe congestion. Simulation results have shown that our proposal can achieve a fast startup and provide fairness with competing flows compared to the original TFRC.

• 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.

• Journal of Institute of Control, Robotics and Systems
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• v.11 no.4
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• pp.385-391
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• 2005

As the Internet real-time multimedia applications increases, the bandwidth available to TCP connections is oppressed by the UDP traffic, result in the performance of overall system is extremely deteriorated. Therefore, developing a new transmission protocol is necessary. The TCP-friendly algorithm is an example satisfying this necessity. The TCP-Friendly Rate Control (TFRC) is an UDP-based protocol that controls the transmission rate that is based on the available round trip time (RTT) and the packet loss rate (PLR). In the data transmission processing, transmission rate is determined based on the conditions of the previous transmission period. If the one-step ahead predicted values of the control factors are available, the performance will be improved significantly. This paper proposes a prediction model of transmission rate control factors that will be used in the transmission rate control, which improves the performance of the networks. The model developed through this research is predicting one-step ahead variables of RTT and PLR. A multiplayer perceptron neural network is used as the prediction model and Levenberg-Marquardt algorithm is used for the training. The values of RTT and PLR were collected using TFRC protocol in the real system. The obtained prediction model is validated using new data set and the results show that the obtained model predicts the factors accurately.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.8
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• pp.1679-1684
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• 2004

This paper designs and implements a VoIP module for embedded system. Many other VoIP modules for desktop environment has widely implemented but the one for embedded system is not prevalent yet. The implemented VoIP module uses the OpenH.323 and is operated on the WinCE PDA. We use the TCP-friendly adaptation for congestion control. The experiment results show smooth communication but a little noise. Future works include the minimization of noise.