• Journal of Communications and Networks
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• v.6 no.3
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• pp.235-244
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• 2004

There have been a lot of approaches to evaluate and predict transmission control protocol (TCP) performance in a numerical way. Especially, under the recent advance in wireless transmission technology, the issue of TCP performance over wireless links has come to surface. It is because TCP responds to all packet losses by invoking congestion control and avoidance algorithms, resulting in degraded end-to-end performance in wireless and lossy systems. By several previous works, although it has been already proved that overall TCP performance is largely dependent on its loss recovery performance, there have been few works to try to analyze TCP loss recovery performance with thoroughness. In this paper, therefore, we focus on analyzing TCP's loss recovery performance and have developed a simple model that facilitates to capture the TCP sender's behaviors during loss recovery period. Based on the developed model, we can derive the conditions that packet losses may be recovered without retransmission timeout (RTO). Especially, we have found that TCP Reno can retransmit three packet losses by fast retransmits in a specific situation. In addition, we have proved that successive three packet losses and more than four packet losses in a window always invoke RTO easily, which is not considered or approximated in the previous works. Through probabilistic works with the conditions derived, the loss recovery performance of TCP Reno can be quantified in terms of the number of packet losses in a window.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.474-479
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• 2004

This paper describes a feedback-based congestion control algorithm to improve TCP performance over wireless network. In this paper, we adjust the packet marking probability at the router for Max-Min fair sharing of the bandwidth and full utilization of the link. Using the successive ECN (Explicit Congestion Notification), the proposed algorithm regulates the window size to avoid the congestion and sees the packet loss only due to the wireless link error. Based on the asymptotic analysis, it is shown that the proposed algorithm guarantees the QoS of the wireless TCP. The effectiveness of the proposed algorithm is demonstrated by simulations.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.9
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• pp.3502-3509
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• 2010

Streaming service in the wireless mobile network environment has been a very challenging issue due to the dynamic uncertain nature of the channels. Overhead such as congestion, latency, and jitter lead to the problem of performance degradation of an adaptive multi-streaming service. This paper proposes a AMSS (Adaptive Multi-level Streaming Service) mechanism to reduce the performance degradation due to overhead such as variable network bandwidth, mobility and limited resources of the wireless mobile network. The proposed AMSS optimizes streaming services by: 1) use of fuzzy similarity metric, 2) minimization of packet loss due to buffer overflow and resource waste, and 3) minimization of packet loss due to congestion and delay. The simulation result shows that the proposed method has better performance in congestion control and packet loss ratio than the other existing methods of TCP-based method, UDP-based method and VBM-based method. The proposed method showed improvement of 10% in congestion control ratio and 8% in packet loss ratio compared with VBM-based method which is one of the best method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.11
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• pp.2553-2562
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• 2013

It is difficult for TCP congestion control algorithm to ensure the bandwidth and delay bound required for media streaming services in broadband wireless network environments. In this paper, we propose the COIN TCP (COncave INcrease TCP) scheme for providing a high-quality media streaming services. The COIN TCP concavely increases the congestion window size by adjusting the increment rate of congestion window, that is inversely proportional to the amount of data accumulated in the router queue. As a result, our scheme can quickly occupy the available bandwidth and prevent the heavy congestion. It also improves the link utilization by adjusting the decrement rate of congestion window according to the packet loss rate with the random loss. Through the simulation results, we prove that our scheme improves the total throughput in broadband wireless network.

• Journal of Advanced Navigation Technology
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• v.16 no.4
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• pp.641-648
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• 2012

The up-streams of the continuous streaming of data packets with lower importance level in the wireless sink node can cause congestion and delay, they affect on energy efficiency, memory size, buffer size, and throughput. This paper proposes a new wireless sink congestion control mechanism based on tournament scheduling. The proposed method consists of two module parts: stream decision module part and service differentiation module part. The final winner in the tournament controls congestion effectively, minimizes packet loss due to congestion, decreases energy consumption, and improves QoS. The simulation result shows that the proposed method is more effective and has better performance compared with those of congestion descriptor-based control method, reliability-based control method, and best-effort transmission control method.

• 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.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.2
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• pp.9-17
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• 2012

Because best-effort Internet provides no guarantees on packet delay and loss, transient network congestion may cause negative effects on H.264/SVC streaming. Thus, the congestion control is required to adjust bit rate by dropping enhancement layers of H.264/SVC streams. This paper differentiates the video streams according to different levels of importance and proposes weighted-based congestion control algorithms to use the rate-distortion characteristics of streams. To maximize the weighted sum of PSNR values of all streams on a bandwidth-constrained node, this paper proposes WNS(Weighted Near-Sighted) and WFS(Weighted Far-Sighted) algorithms to control the number of enhancement layers of streams. Through simulation, this paper shows that weighted-based congestion control algorithm can efficiently adapt streams to network conditions and analyzes the characteristics of congestion control algorithms.

• The KIPS Transactions:PartC
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• v.16C no.2
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• pp.237-244
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• 2009

The congestion in wireless sensor network increases the ratio of data loss and causes the delay of data. The existing congestion protocols for wireless sensor network reduces the amount of transmission by control the sampling frequency of the sensor nodes related to the congestion when the congestion has occurred and was detected. However, the control method of sampling frequency is not applicable on the situation which is sensitive to the temporal data loss. In the paper, we propose a new congestion control, ACT - Adaptive Congestion conTrol. The ACT monitors the network traffic with the queue usage and detects the congestion based on the multi level threshold of queue usage. Given network congestion, the ACT increases the efficiency of network by adaptive flow control method which adjusts the frequency of packet transmission and guarantees the fairness of packet transmission between nodes. Furthermore, ACT increases the quality of data by using the variable compression method. Through experiment, we show that ACT increases the network efficiency and guarantees the fairness to sensor nodes compared with existing method.

• Proceedings of the IEEK Conference
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• 2002.07c
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• pp.1831-1834
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• 2002

In this paper, we present results of a study of congestion control for real-time communication based on ATM networks. In ATM networks, congestion usually results in cell loss. Based on the time limit and priority, the cells that compete for the same output line could be lined according to the character of real-time service. We adopt priority control algorithm for providing different QoS bearer services that can be implemented by using threshold methods at the ATM switching nodes, the cells of different deadline and priority could be deal with according to the necessity. Experiments show the proposed algorithm is effective in the congestion control of ATM real-time networks

• The KIPS Transactions:PartC
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• v.12C no.4 s.100
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• pp.571-580
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• 2005

In fast long-distance networks, TCP's congestion control algorithm has the problem of utilizing bandwidth effectively. Several window-based congestion control protocols for high-speed and large delay networks have been proposed to solve this problem. These protocols deliberate mainly three properties : scalability, TCP-friendliness, and RTT-fairness. These protocols, however, cannot satisfy above three properties at the same time because of the trade-off among them This paper presents a new window-based congestion control algorithm, called EM (Exponential Increase/ Multiplicative Decrease), that simultaneously supports all four properties including fast convergence, which is another important constraint for fast long-distance networks; it can support scalability by increasing congestion window exponentially proportional to the time elapsed since a packet loss; it can support RTT-fairness and TCP-friendliness by considering RTT in its response function; it can support last fair-share convergence by increasing congestion window inversely proportional to the congestion window just before packet loss. We evaluate the performance of EIMD and other algorithms by extensive computer simulations.