• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 한국해양정보통신학회 2003년도 추계종합학술대회
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• pp.149-153
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• 2003

네트워크를 설계하고 서비스를 구현하는데 있어서 중요한 변수중의 하나는 트래픽의 특성을 파악하는 것이다. 트래픽 특성에 관한 최근의 실험적 연구들은 기존의 모델들이 실제 트래픽의 특성을 제대로 나타낼 수 없다는 것을 지적해 왔고 최근 실제 트래픽 모델과 유사한 모델로서 자기 유사한 특성을 이용한 접근법이 대두되고 있다. 따라서, 본 논문에서는 실제 트래픽과 유사한 자기 유사 데이터 트래픽을 백그라운드 부하로 발생시켜 기존의 ABR(Available Bit Rate) 혼잡제어 알고리즘 중 fairshare을 지원하는 대표적인 EPRCA(Enhanced Proportional Rate Control Algorithm), ERI-CA(Explicit Rate Indication for Congestion Avoidance), NIST(National Institute of Standards and Technology) 스위치 알고리즘이 버스트한 트래픽에 대해 효율적으로 fairshare을 할당하는지를 알아 보았다.

• Journal of Communications and Networks
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• 제8권3호
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• pp.351-359
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• 2006

In this article, we present a new slow-start variant, which improves the throughput of transmission control protocol (TCP) Vegas. We call this new mechanism Gallop-Vegas because it quickly ramps up to the available bandwidth and reduces the burstiness during the slow-start phase. TCP is known to send bursts of packets during its slow-start phase due to the fast window increase and the ACK-clock based transmission. This phenomenon causes TCP Vegas to change from slow-start phase to congestion-avoidance phase too early in the large bandwidth-delay product (BDP) links. Therefore, in Gallop-Vegas, we increase the congestion window size with a rate between exponential growth and linear growth during slow-start phase. Our analysis, simulation results, and measurements on the Internet show that Gallop-Vegas significantly improves the performance of a connection, especially during the slow-start phase. Furthermore, it is implementation feasible because only sending part needs to be modified.

• Journal of Communications and Networks
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• 제6권3호
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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.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제8권8호
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• pp.2796-2813
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• 2014

This paper presents a new algorithm called the adaptive logarithmic increase and adaptive decrease algorithm (A-LIAD), which mainly addresses the Round-Trip Time (RTT) fairness problem in satellite networks with a very high propagation delay as an alternative to the current TCP congestion control algorithm. We defined a new increasing function in the fashion of a logarithm depending on the increasing factor ${\alpha}$, which is different from the other logarithmic increase algorithm adopting a fixed value of ${\alpha}$ = 2 leading to a binary increase. In A-LIAD, the ${\alpha}$ value is derived in the RTT function through the analysis. With the modification of the increasing function applied for the congestion avoidance phase, a hybrid scheme is also presented for the slow start phase. From this hybrid scheme, we can avoid an overshooting problem during a slow start phase even without a SACK option. To verify the feasibility of the algorithm for deployment in a high-speed and long-distance network, several aspects are evaluated through an NS-2 simulation. We performed simulations for intra- and interfairness as well as utilization in different conditions of varying RTT, bandwidth, and PER. From these simulations, we showed that although A-LIAD is not the best in all aspects, it provides a competitive performance in almost all aspects, especially in the start-up and packet loss impact, and thus can be an alternative TCP congestion control algorithm for high BDP networks including a satellite network.

• The KIPS Transactions:PartC
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• 제10C권1호
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• pp.51-60
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• 2003

One of the most important matters in designing network and realizing service, is to grip on the traffic characteristics. Conventional traffic prediction and analysis used the models which based on the Poisson or Markovian. Recently, experimental research on the LAN, WAN and VBR traffic properties have been pointed rut that they weren't able to display actual real traffic specificities because the models based on the Poisson assumption had been underestimated the long range dependency of network traffic and self-similar peculiarities, it has been lately presented that the new approach method using self-similarity characteristics as similar as the real traffic models. Therefore, in this paper, we generated self-similar data traffic like real traffic as background load. On the existing ABR congestion control algorithm, we analyzed by classify into ACR, buffer utilization. cell drop rate, transmission throughput with the representative EFCI, ERICA, EPRCA and NIST twitch algorithm to show the efficient reaction about the burst traffic.

• Journal of KIISE:Information Networking
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• 제32권1호
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• pp.12-19
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• 2005

Reliable transport protocols such as TCP are tuned to Perform well in traditional networks where packet losses occur mainly because of congestion. In a wireless network, however, packet losses will occur more often due to reasons such as the high bit error rate and the handoff rather than due to congestion. When using TCP over wireless network, TCP responds to losses due to the high bit error rate and the handoff by invoking congestion control and avoidance algorithms, resulting in the degraded end-to-end performance in the wireless network. There have been several schemes for improving TCP performance over wireless links. Among them, SNOOP Is a very promising scheme because of the localized retransmission. In this thesis, an efficient scheme is proposed by modifying SNOOP scheme. The invocation of congestion control mechanism is now minimized by knowing the cause of packet loss.

• Journal of the Institute of Electronics Engineers of Korea TC
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• 제42권12호
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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.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 한국신호처리시스템학회 2000년도 하계종합학술대회논문집
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• pp.9-12
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• 2000

본 논문에서는 기존의 지상 ATM망의 제어기법을 바탕으로 위성망으로 확대 적용할 수 방안으로 기존의 지상 ATM망의 막힘제어 기법중 ERICA스위치 알고리즘을 적용하였으며, 기존의 ERICA스위치 알고리즘은 지상과 위성간의 긴 전파지연시간을 고려하지 않아 위성내의 막힘을 알려주는 정보가 지구국까지 전달되어 다시 위성에 전달될 때 위성의 막힘 상태변화로 인한 손실이 발생할 수 있다 본 논문에서는 이러한 손실을 피하기 위한 제어구조에 대해서 다루었다.

• The Journal of the Korea institute of electronic communication sciences
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• 제1권2호
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• pp.102-109
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• 2006

In this paper, we improve the performance of bidirectional TCP connection over end-to-end network that uses transfer rate-based flow and congestion control. The sharing of a common buffer by TCP packets and acknowledgement has been known to result in an effect called ack compression, where acks of a connection arrive at the source bunched together, resulting in unfairness and degraded throughput. The degradation in throughput due to bidirectional traffic can be significant. For example, even in the simple case of symmetrical connections with adequate window size, the connection efficiency is improved about 20% for three levels of background traffic 2.5Mbps, 5.0Mbps and 7.5Mbps. Otherwise, the throughput of jitter is reduced about 50% because round trip delay time is smaller between source node and destination node. Also, we show that throughput curve is improved with connection rate algorithm which is proposed for TCP congestion avoidance as a function of aggressiveness threshold for three levels of background traffic 2.5Mbps, 5Mbps and 7.5Mbps.

• Journal of Internet Computing and Services
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• 제3권6호
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• pp.43-51
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• 2002

In this paper, we study the new packet dropping scheme using an active queue management algorithm. Active queue management mechanisms differ from the traditional drop tail mechanism in that in a drop tail queue packets are dropped when the buffer overflows, while in active queue management mechanisms, packets may be dropped early before congestion occurs, However, it still incurs high packet loss ratio when the buffer size is not large enough, By detecting congestion and notifying only a randomly selected fraction of connection, RED causes to the global synchronization and fairness problem. And also, it is the biggest problem that the network traffic characteristics need to be known in order to find the optimum average queue length, We propose a new efficient packet dropping method based on the active queue management for congestion control. The proposed scheme uses the per-flow rate and fair share rate estimates. To this end, we present the estimation algorithm to compute the flow arrival rate and the link fair rate, We shows the proposed method improves the network performance because the traffic generated can not cause rapid fluctuations in queue lengths which result in packet loss