• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.12B
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• pp.1106-1114
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• 2003

Several analytic models describe transmission control protocol (TCP) performance such as steady-state throughput as an averaged ratio of number of transmissions to latency. For more detailed analysis of TCP latency, the latency during packet losses are recovered should be considered. In this paper, we derive the expected duration of loss recovery latency considering the number of packet losses recovered by retransmissions. Based on the numerical results verified by simulations, TCP using selective acknowledgement (SACK) option is more effective than TCP NewReno from the aspect of loss recovery latency.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.4B
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• pp.288-297
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• 2003

The fast retransmit and fast recovery algorithm of TCP Reno, when multiple packets in the same window are lost, cannot recover them without RTO (Retransmission Timeout). TCP New-Reno can recover multiple lost packets by extending fast recovery using partial acknowledgement. If the retransmitted packet is lost again during fast recovery, however, RTO cannot be avoided. In this paper, we propose an algorithm called "Duplicate Acknowledgement Counting(DAC)" to alleviate this problem. DAC can detect the retransmitted packet loss by counting duplicate ACKs. Conditions that a lost packet can be recovered by loss recovery of TCP Reno, TCP New-Reno and TCP New-Reno using DAC are derived by modeling loss recovery behavior of each TCP. We calculate the loss recovery probability for random packet loss probability numerically, and show that DAC can improve loss recovery behavior of TCP New-Reno.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.660-666
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• 2004

Overall TCP performance represented by end-to-end throughput is largely dependent upon its loss recovery performance. In particular non-congestion packet losses caused by transmission errors degrade TCP performance seriously. Using Markov process, we analyze TCP loss recovery performance for correlated packet losses caused by multipath fading. The results show that loss recovery performance can be severely affected by burstiness in packet losses, even if overall packet loss ratio is very low.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.667-674
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• 2004

The performance of transmission control protocol (TCP) is largely dependent upon its loss recovery. Therefore, it is a very important issue whether the packet losses may be recovered without retransmission timeout (RTO) or not. Although TCP SACK can recover multiple packet losses in a window, it cannot avoid RTO if a retransmitted packet is lost again. In order to alleviate this problem, we propose a simple change to TCP SACK, which is called TCP SACK+ in simple. We use a stochastic model to evaluate the performance of TCP SACK+, and compare it with TCP SACK. Numerical results evaluated by simulations show that SACK+ can improve the loss recovery of TCP SACK significantly in presence of random losses.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.4
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• pp.741-746
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• 2012

Transmission control protocol(TCP) widely used as a transport protocol in the Internet includes a loss recovery function that detects and recovers packet losses by retransmissions. The loss recovery function consists of the two algorithms; fast retransmit and fast recovery. There have been researches to avoid nonnecessary retransmission timeouts (RTOs), which leads to selective acknowledgement (SACK) option and limited transmit scheme that are standardized by IETF (Internet Engineering Task Force). Recently, a method that covers the case in which a retransmitted packet is lost again has been propsed. The method, however, is not proved in terms of the additive increase multiplicative decrease (AIMD) principle of TCP congestion control. In this paper, therefore, we analyzed the method in terms of the principle by ns-simulations.

• Journal of KIISE:Information Networking
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• v.32 no.3
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• pp.382-390
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• 2005

As today's networks evolve towards an If-based integrated network, the role of transmission control protocol(TCP) has been increasing as well. As a well-known issue, the performance of TCP is affected by its loss recovery mechanism that is comprised of two algorithms; fast retransmit and fast recovery. Although retransmission timeout(RTO) caused by multiple packet losses can be avoided by using selective acknowledgement(SACK) option, RTO cannot be avoided if a retransmitted packet is lost. Therefore, we propose a simple modification to make it possible for a TCP sender using SACK option to detect a lost retransmission. In order to evaluate the proposed algorithm, simulations have been performed for two scenarios where packet losses are random and correlated. Simulation results show that the proposed algorithm can improve TCP performance significantly.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.675-684
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• 2004

There have been several efforts to avoid unnecessary retransmission timeouts (RTOs), which is the main cause for TCP throughput degradation. Unnecessary RTOs can be classified into three groups according to their cause. RTOs due to multiple packet losses in the same window for TCP Reno, the most prevalent TCP version, can be avoided by TCP NewReno or using selective acknowledgement (SACK) option. RTOs occurring when a packet is lost in a window that is not large enough to trigger fast retransmit can be avoided by using the Limited Transmit algorithm. In this Paper, we comparatively analyze these schemes to cope with unnecessary RTOs by numerical analysis and simulations. On the basis of the results in this paper, TCP performance can be quantitatively predicted from the aspect of loss recovery probability. Considering that overall performance of TCP is largely dependent upon the loss recovery performance, the results shown in this paper are of great importance.

• Journal of Digital Contents Society
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• v.10 no.2
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• pp.279-289
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• 2009

As the TCP is the protocol designed for the wired network that packet loss probability is very low, because TCP transmitter takes it for granted that the packet loss by the wireless network characteristics is occurred by the network congestion and lowers the transmitter's transmission rate, the performance is degraded. In this article, we suggest the newly improved algorithm using two parameters, the local retransmission time value and the local retransmission critical value to the BS based on the Snoop. This technique adjusts the base stations local retransmission timer effectively according to the wireless link status to recover the wireless packet loss rapidly. We checked that as a result of the suggested algorithm through various simulations, A-Snoop protocol improve more the wireless TCP transmission rate by recovering the packet loss effectively in the wireless link that the continuous packet loss occur than the Snoop protocol.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.3B
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• pp.130-137
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• 2005

Wireless communication environment is changing rapidly as we use new wireless communication technology such as WiBro to access high speed Internet. As a result, reliable data transmission using TCP is also expected to increase. Since TCP assumes that it is used in wired network, TCP suffers significant performance degradation over wireless network where packet losses are related to non-congestion loss. Especially RTO imposes a great performance degradation of TCP. In this paper, we analyze the loss recovery probabilities based on previous researches, and use simulation results of our algorithm to show that it prevents performance degradation by quickly detecting and recovery losses without RTO during fast recovery.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1496-1502
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• 1998

In this paper, a real-time audio broadcasting service system which is robust to loaded traffic on the Internet is developed. For implementing reliable real-time data transfer, the transfer characteristics of TCP/IP and UDP/IP was compared and analyzed. For lost packet recovery, redundant audio data algorithm was used and interleaving technique was applied for scattering consecutive packet loss. Test results showed, when using TCP/IP, pause occurred during playback, and when using UDP/IP, a stable receive rate was noticeable but the quality of the sound was lower than that of uisng TCP/IP. The recovery rate using redundant audio data and interleaving technique is shown in Fig. 9 and the delay is shown in Fig 4.