• 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.

• 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.

• Journal of Communications and Networks
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• v.4 no.2
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• pp.108-117
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• 2002

TCP, which was developed on the basis of wired links, supposes that packet losses are caused by network congestion. In a wireless network, however, packet losses due to data corruption occur frequently. Since TCP does not distinguish loss types, it applies its congestion control mechanism to non-congestion losses as well as congestion losses. As a result, the throughput of TCP is degraded. To solve this problem of TCP over wireless links, previous researches, such as split-connection and end-to-end schemes, tried to distinguish the loss types and applied the congestion control to only congestion losses; yet they do nothing for non-congestion losses. We propose a novel transport protocol for wireless networks. The protocol called VS-TCP (Variable Segment size Transmission Control Protocol) has a reaction mechanism for a non-congestion loss. VS-TCP varies a segment size according to a non-congestion loss rate, and therefore enhances the performance. If packet losses due to data corruption occur frequently, VS-TCP decreases a segment size in order to reduce both the retransmission overhead and packet corruption probability. If packets are rarely lost, it increases the size so as to lower the header overhead. Via simulations, we compared VS-TCP and other schemes. Our results show that the segment-size variation mechanism of VS-TCP achieves a substantial performance enhancement.

• ETRI Journal
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• v.25 no.5
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• pp.356-368
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• 2003

This paper analyzes the characteristics of packet losses in mobile computing environments based on the Gilbert model and then describes a mechanism that can recover the lost audio packets using redundant data. Using information periodically reported by a receiver, the sender dynamically adjusts the amount and offset values of redundant data with the constraint of minimizing the bandwidth consumption of wireless links. Since mobile computing environments can be often characterized by frequent and consecutive packet losses, loss recovery mechanism need to deal efficiently with both random and consecutive packet losses. To achieve this, the suggested mechanism uses relatively large, discontinuous exponential offset values. That gives the same effect as using both the sequential and interleaving redundant information. To verify the effectiveness of the mechanism, we extended and implemented RTP/RTCP and applications. The experimental results show that our mechanism, with an exponential offset, achieves a remarkably low complete packet loss rate and adapts dynamically to the fluctuation of the packet loss pattern in mobile computing environments.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.39 no.5
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• pp.21-31
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• 2002

As the MPEG streaming traffics are transmitted over the Internet, MPEG packet losses cause error propagation in both spatial and temporal domains which in turn leads to severe degradation in image quality. This image degradation is become worse by the bursty packet loss at the network. In this paper, we propose a new error spreading method to reduce the effects of bursty packet losses over the Internet. The proposed error spreading method spreads out the not utilize packet losses widely, and can be used for videocasting and video-broadcasting that do not utilize the ARQ method. The proposed method can not only reduce the consecutive losses of image blocks, but also improve the performance of error concealment. Experimental results show that the deterioration of MPEG video quality can be reduced by spreading out the MPEG packet losses.

• Proceedings of the KSPS conference
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• 2007.05a
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• pp.307-310
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• 2007

In this paper, a packet loss concealment (PLC) algorithm for CELP-type speech coders is proposed to improve the quality of decoded speech under a burst packet loss condition. The proposed algorithm is based on the recovery of voiced excitation using an estimate of the voicing probability and the generation of random excitation by permutating the previously decoded excitation. The voicing probability is estimated from the correlation using the previous correctly decoded excitation and pitch. The proposed algorithm is implemented as a PLC algorithm for G.729 and its performance is compared with PLC employed in G.729 by means of perceptual evaluation of speech quality (PESQ) and an A-B preference test under the random and burst packet losses with rates of 3% and 5%. It is shown that the proposed algorithm provides better speech quality than the PLC of G.729, especially under burst pack losses.

• The KIPS Transactions:PartC
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• v.10C no.7
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• pp.937-942
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• 2003

In this paper, we measured and examined RTT delays and packet losses according to the changes of stationary loads for two typical stream-type traffics, a DV and a MPGE2 on the R＆D Gigabit Network testbed, JGN. As the result of our actual measurements, we realized that the packet size of stationary load have no effects on a DV and a MPGE2 stream on the very high-speed network(50Mbps, IP over ATM). When its bandwidth and stationary load exceeds 95％ of network bandwidth, packet losses appeared and RTT delay increased rapidly. Also we realized that the number and size of Receive ＆ Transmit buffer on the end systems have no effects on packet losses and RTT delays.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.42 no.2 s.302
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• pp.1-10
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• 2005

The current implementation of TCP for the Internet is not efficient when used for Mobile Ad hoc Networks (MANENTs). This is because TCP assumes that all packet losses are caused by congestion, whereas transmission errors are a main reason for packet losses in wireless networks. To remedy this situation and increase performance, we propose an end-to-end method of using propagation delays and the differences between propagation delays to distinguish the causes for packet losses. The proposed method has two characteristics: Firstly, it is energy-efficient because this solution is only initiated when a packet loss is detected. Secondly, our approach considers only the one way propagation delay and is more accurate in determining causes for packet losses than existing methods which consider round trip time. Petri net models of the proposed TCP and of the standard TCP have been built and simulations have been performed on them. Our simulation results show that the proposed approach increases throughput and reduces propagation delay compared with standard TCP.

• The KIPS Transactions:PartC
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• v.9C no.4
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• pp.473-480
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• 2002

Although many researches on IDS (Intrusion Detection System) have been performed, the most of them are limited to the algorithm of detection software. However, even an IDS with superior algorithm can not detect intrusion, if it loses packets which nay have a clue of intrusions. In this paper, we suggest an efficient wav to improve the performance of IDS by reducing packet losses occurred due to hardware limitation and abundant processing overhead introduced by massive detection software itself. The reduction in packet losses is achieved by dropping hacking-free packets. The result shows that this decrease of packet losses leads an IDS to improve the detection rate of real attack.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.7B
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• pp.477-487
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• 2008

Congestion control of the TCP reduces transmission rate when it detects packet loss because packet loss origines from congestion in the wired network. In the wireless network, packet loss comes from channel errors. Wired TCP degrades performance when there are wireless losses because it does not classify type of loss. These day, there are many researches which classify type of loss between congestion loss and wireless loss for wired-wireless hybrid network. For wireless TCP, many of existing algorithms are based on the estimated bandwidth or variations of packet arrival time. In this paper, we propose a new TCP scheme to distinguish the wireless packet losses from the congestion packet losses using MIB of the IEEE 802.11 MAC. We perform excessive simulations using the NS-2 network simulator and analyze the simulation results to compare the performance of the proposed algorithm to other well-known algorithms. From simulation results, we know that proposed algorithm improves performance about 12% and 32% compared with Spike algorithm and mBiaz algorithm, respectively.