• Journal of Broadcast Engineering
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• v.16 no.3
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• pp.530-541
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• 2011

In this paper, we propose an effective network-adaptive ARQ based error control scheme to provide video streaming services through IP networks where packet error usually occurs. If time delay and feedback channel are allowed, client can request server to retransmit lost packets through IP networks. However, if retransmission is unconditionally requested without considering network condition and number of simultaneous feedback messages, retransmitted packets may not arrive in a timely manner so that decoding may not occur. In the proposed ARQ, a client conditionally requests retransmission based on assumed network condition, and it further determines valid retransmission time so that effective ARQ can be applied. In order to verify the performance of the proposed adaptive ARQ based error control, NIST-Net is used to emulate packet-loss network environment. It is shown by simulations that the proposed scheme provides noticeable error resilience with significantly reduced traffics required for ARQ.

• Journal of KIISE:Information Networking
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• v.34 no.6
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• pp.435-445
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• 2007

Snoop protocol is one of the efficient schemes to compensate TCP packet loss and enhance TCP throughput in wired-cum-wireless networks. However, Snoop protocol has a problem: it cannot perform local retransmission efficiently under the bursty-error prone wireless link. To solve this problem, SACK-Aware-Snoop and SNACK mechanism have been proposed. These approaches improve the performance by using SACK option field between base station and mobile host. However in the wireless channel with high packet loss rate, SACK-Aware-Snoop and SNACK mechanism do not work well because of two reason: (a) end-to-end performance is degraded because duplicate ACKs themself can be lost in the presence of bursty error, (b) energy of mobile device and bandwidth utilization in the wireless link are wasted unnecessarily because of SACK option field in the wireless link. In this paper, we propose a new local retransmission scheme based on Cross-layer approach, called Cross-layer Snoop(C-Snoop) protocol, to solve the limitation of previous localized link layer schemes. C-Snoop protocol includes caching lost TCP data and performing local retransmission based on a few policies dealing with MAC-layer's timeout and local retransmission timeout. From the simulation result, we could see more improved TCP throughput and energy efficiency than previous mechanisms.

• Proceedings of the Korean Information Science Society Conference
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• 2001.10c
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• pp.307-309
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• 2001

In this paper, we present a scheme for the improving reliable transport and the efficient multicast support in mobile environment. The proposed scheme solves a problem of TCP layer resulted from mobility by using a Representative FA. RFA has a mechanism like a snoop module which has a cache and can provide retransmission of a multicast packet lost and solve the rock implosion problem. Also, we present an additional IGMP message. By using it, We can remove a delay for IGMP query cycle and serve a multicast service more promptly.

• 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.30 no.4
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• pp.513-519
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• 2003

In order to deliver multicast traffic efficiently, multicast routing and reliable transmission mechanisms are required. The reliable delivery implies that lost packets must be retransmitted, which in turn requires that transmitted packets be stored in a retransmission buffer. Therefore how to manage a retransmission buffer is important and, in this paper, we try to solve the problem of how many packets should be maintained in the buffer. Our proposed scheme, the timer-based buffer management (TBM), maintains only necessary amount of buffer based on the timer value calculated from the NAKs between the replier and receivers on a multicast tree and can adjust to the dynamic network conditions. By performing simulations, we show that TBM manages the buffer efficiently regardless of the error situation, network size, and so on.

• 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 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 Korea Multimedia Society
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• v.7 no.1
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• pp.54-63
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• 2004

This paper describes a hybrid QoS guarantee scheme for high quality audio streaming services on the Internet. The continuous playback of the audio data requires the isochronous transmission of the audio data packet through the Internet. In order to retain the QoS at the ultimate destination (client) as the same as servers provide, the transmission protocols should consider the error conditions such as packet loss, and out of order delivery. Generally, the protocols supporting the transmission of continuous media data do not try to recover the errors. The protocols are working somehow for the toll quality multimedia streaming services, but rot for the high quality streaming services, such as the DVD sound/music payback. The hybrid QoS guarantee scheme includes the three mechanisms to overcome the problem. The selective retransmission for the lost packet, the adaptive buffering at client-side, and the adaptive transmission rate at server-side are totally adopted to recover the packet loss with the minimal overhead, to prevent from the buffer starvation during the retransmission, and to maintain the isochronous transmission even after the retransmission. The experiments have shown good results for the high Quality audio streaming services on the Internet.

• Journal of Internet Computing and Services
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• v.1 no.2
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• pp.1-10
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• 2000

ATM network technology is generally used for the solution of integrating multimedia service in high-speed Internet. In Internet protocol based on ATM services, If single cell is lost in ATM layer, the entire TCP packet will be lost. Therefore, TCP performance will be degraded. In order to reduce cell loss, when congestion occur, UBR+EPD mechanism is proposed to improve the throughput in TCP over UBR, and ER scheme is suggested in TCP over ABR. In this paper, we analyzed the performance improvement effect of UBR+EPD with FRR (Fast Retransmission and Recovery), the adjusting EPD threshold parameter (R), and variation of MTU (Maximum Transport Unit) size. As a result, through the analysis of performance, we know that the improved throughput and fairness are shown by the proposed scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.7B
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• pp.1322-1330
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• 1999

The major source of errors in high-speed networks such as Broadband ISDN(B-ISDN) is buffer overflow during congested conditions. These congestion errors are the dominant sources of errors in high-speed networks and result in cell losses. Conventional communication protocols use error detection and retransmission to deal with lost packets and transmission errors. As an alternative, we have presented a method to recover consecutive cell losses using forward error correction(FEC) in ATM(Asynchronous Transfer Mode) networks to reduce the problem. The method finds the lost cells by observing new cell sequence number($SN^{\ast}$). We have used the LI field together with SN and ST fields to consider the $SN^{\ast}$ which provides more correcting coverage than SN in ATM standards. The $SN^{\ast}$ based on the additive way such as the addition of LI capacity to original SN capacity is numbered a repeatedly 0-to-80 cycle. Another extension can be based on the multiplicative way such that LI capacity is multiplied by SN capacity. The multiplicative $SN^{\ast}$ is numbered in a repeatedly 0-to-1025 cycle.