• Journal of information and communication convergence engineering
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• v.9 no.5
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• pp.551-555
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• 2011

Optical packet switching is a promising technology, which can integrate both data and optical network. In this paper, we present a comparative study of various traffic arrival approaches in WDM packet switches. The comparison is made based on packet loss rate and average delay under uniform and self-similar Pareto traffic. Computer simulations are performed in order to obtain the switch performance metrics. Study shows that burstiness of data traffic has a strong negative impact in the performance of WDM packet switches.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.26 no.6
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• pp.1571-1582
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• 2016

One of the issues, which is dealed with in undirectional data transmission technology, is reducing the packet loss in TCP based data transfer. We can decrease the packet loss by using several well known error correction approaches. Although we utilize those previous approaches, the packet loss by both link errror and buffer overflow could be occurred. In this paper, we propose the RED(REliable Data diode). RED also uses the TCP proxy approach for supporting the TCP based data transfer which is similar with the existing unidirectional data transmission technologies. The RED transmission system could alleviate the packet loss caused by buffer overflow by exploiting the delaying transmission of TCP packets. Furthermore, in order to reduce the packett loss caused by link error in the unidirectional transmission link, the RED transmission system transmits one or more duplicated packets to the RED reception system by considering both the remaining resources and packet importance.

• The Journal of the Acoustical Society of Korea
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• v.29 no.1
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• pp.34-40
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• 2010

As internet spread widely, various service which use the internet have been provided. One of the service is a internet phone. Its usage is increasing by the advantage of cost. But it has a falling off in quality of speech. because it use packet switching method while existing telephone use circuit switching method. Although vocoder use PLC (Packet Loss Concealment) algorithm, it has a weakness of continuous packet loss. In this paper, we propose methods to improve a lowering in quality of speech under continuous loss of packet by using PLC algorithm used in advanced G.729 and G.711. The proposed methods are LP (Linear Prediction) parameter interpolation, excitation signal reconstruction and excitation signal gain reconstruction. As a result, the proposed method shows superior performance about 11%.

• IEMEK Journal of Embedded Systems and Applications
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• v.1 no.1
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• pp.24-30
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• 2006

With the original Transmission Control Protocol(TCP) design, which is particularly targeted at the wired networks, a packet loss is assumed to be caused by the network congestion. In the wireless environment where the chances to lose packets due to transmission bit errors are not negligible, though, this assumption may result in unnecessary TCP performance degradation. In these days, many papers describe about wireless-TCP which has suggested how to avoid congestion control when packet loss over the wireless network. In this paper, an enhancement scheme is proposed by modifying SNOOP scheme. To enhance the original SNOOP scheme, CPC(Consecutive Packet Control) and ZWSC(Zero Window Size Control) are added. The invocation of congestion control mechanism is now minimized by knowing the cause of packet loss. We use simulation to compare the overhead and the performance of the proposed schemes, and to show that the proposed schemes improve the TCP performance compares to SNOOP by knowing the cause of packet loss at the base station.

• 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.34 no.11B
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• pp.1225-1233
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• 2009

ISPs and Enterprises are equipping their networks with sufficiently high speed facilities and provide large bandwidths members. However the high speed enterprise network does not have satisfying end-to-end network performance within the network in spite of under utilization. The root cause of this performance degradation is a micro-congestion, which is a short-live event of traffic congestion. A micro-congestion causes packet loss, delay and packet reodering, and finally results in end-to-end network performance degradation. In this paper, we propose a micro-congestion detection method and find out the characteristics of performance degradation by analyzing traffic archives which is collected from a network link when a micro-congestion occurs.

• Journal of KIISE:Computer Systems and Theory
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• v.33 no.10
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• pp.722-733
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• 2006

In scalable streaming application, there are two important knobs to tune to effectively exploit the underlying network resource and to maximize the user perceivable quality of service(QoS): layer selection and packet scheduling. In this work, we propose Semantics Aware Packet Scheduling (SAPS) algorithm to address these issues. Using packet dependency graph, SAPS algorithm selects a layer to maximize QoS. We aim at minimizing distortion in selecting layers. In inter-frame coded video streaming, minimizing packet loss does not imply maximizing QoS. In determining the packet transmission schedule, we exploit the fact that significance of each packet loss is different dependent upon its frame type and the position within group of picture(GOP). In SAPS algorithm, each packet is assigned a weight called QoS Impact Factor Transmission schedule is derived based upon weighted smoothing. In simulation experiment, we observed that QOS actually improves when packet loss becomes worse. The simulation results show that the SAPS not only maximizes user perceivable QoS but also minimizes resource requirements.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.3
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• pp.1-7
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• 2012

Cognitive radio, a new paradigm for wireless communication, is being recently expected to support various types of multimedia traffics. To guarantee Quality of Service (QoS) from SUs, a static packet priority policy can be considered. However, this approach can easily satisfy Quality of Service of high priority application while that of lower priority applications is being degraded. In the paper, we propose a fuzzy-based dynamic packet scheduling algorithm to support multimedia traffics in which the dynamic packet scheduler modifies priorities of packets according to Fuzzy-rules with the information of priority and delay deadline of each packet, and determines which packet would be transmitted through the channel of the primary user in the next time slot in order to reduce packet loss rate. Our simulation result shows that packet loss rate can be improved through the proposed scheme when overall traffic load is not heavy.

• Journal of Korea Society of Digital Industry and Information Management
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• v.5 no.2
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• pp.141-156
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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. The Snoop Protocol was designed for the wired network by putting the Snoop agent module on the BS(Base Station) that connect the wire network to the wireless network to complement the TCP problem. The Snoop agent cash the packets being transferred to the wireless terminal and recover the loss by resending locally for the error occurred in the wireless link. The Snoop agent blocks the unnecessary congestion control by preventing the dupack (duplicate acknowledgement)for the retransmitted packet from sending to the sender and hiding the loss in the wireless link from the sender. We evaluated the performance in the wired/wireless network and in various TCP versions using the TCP designed for the wired network and the Snoop designed for the wireless network and evaluated the performance of the wired/wireless hybrid network in the wireless link environment that the continuous packet loss occur.

• Journal of information and communication convergence engineering
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• v.14 no.4
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• pp.240-245
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• 2016

Scalable video streaming over wireless networks has many challenges. The most significant challenge is related to packet loss. To overcome this problem, in this paper, we propose an unequal loss protection (ULP) method using a new forward error correction (FEC) mechanism for robust scalable video streaming over wireless networks. For an efficient FEC assignment considering video quality, we first introduce a simple and efficient performance metric, the layer-based recovery rate (LRR), for quantifying the unequal error propagation effects of the temporal and quality layers on the basis of packet losses. LRR is based on the unequal importance in both the temporal and the quality layers of a hierarchical scalable video coding structure. Then, the proposed ULP-LRR method assigns an appropriate number of FEC packets on the basis of the LRR to protect the video layers against packet lossy network environments. Compared with conventional ULP algorithms, the proposed ULP-LRR algorithm demonstrates a higher performance for various error-prone wireless channel statuses.