• The KIPS Transactions:PartC
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• v.9C no.3
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• pp.375-384
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• 2002

In the current Internet, the QoS of interactive applications is hardly guaranteed because of variable bandwidth, packet loss and delay. Moreover, VoIP which is becoming an important part of the information infra-structure in these days, is susceptible to network packet loss and end-to-end delay. Therefore, it needs error control mechanisms in network level or application level. The FEC-based error control mechanisms are used for interactive audio application such as VoIP. The FEC sends a main information along with redundant information to recover the lost packets and adjusts redundant information depending on network conditions to reduce the bandwidth overhead. However, because most of the error control mechanisms do not consider end-to-end delay but packet loss rate, their performances are poor. In this paper, we propose a new error control algorithm, SCCRP, considering packet loss rate as well as end-to-end delay. Through experiments, we confirm that the SCCRP has a lower packet loss rate and a lower end-to-end delay after reconstruction.

• The Transactions of the Korea Information Processing Society
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• v.2 no.6
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• pp.927-936
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• 1995

ATM Forum has defined a new service class for data applications called Available Bit Rate(ABR) Service, which has highly bursty traffic and unpredictable burst size. It is desirable that we reduce the probability of retrans mission of packets by minimizing the loss of cells because the traffic is much more sensitive to loss than delay. The Forum has also selected the Rate-Based Control for the ABR service and proposed EPRCA as the control mechanism for the service. This paper proposes the combination of EPRCA and the other feedb ack control mechanisms such as BECN and BP. The combined control mechanism control ABR traffic more efficiently and the simulation results show that the network performance can be improved by choosing the appropriate parameters.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.66 no.3
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• pp.575-580
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• 2017

In these days, the big-size and high resolution multimedia file is widely used through networks. To transfer and service effectively, the internet network technology is necessary to substitute broadcasting. Normally Content Delivery Network(CDN) is widely used in conventional internet for multimedia services. But it has a small bandwidth to service. So to solve this problems, many researchers have suggest the protocol for download, content distribution/saving, server synchronization, caching, pushing rate, and streaming etc. But all of these has some defects like low resolution, packets loss and delay, real application implementations etc. So, this paper suggests a new method of network protocol based on P2P collaboration and user's content using information. And it evaluated the performance of suggested method. As the results, it showed the effectiveness of 4 performances indices : download speed, decreasing rate of connected user in same time, adaptive hit ratio, traffic decreasing rate.

• Proceedings of the IEEK Conference
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• 2002.07b
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• pp.1094-1097
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• 2002

Realtime applications like VoIP(Voice over IP) in All-IP networks need smooth handoffs in order to minimize or eliminate packet loss as a Mobile Host(MH) transitions between network links. In this paper, we design a new dynamic buffering(DB) mechanism for IPv6 by which an MH can request that the router on its current subnet buffers packets on its behalf while the MH completes registration procedures with the router of a new subnet. Performance results show that our proposed buffering scheme with a dynamic buffer space allocation is quite appropriate for mobile Internet, or the All-IP environment in terms of the datagram loss rate.

• 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.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.11
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• pp.63-71
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• 2003

This paper suggests the packet loss recovery method to communicate in real time in the Internet. To reduce the effects of packet loss, Forward Error Correction (FEC) that adds redundant information to voice packets can be used. Adaptive Multi Rate Wideband(AMR-WB) codec which is recently selected by the Third Generation Partnership Project(3GPP) for GSM and the third generation mobile communication WCDMA system and has also been standardized in ITU-T for providing wideband speech services is used. The major cause for speech qualitly degradation in IP-networks is packet loss. So, We recovered single lossy packet by using FEC method and concealed continued errors. The proposed scheme if evaluated in the Gilbert Internet channel model. The high quality of audio maintained up to 30% packet loss.

• Journal of Computing Science and Engineering
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• v.5 no.4
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• pp.316-323
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• 2011

Recently, Proxy Mobile IPv6 (PMIPv6) has received much attention as a mobility management protocol in next-generation all-IP mobile networks. While the current research related to PMIPv6 mainly focuses on providing efficient handovers for unicast-based applications, there has been relatively little interest in supporting multicast services with PMIPv6. To provide support for multicast services with PMIPv6, there are two alternative approaches called Mobile Access Gateway (MAG)-based subscription and Local Mobility Anchor (LMA)-based subscription. However, MAG-based subscription causes a large overhead for multicast joining and LMA-based subscription provides non-optimal multicast routing paths. The two approaches may also cause a high packet loss rate. In this paper, we propose an efficient PMIPv6-based multicast protocol that aims to provide an optimal delivery path for multicast data and to reduce handover delay and packet loss rate. Through simulation studies, we found that the proposed protocol outperforms existing multicast solutions for PMIPv6 in terms of end-to-end delay, service disruption period, and the number of lost packets during handovers.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.9B
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• pp.783-792
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• 2003

We consider a finite-buffer discrete-time queueing system with fixed-size bulk-service discipline: Geo/ $G^{B}$1/K＋B. The main purpose of this paper is to present a performance analysis of this system that has a wide range of applications in Asynchronous Transfer Mode (ATM) and other related telecommunication systems. For this purpose, we first derive the departure-epoch probabilities based on the embedded Markov chain method. Next, based on simple rate in and rate out argument, we present stable relationships for the steady-state probabilities of the queue length at different epochs: departure, random, and arrival. Finally, based on these relationships, we present various useful performance measures of interest such as the moments of number of packets in the system at three different epochs and the loss probability. The numerical results are presented for a deterministic service-time distribution - a case that has gained importance in recent years.s.

• Journal of Communications and Networks
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• v.11 no.5
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• pp.464-472
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• 2009

In this paper, we propose a packet scheduling algorithm for cellular relay networks by considering relay selection, variation of channel quality, and packet delay. In the networks, mobile users are equipped with not only cellular but also user relaying radio interfaces, where base station exploits adaptive high speed downlink channel. Our proposed algorithm selects a user with good cellular channel condition as a relay station for other users with bad cellular channel condition but can get access to relay link with good quality. This can achieve flexible packet scheduling by adjusting transmission rates of cellular link. Packets are scheduled for transmission depending on scheduling indexes which are calculated based on user's achieved transmission rate, packet utility, and proportional fairness of their throughput. The performance results obtained by using computer simulation show that the proposed scheduling algorithm is able to achieve high network capacity, low packet loss, and good fairness in terms of received throughput of mobile users.

• Nuclear Engineering and Technology
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• v.54 no.10
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• pp.3587-3594
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• 2022

A distributed high-speed data acquisition and monitoring system for the RF negative ion source at Huazhong University of Science and Technology (HUST) is developed, which consists of data acquisition, data forwarding and data processing. Firstly, the data acquisition modules sample physical signals at high speed and upload the sampling data with corresponding absolute-time labels over UDP, which builds the time correlation among different signals. And a special data packet format is proposed for the data upload, which is convenient for packing or parsing a fixed-length packet, especially when the span of the time labels in a packet crosses an absolute second. The data forwarding modules then receive the UDP messages and distribute their data packets to the real-time display module and the data storage modules by PUB/SUB-pattern message queue of ZeroMQ. As for the data storage, a scheme combining the file server and MySQL database is adopted to increase the storage rate and facilitate the data query. The test results show that the loss rate of the data packets is within the range of 0-5% and the storage rate is higher than 20 Mbps, both acceptable for the HUST RF negative ion source.