• Journal of the Korea Institute of Information Security & Cryptology
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• v.23 no.5
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• pp.957-962
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• 2013

This paper proposes the new enhanced 2-D(2-Dimension) FEC scheme. It analyzes the probability of entire packet loss rate of the existing 2-D FEC by mathematical modeling, finds the problem of the existing 2-D FEC, and deduces the new enhanced 2-D FEC scheme that reduces the entire packet loss probability.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.8C
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• pp.810-823
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• 2005

The transmission of duplicate packets provides a great loss-resilience without undue time-delay in the video transmission over packet loss networks. But this method generally deteriorates the problem of traffic congestion because of the increased bit-rate required for duplicate transmission. In this paper, we propose an efficient packetization and duplicate transmission of video packets. The proposed method transmits only the video signal with high priority for each video macroblock that is quite small in volume but very important for the reconstruction of the video. The proposed method significantly reduces the required bit-rate for duplicate transmission. An efficient packetization method is also proposed to reduce additional packet overhead which is required for transmitting the duplicate data. The duplicated high priority data of the Previous video slice is transmitted as a Piggyback to the data Packet of the current video slice. It is shown by simulations that the proposed method remarkably improves the packet loss-resilience for video transmission only with small increase of redundant duplicated data for each slice.

• Proceedings of the Korea Society for Simulation Conference
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• 1999.10a
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• pp.127-132
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• 1999

A mobile IP is proposed to support host mobility over the current Internet. One of the most important issues on the host mobility is location and routing schemes that allow mobile hosts to move effectively from one site to another. In a Mobile IP environment, frequent handoffs are likely to degrade the performance by minimizing the loss of datagrams during handoffs. The handoff scheme is using routing optimization and binding extension to improve the performance by minimizing the average transfer delay of messages and packet loss. Simulation details show the improvement of transport delays and packet loss rate.

• Journal of Internet Computing and Services
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• v.8 no.5
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• pp.125-132
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• 2007

Transmission mechanisms that include an available bandwidth estimation algorithm and a packet loss differentiation scheme, in general, exhibit higher TCP performance in wireless networks. TCP New Jersey, known as the best existing scheme in terms of goodput, improves wireless TCP performance using the available bandwidth estimation at the sender and the congestion warning at intermediate routers. Although TCP New Jersey achieves 17% and 85% improvements in goodput over TCP Westwood and TCP Reno, respectively, we further improve TCP New Jersey by exploring improved available bandwidth estimation, retransmission timeout, and recovery mechanisms. Hence, we propose TCP New Jersey PLUS (shortly TCP NJ+), showing that under 1% packet loss rate, it outperforms 3% by TCP New Jersey and 5% by TCP Wes1wood. In 5% packet loss rate, a characteristic of high bit-error-rate wireless network, it outperforms other TCP variants by 19% to 104% in terms of goodput even when the network is in bi-directional congestion.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.9
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• pp.3159-3176
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• 2014

This work presents an analysis of the buffer features of an access router, especially the size, the impact on delay and the packet loss rate. In particular, we study how these features can affect the Quality of Service (QoS) of multimedia applications when generating traffic bursts in local networks. First, we show how in a typical SME (Small and Medium Enterprise) network in which several multimedia flows (VoIP, videoconferencing and video surveillance) share access, the upgrade of the bandwidth of the internal network may cause the appearance of a significant amount of packet loss caused by buffer overflow. Secondly, the study shows that the bursty nature of the traffic in some applications traffic (video surveillance) may impair their QoS and that of other services (VoIP and videoconferencing), especially when a certain number of bursts overlap. Various tests have been developed with the aim of characterizing the problems that may appear when network capacity is increased in these scenarios. In some cases, especially when applications generating bursty traffic are present, increasing the network speed may lead to a deterioration in the quality. It has been found that the cause of this quality degradation is buffer overflow, which depends on the bandwidth relationship between the access and the internal networks. Besides, it has been necessary to describe the packet loss distribution by means of a histogram since, although most of the communications present good QoS results, a few of them have worse outcomes. Finally, in order to complete the study we present the MOS results for VoIP calculated from the delay and packet loss rate.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2019.05a
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• pp.260-263
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• 2019

Recently, the need of researches and developments about WSN(Wireless Sensor Network) technologies, which can be applied to services to regions where the access is difficult or services that require continuous monitoring, has gradually increased due to its expansion and efficiency of the application areas. In this paper, we analyze existing researches which focused on reducing packet loss rate and increasing lifetime of sensor nodes. Then, we conduct studies about performance improvement factors where some schemes - clustering and location-based approaches - are applied and compare our study results with existing researches. Based on our studies, we are planning to conduct researches about a new scheme that could contribute to improve WSN's performance in terms of packet loss rate and network lifetime.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7C
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• pp.691-699
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• 2006

Despite of the in-depth effort to cantrol the variability in IP networks, quality of service (QoS) is still not guaranteed in the IP networks. Thus, it is necessary to deal with the audible artifacts caused by packet lasses. To overcame the packet loss problem, most speech coding standard have their own embedded packet loss concealment (PLC) algorithms which adapt extrapolation methods utilizing the dependency on adjacent frames. Since many low bit rate CELP coders use predictive schemes for increasing coding efficiency, however, error propagation occurs even if single packet is lost. In this paper, we propose an efficient PLC algorithm with consideration about the speech characteristics of lost frames. To design an efficient PLC algorithm, we perform several experiments on investigating the error propagation effect of lost frames of a predictive coder. And then, we summarize the impact of packet loss to the speech characteristics and analyze the importance of the encoded parameters depending on each speech classes. From the result of the experiments, we propose a new PLC algorithm that mainly focuses on reducing the error propagation time. Experimental results show that the performance is much higher than conventional extrapolation methods over various frame erasure rate (FER) conditions. Especially the difference is remarkable in high FER condition.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.687-695
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• 2002

Media-specific FEC techniques, suggested to confront with VoIP speech packet loss, improve speech quality at the expense of generating additional one-frame delay. In this paper, we suggest new media-specific FEC, i.e, LSF FEC technique which is able to improve speech quality with much shortened additional delay. In the proposed technique, the LSF parameters of the future frame are utilized to recover a lost packet. To evaluate performance of the proposed technique, we use ITU-T G.723.1 and G.729 Codec and apply Gilbert packet loss model and estimate MOS per every packet loss rate using PESQ speech quality estimation algorithm. The proposed technique has effect of shortening delay over from 6.5ms to 27ms compared with existing media-specific FEC techniques. Simulation results for comparison of reconstructed speech quality show this novel technique improves the MOS over 0.1 in practical lossy environment of 5 % packet loss rate.