• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1997.06a
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• pp.49-54
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• 1997

For the purpose of a flexible coded video transmission over a heterogeneous network, we propose a new packetization method for coded video data. The proposed method achieves small degradation of coded picture quality in case of packet discard at the network node and does not require heavy processing load for bitrate control operation. Computer simulation results show that the bitrate reduction from 384 kb/s to 192 kb/s does not cause severe degradation in picture quality.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.5B
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• pp.871-878
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• 2000

A divided matched filter-reference filter(MF-RF) technique for LEO satellite packet transmission is proposed to increase the packet throughput in the presence of severe Doppler shift and fading. To overcome the severe Doppler shift, the divided matched filter is adopted where the integration region of matched filter is divided and ouputs of divided matched filer are added to decide the correct pseudo-noise (PN) phase. To maintain the constant false alarm rate in time varying interference and fading channel, the adaptive threshold for acquisition is obtained from the reference filter. As a performance measure, average acquisition time and packet throughput are used, and the effets of the parameters, i.e., Doppler shift, chip energy to noise ratio, user velocity, standard deviation of shadowing, and preamble length are shown.

• The KIPS Transactions:PartC
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• v.10C no.5
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• pp.641-646
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• 2003

Packet processing speed of routers as well as transmission speed of physical links gives a great effect on IP packet transfer rate in Internet. The router forwards a packet after determining the next hop to the packet's destination. IP address lookup is a main design issue for high performance routers. In this paper, we propose a partitioned compressed-trie for speeding-up IP address lookup algorithms based on tie data structure by exploiting path compression. In the ,proposed scheme, IP prefixes are divided into several compressed-tries and lookup is performed on only one partitioned compressed-trie. Memory access time for IP address lookup is lessen due to compression technique and memory required for maintaining partition does not increased.

• 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.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.4 no.3
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• pp.341-357
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• 2010

Forward Error Correction (FEC) techniques have been adopted to overcome packet losses and to improve the quality of video delivery. The efficiency of the FEC has been significantly compromised, however, due to the characteristics of the wireless channel such as burst packet loss, channel fluctuation and lack of Quality of Service (QoS) support. We propose herein an Adaptive Cross-layer FEC mechanism (ACFEC) to enhance the quality of video streaming over 802.11 WLANs. Under the conventional approaches, FEC functions are implemented on the application layer, and required feedback information to calculate redundancy rates. Our proposed ACFEC mechanism, however, leverages the functionalities of different network layers. The Automatic Repeat reQuest (ARQ) function on the Media Access Control (MAC) layer can detect packet losses. Through cooperation with the User Datagram Protocol (UDP), the redundancy rates are adaptively controlled based on the packet loss information. The experiment results demonstrate that the ACFEC mechanism is able to adaptively adjust and control the redundancy rates and, thereby, to overcome both of temporary and persistent channel fluctuations. Consequently, the proposed mechanism, under various network conditions, performs better in recovery than the conventional methods, while generating a much less volume of redundant traffic.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.14 no.5
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• pp.453-461
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• 1989

Mean waiting time for each priority packet of each node in packet based priority token ring local area networks(LAN) was approximately analyzed using Bux's token ring LAN results which have not considered priority and Cohbam's head of line(HOL) priority results. In this paper, priority reservation method suggested in the IEEE 802.5 standard was not used. Relative error between numerical results which was presented in this paper and simulation results was identified by +-5%. For traffic intenity, number of node, packet length, transmission speed, line length, token latency, number of priority class and traffic percentage to some heavy trafficd node, mean waiting time of each priority was analyzed.

• The Journal of the Korea Contents Association
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• v.8 no.4
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• pp.79-85
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• 2008

These days, orthogonal frequency division multiplexing (OFDM) transmission method is widely used for broadband communication systems. The OFDM, which uses sine waves as orthogonal basis functions, is one of the orthogonal waveform modulation techniques. In this paper, we investigate a wavelet packet modulation method which uses wavelet packets instead of sine waves as the basis functions. The wavelet packets may have different patterns in two dimensional time-frequency domain, and we can design the packets appropriate for the channel environments with much flexibility. In this paper, we investigate the characteristics of the wavelet packet modulation as one of the multicarrier modulation methods, And we illustrate by simulations that narrowband interference can be reduced effectively by control the bandwidth of the wavelet packets.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.4
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• pp.659-665
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• 2015

Recent studies on network coding scheme for increasing transmission efficiency of the network has been actively conducted. In this paper, we apply RLNC in sub-packet unit and propose a joint scheme of sub-packet level network coding and LDPC code. The proposed method can have similar ability of network coding and obtain further error correction capability. The simulation results show that the proposed one enhances error correction capability compared to the case using only LDPC when extra packets are received.

• The KIPS Transactions:PartC
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• v.13C no.3 s.106
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• pp.331-338
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• 2006

Most fair packet schedulers supporting quality-of-services of real-time multimedia applications are based on the finish time design scheme in which the expected transmission finish time of each packet is used as its timestamp. This scheme can adjust the latency of a flow with raising the flow's scheduling rate but it may suffer from severe bandwidth loss due to the coupled rate and delay allocation. This paper first introduces the concept of delay resource, and then proposes a scheduling method to improve the bandwidth utilization in which delay resource being lost due to the coupled allocation is transformed into bandwidth one. The performance evaluation shows that the proposed method gives higher bandwidth utilization by up to 50%.