• Journal of the Korea Society of Computer and Information
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• v.4 no.4
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• pp.148-154
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• 1999

TCP has improved by many papers which suggest the new algorithms and modify the previous algorithms. This paper compares Tahoe, Reno, New-Reno, Vegas, and SACK. The first version is Tahoe and is globally used. Reno has optimal performance during occurring one packet loss within a window of data, but can suffer from performance when multiple packets are dropped from a window of data. New-Reno avoids some of the performance problems of Reno TCP when multiple packets are dropped from a window of data. but is occurring the problem of the necessary retransmission. SACK resolves the all above problems and is used in bandwidth delay product environment. Vegas uses network bandwidth more efficiently and is a new implementation of TCP that achieves between 40 and 70 better throughput, with one-fifth to one-half the losses, as compared to the implementation of Reno TCP.

• Proceedings of the IEEK Conference
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• 2000.07a
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• pp.101-104
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• 2000

Nowadays, the issue of congestion control in high-speed communication networks becomes critical in view of the bandwidth-delay products for efficient data flow. In particular, the fact that the congestion is often accompanied by the data flow from the high-speed link to low-speed link is important with respect to the stability of closed-loop congestion control. The Virtual-Connection Network (VCN) in Gigabit Ethernet networks is a packet-switching based network capable of implementing cell- based connection, link-by-link flow-controlled connection, and single- or multi-destination virtual connections. VCN described herein differ from the virtual channel in ATM literature in that VCN have link-by-link flow control and can be of multi-destination. VCNs support both connection-oriented and connectionless data link layer traffic. Therefore, the worst collision scenario in Ethernet CSMA/CD with virtual collision brings about end-to-end delay. Gigabit Ethernet networks based on CSMA/CD results in non-deterministic behavior because its media access rules are based on random probability. Hence, it is difficult to obtain any sound mathematical formulation for congestion control without employing random processes or fluid-flow models. In this paper, an analytical method for the design of a congestion control scheme is proposed based on Smith＇s principle to overcome instability accompanied with the increase of end-to-end delays as well as to avoid cell losses. To this end, mathematical analysis is provided such that the proposed control scheme guarantees the performance improvement with respect to bandwidth and latency for selected network links with different propagation delays. In addition, guaranteed bandwidth is to be implemented by allowing individual stations to burst several frames at a time without intervening round-trip idle time.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2017.05a
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• pp.77-80
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• 2017

One of the fundamental limits of Bluetooth Low Energy(BLE) is that the data transmission is available via singlehop connection. In this research, we suggested the stable multihop transmission method to overcome this limitation. In multihop connection situation, multiple singlehop connection should be made and disconnected dynamically. Therefore, we stored the data within the GATT layer and tried to send it dynamically. We divided whole process as 4 states, and let each nodes transfers around each states to make data connection safely. Also, we set the transfer policy between each states during the transmission to make a robust system. From the experiment in real-time environment, we proved that our method showed high rate of packet delivery in a multihop network, which consists of more than 3 nodes.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.2A
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• pp.86-95
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• 2003

With the advent of future mobile communication systems, the wireless transmission of the huge amount of multimedia data over the error-prone multipath fading channel has to overcome the inherent sensitivity to channel errors. To alleviate the effect of the channel errors, hosts of techniques based on the forward error correction(FEC) has been proposed at the cost of overhead rate. Among the FEC techniques, turbo code, whose performance has been shown to be very close to the Shannon limit, can be classified as a block-based error correction code. In this paper, considering the variable packet size of the multimedia data, we analyzed turbo codes employing the variable-sized interleaver. The effect of the various parameters on the BER performance is analyzed. We show that the turbo codes can be used as efficient error correction codes of multimedia data.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.7
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• pp.9-19
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• 2004

WLAN (Wireless Local Area Networks) needs error recovery and flow control schemes to support reliable multicast protocol. Limited wireless bandwidth, as well as queuing losses caused by the asymmetric wired/wireless interactions, demands more effective approaches for reducing packet losses. Moreover, since the wireless channel is a shared broadcast medium, if sender receives feedback information simultaneously from several receivers, the feedback delays data frame transmission of forward direction by introducing channel congestion and burden at the sender. Therefore, it is important to minimize the amount of feedback information from receivers. In this paper, we propose an ARS(Adaptive Redundancy Scheme) that combines FEC(Forward Error Correction) using channel state estimation and ARQ(Automatic Repeat Request) both to reduce the amount of feedback information and the number of retransmissions and to guarantee high data reliability in a WLAN multicast environment. Performance of the proposed scheme is evaluated by means of analysis and simulations in AWGN and Rayleigh fading channels. The results show that the proposed scheme reduces the amount of feedback information and the number of retransmissions and guarantees high data reliability, while keeping throughput efficiency similarly with the conventional FEC and ARQ scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.7B
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• pp.745-759
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• 2011

In this paper, we propose the global handover mechanism that is able to provide the unlimited range of next-generation multimedia mobile services in an integrated environment. This mechanism consists of a multicast source based forwarding scheme and a global session management scheme. Global session management scheme provides LMA session information management, global mobility and route optimization. Multicast source based forwarding scheme delivers data between previously attached LMA and newly attached LMA without packet loss. In addition, this scheme removes the redundancy of buffered data. We present a performance evaluation and features analysis by the simulations using the ns-2. Global session management scheme has a less handover latency, propagation delay and signaling cost than the conventional methods. Multicast source based forwarding scheme can deliver buffer data without loss and it has less buffer size than conventional method.

• Journal of Broadcast Engineering
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• v.6 no.3
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• pp.205-214
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• 2001

In this Paper, we present an MP4 file streaming system over IP networks. Using the proposed system a user can access MP4 contents in the servers via H) networks and interact with the contents. The MP4 file format is designed to contain the media information of MPEG-4 and is object-oriented. The presented streaming server system contains GUI, session manager, splitter, SL- packetizer and transmitter. In addition, we knave implemented the client system based on the GUI -2D player, the MPEG-4 reference software. The Presented streaming system is designed use RTF for the media data requiring real-time streaming RTCP for QoS management and TCP for the data such as IOD(Initial Object Descriptor), OD(Object Descriptor) BIFS(Binary Format for Scene), which should be transmitted for the streaming and the data, such as still image and text, which can be entirely transmitted in a packet.

• Journal of Satellite, Information and Communications
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• v.1 no.1
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• pp.12-19
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• 2006

In this paper, we propose the bandwidth borrowing scheme which improves the performance of the cal admission control of the integrated GEO satellite networks. In general, target transmission rates of communications and streaming services are fixed, but data services do not have the target transmission rates. Therefore, we can control the transmission rates for data services flexibly according to the system loading situation. When the available bandwidth of the system is insufficient, the bandwidth borrowing scheme gives the bandwidth to request real time services by the transmission rates control of data services through packet scheduler. We make the queueing model for our system model and demonstrate the results through simulations. The simulation results show that there is a 8.7-35.2 dB gain at the total blocking probability according to the use of bandwidth borrowing scheme.

• The KIPS Transactions:PartA
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• v.15A no.6
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• pp.317-324
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• 2008

In this paper, we propose a system which easily controls the existing serially connected multi-channel speakers in a general personal computer by using a USB(Universal Serial Bus) interface. The personal computer as a host of the USB interface analyzes a sound source and sends audio data in a real-time fashion by the use of the isochronous transmission, one of four transmission methods provided by the USB interface. In addition, a channel is assigned by means of the bulk transmission, one of four transmission methods provided by the USB interface. Transmitted data from the USB host are sent to each speaker through compression and packet generation process. Each speaker detects corresponding digital data and regenerates audio signals through DAC(Digital-to-Analog Converter). A user can easily select a sound source file and a channel by the use of a GUI environment in a personal computer.

• Smart Media Journal
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• v.8 no.2
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• pp.21-28
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• 2019

Recent research on energy harvesting wireless sensor networks focuses on the development of techniques to solve the limited energy resource problem and to extend the whole network life efficiently. Energy harvesting technology can increase the lifetime of a network, but data transmission becomes unavailable when it harvests energy from radio frequency, resulting longer network delay with respect to the increased time in energy harvesting. Therefore, building energy harvesting wireless sensor network should consider the possible network delay as well as the network lifetime problem. In this paper, we propose a new MAC protocol that minimizes end-to-end network delay by adjusting the data transmission time for a packet based on estimating the energy for data transmission along with the amount of traffic flowing into the network and harvested energy. For this goal, it engineers an energy management mechanism that adjusts the sleep time of the network by measuring energy harvesting time. In addition, with simulation results it shows that the proposed MAC protocol improves the performance in terms of energy consumption and end-to-end delay, compared to the existing MAC protocols.