• Proceedings of the KIEE Conference
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• 2000.07d
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• pp.3028-3030
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• 2000

In general. a speech coding is dedicated to the compression performance or the speech quality. But. the speech coding in this paper is focused on the performance of flexible transmission to the, network speed. For this. the subbanding coding is needed. which is used the wavelet packet concept in the signal analysis. The extraction of each frequency-band is difficult to general signal analysis methods, after coding each band, the reconstruction of these is also a difficult problem. But. with the wavelet packet concept(perfect reconstruction) and its fast computation algorithm. the extraction of each band and the reconstruction are more natural. Also, this paper describes a direct solution of the voice transmission on network and implement this algorithm at the TCP/IP network environment of PC.

• ETRI Journal
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• v.27 no.4
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• pp.469-472
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• 2005

To enhance the scalability of high performance packet switches, a two-stage load-balanced switch has recently been introduced, in which each stage uses a deterministic sequence of configurations. The switch is simple to make scalable and has been proven to provide 100% throughput. However, the load-balanced switch may mis-sequence the packets. In this paper, we propose an algorithm called full frame stuff (FFS), which maintains packet order in the two-stage load-balanced switch and has excellent switching performance. This algorithm is distributed and each port can operate independently.

• The Journal of the Korea Contents Association
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• v.3 no.2
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• pp.65-72
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• 2003

In this paper, the new scheme using a Combined Cache(CC) that combing the Paging and Routing Cache(PRC) and an alternative handoff method according to the type of data transmission for achieving the efficient realtime communication. The PRC and quasi-soft handoff method reduce the path duplication. But they increase the network traffic load because of the handoff state packet of Mobile Host(MH). Moreover the use the same handoff method without differentiating the type of transmission data. Those problems are solved by operating U with a semi-soft handoff method for realtime data transmission and with a hard handoff method for non-realtime data transmission. As a result or simulation a better performance is obtained because of the reduction of the number of control packet in case that the number of cells are below 20. And the packet arrival time and loss of packet decreased significantly for realtime data transmission.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.41 no.7
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• pp.71-79
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• 2004

Packet-type data transmission is characterized by the continuous transmission of massive data with relatively constant rate. In such transmission, the dynamic power consumed on buses is influenced by the sequence of transmitted data. A new coding scheme called Sequence-Switch Coding (SSC) is proposed in this paper. SSC reduces the number of bus transitions in the transmission of packet-type data by changing the sending order of the data. Some simple algorithms are presented, In. The simulation results show that SSC outperforms the well-known Bus-Invert Coding with these algorithms. SSC is not a specific algerian but a method to reduce the number of bus-transitions. There could be lots of algorithms for realizing SSC. The variety of SSC algorithms provides circuit designers a wide range of trade-off between performance and circuit complexity.

• Journal of Internet Computing and Services
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• v.20 no.2
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• pp.21-27
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• 2019

The terrestrial ATSC 3.0 broadcasting system, which is capable of converging broadcast and communication services, uses IP based technology for data transmission between broadcasting equipment. In addition, data transmission between broadcasting equipment uses IP-based technology like existing wired communication network, which has advantageous in terms of equipment construction and maintenance In case IP based data transmission technology is used, however, it may inevitably cause an error that a packet is lost during transmission depending on the network environments. In order to cope with a broadcasting accident caused by such a transmission error or a malfunction of a broadcasting apparatus, a broadcasting system is generally configured as a duplication, which can transmit a normal packet when various types of error may occur. By this reason, correction method of error packets and intelligent switching technology are essential. Therefore, in this paper, we propose a design and implementation of intelligent IP switch for Ensuring Reliability of ATSC 3.0 Broadcast Streams. The proposed intelligent IP consists of IP Stream Analysis Module, ALP Stream Analysis Module, STL Stream Analysis Module and SMPTE 2022-1 based FEC Encoding/Decoding Module.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.11
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• pp.5360-5376
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• 2017

In this paper, we study the cross-layer design for the AMC/HARQ-based wireless networks, and propose a new dynamic transmission-mode selection scheme to improve system spectrum efficiency. In the proposed scheme, dynamic thresholds for transmission-mode selection in each packet transmission and retransmission are jointly designed under the constraint of the overall packet error rate. Comparing with the existing schemes, the proposed scheme is inclined to apply higher modulation order at the first several (re)transmissions, which corresponds to higher-rate transmission modes thus higher average system spectrum efficiency. We also extend the cross-layer design to MIMO (Multi-input Multi-output) communication scenarios. Numerical results show that the proposed new dynamic transmission-mode selection scheme generally achieves higher average spectrum efficiency than the conventional and existing cross-layer design.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.3
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• pp.637-644
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• 2004

This paper proposed and implemented the adaptive transmission middleware for video streaming, which is able to support the adaptive transmission of video data to the fluctuating changes of network environment in the packet-based network and the properties of transmitted video data. The adaptive transmission middleware is made up SR-RTP-based transfer module and TFRC(TCP Friendly Rate Control)-based transfer-rate control module. The SR-RTP-based transfer module supports RTP-based real-time transfer of video data and packet retransmission scheme retransmitting the high-priority packets selectively in the damaged video data to reduce the error induced by the packet loss. Sharing the transmission bandwidth of network with the TCP-based data transfer, the TFRC-based transfer-rate control module controls the transfer rate of video data according to the most allowable transmission bandwidth in the network, so that the transfer rate is controlled adaptively to the fluctuating changes of transmission bandwidth. This paper, for the experiment, applied the adaptive transmission middleware to video streaming in the external Internet environment, and analyzed the effective frame transfer rate and the degree of the streaming jitter to evaluate the performance of packet-loss recovery and adaptive transfer rate control. In the external Internet environment where the packet-loss rate is high a bit, the relatively high streaming performance was showed compared with the case that didn't apply the adaptive transmission middleware.

• Publications of The Korean Astronomical Society
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• v.29 no.3
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• pp.45-52
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• 2014

This paper describes the development of algorithm for direct data transmission between Raw VLBI Data Buffer (RVDB) and Huge Capacity Data Server (HCDS) operated in Korea-Japan Correlation Center (KJCC). The transmitted data is the VLBI observation data, which is recorded at each radio telescope site, and the data transmitting rate is varying from 1 Gbps, in usual case, upto 8 Gbps. The developed algorithm for data transmission enables the direct data transmission between RVDB and HCDS through 10 Gbps optical network using VLBI Data Interchange Format (VDIF). Proposed method adopts the conventional UDP/IP protocol, but in order to prevent the loss of data during data transmission, the packet error monitoring and data re-transmission functions are newly designed. The VDIF specification and VDIFCP (VDIF Control Protocol) are used for the direct data transmission between RVDB and HCDS. To validate the developed algorithm for data transmission, we conducted the data transmission from RVDB to HCDS, and compared to the transmitted data with the original data bit by bit. We confirmed that the transmitted data is identical to the original data without any loss and it has been recovered well even if there were some packet losses.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.4
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• pp.581-587
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• 2021

Real-time video streaming scheme for multimedia content delivery and remote conference services is one of technologies that are significantly sensitive to data transmission delay. Recently, because of COVID-19, real-time video streaming contents for the services are significantly increased such as personal broadcasting and remote school class. In order to support the services, there is a growing emphasis on low transmission delay and secure content delivery, respectively. Therefore, our research proposed a packet aggregation algorithm to reduce the transmission delay of selectively encrypted video transmission for real-time video streaming services. Through the application of the proposed algorithm, the selectively encrypted video framework can control the amount of MPEG-2 TS packets for low latency transmission with a consideration of packet priorities. Evaluation results on testbed show that the application of the proposed algorithm to the video framework can reduce approximately 11% of the transmission delay for high and low resolution video.

• Journal of Multimedia Information System
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• v.3 no.4
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• pp.141-148
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• 2016

This paper proposes a medium access control (MAC) protocol for multiple-input multiple-output (MIMO) ad-hoc networks. Multiple-packet receptions in MIMO systems have attracted as a key technique to achieve a high transmission rate. In the conventional protocols for multiple-packet receptions, timing offsets among multiple-frame transmissions cause frame collisions induced by hidden nodes, which degrades network performance. In the proposed protocol, transmission synchronization among hidden nodes can be achieved by applying pulse/tone exchanges. By applying the pulse/tone exchanges, multiple-packet receptions among hidden nodes can be achieved, which enhances network throughputs compared with the conventional protocol. Simulation results show effectiveness of the proposed protocol.