• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2004.05b
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• pp.475-478
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• 2004

In this paper, a CSSMA/AI MAC protocol for data services in packet CDMA network is presented. The proposed protocol is based on the code status sensing and reservation scheme. In the proposed protocol, the base station broadcast the rode status on a frame-by-frame basis just before the beginning of each preamble transmission, and the mobile station transmits a preamble for reserving a randomly selected code based on the received code status. After having transmitted the preamble, the mobile station listens to the downlink of the selected rode and waits for the base station reply. If this reply indicates that the code has been correctly acquired, it continues the packet transmission lot the rest of the frame. If there are other packets waiting for transmission, the base station broadcasts the status of the code as reserved, and the mobile station transmits a packet through the reserved code for the successive frames.

• Journal of KIISE:Computer Systems and Theory
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• v.33 no.10
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• pp.722-733
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• 2006

In scalable streaming application, there are two important knobs to tune to effectively exploit the underlying network resource and to maximize the user perceivable quality of service(QoS): layer selection and packet scheduling. In this work, we propose Semantics Aware Packet Scheduling (SAPS) algorithm to address these issues. Using packet dependency graph, SAPS algorithm selects a layer to maximize QoS. We aim at minimizing distortion in selecting layers. In inter-frame coded video streaming, minimizing packet loss does not imply maximizing QoS. In determining the packet transmission schedule, we exploit the fact that significance of each packet loss is different dependent upon its frame type and the position within group of picture(GOP). In SAPS algorithm, each packet is assigned a weight called QoS Impact Factor Transmission schedule is derived based upon weighted smoothing. In simulation experiment, we observed that QOS actually improves when packet loss becomes worse. The simulation results show that the SAPS not only maximizes user perceivable QoS but also minimizes resource requirements.

• Proceedings of the IEEK Conference
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• 2003.07d
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• pp.1629-1632
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• 2003

Generally an increasing number of recent applications have found TCP too limiting. There are some characteristics in the transmission of document and binary data which some transmission delay are tolerant but the content must completely be transferred. However voice signals are more sensitive with not some packet loss but some transmission delay. Therefore, Stream Control Transmission Protocol(SCTP) is proposed to minimize the delay and packet loss in the field of delivery of voice signal. SCTP is designed to transport PSTN signalling messages over IP networks, but is capable of broader applications. In this paper, the architecture of SCTP implementation is designed and some interface of SCTP software library which are implemented are specified.

• Journal of Korea Multimedia Society
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• v.19 no.5
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• pp.924-931
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• 2016

In this paper, Various network adaptive MAC scheduling technique is proposed to effectively overcome limits of narrow bandwidth and low transmission speed in underwater. UPFC(Underwater Packet Flow Control) is a technique to reduce both the number of transmission and transmission time using three types (Normal, Blocked, Parallel) of data transmission. In this technique, the load information, in which a transmission node have, is transmitted to destination node using marginal bit in reserved header. Then the transmitted information is referred to determine weighting factor. According to the weighting factor, scheduling is dynamically changed adaptively. The performance of UPFC is analyzed and flow control technique which can be applied to Cluster Based Network and Ad Hoc network as well.

• Journal of electromagnetic engineering and science
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• v.12 no.3
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• pp.189-195
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• 2012

In this paper, we propose a novel protocol called a Multi-hop Diversity Transmission protocol in which the retransmission is realized by a relay that is Nearest to a current Source (MDTNS). We derive the mathematical expressions of the packet error rate (PER) and the average number of transmissions over Nakagami-m fading channels, and verify them by Monte Carlo simulations. The simulation results show that the MDTNS protocol improves the performance of the network in terms of PER when compared to the Multi-hop Diversity Transmission protocol in which the retransmission is done by a relay that is Nearest to Destination (MDTND) and to the conventional multi-hop transmission (CMT) protocol.

• Journal of Multimedia Information System
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• v.5 no.3
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• pp.163-170
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• 2018

We developed a multicast communication packet radio protocol using a time-sharing tablet system ("wireless token ring") to achieve the efficient exchange of files among packet radio terminals attached to swans. This paper provides an overview of the system and the protocol of the packet communications. The packet device forming the main part of the transceiver developed is the Texas Instruments CC2500. This device consists of one call-up channel and one data transmission channel and could improve error frame correction using FEC (forward error correction) with 34.8 kbps MSK and receiving power of at least -64 dBm (output 1 dBm at distance of 200 m using 3 dBi antenna). A time-sharing framework was determined for the wireless token ring using call sign ordinals to prevent transmission right loss. Tests using eight stations showed that resend requests with the ARQ (automatic repeat request) system are more frequent for a receiving power supply of -62 dBm or less. A wireless token ring system with fixed transmission times is more effective. This communication protocol is useful in cases in which frequency resources are limited; the energy consumed is not dependent on the transmission environment (preset transmission times); multiple terminals are concentrated in a small area; and information (position data and vital data) is shared among terminals under circumstances in which direct communication between a terminal and the center is not possible. The method allows epidemiological predictions of avian influenza infection routes based on vital data and relationships among individual birds based on the network topology recorded by individual terminals. This communication protocol is also expected to have applications in the formation of multiple in vivo micromachines or terminals that are inserted into living organisms.

• Journal of Korea Multimedia Society
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• v.7 no.1
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• pp.54-63
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• 2004

This paper describes a hybrid QoS guarantee scheme for high quality audio streaming services on the Internet. The continuous playback of the audio data requires the isochronous transmission of the audio data packet through the Internet. In order to retain the QoS at the ultimate destination (client) as the same as servers provide, the transmission protocols should consider the error conditions such as packet loss, and out of order delivery. Generally, the protocols supporting the transmission of continuous media data do not try to recover the errors. The protocols are working somehow for the toll quality multimedia streaming services, but rot for the high quality streaming services, such as the DVD sound/music payback. The hybrid QoS guarantee scheme includes the three mechanisms to overcome the problem. The selective retransmission for the lost packet, the adaptive buffering at client-side, and the adaptive transmission rate at server-side are totally adopted to recover the packet loss with the minimal overhead, to prevent from the buffer starvation during the retransmission, and to maintain the isochronous transmission even after the retransmission. The experiments have shown good results for the high Quality audio streaming services on the Internet.

• Journal of Korea Multimedia Society
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• v.13 no.12
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• pp.1767-1777
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• 2010

In digital TV broadcasting, such as terrestrial, cable, satellite, and IPTV, the head-end of digital TV broadcasting has a more complicated transmission structure than that of analog TV broadcasting. Furthermore, digital TV broadcasting has a feature that supports multiplex models, such as Multiple Program Transport Stream (MPTS). Therefore, the purpose of our work was to design and examine a more efficient new system of emergency alert message transmission to support the digital TV broadcasting environments. Digital TV broadcasting is the IP generation or RF transmission of 8-VSB, QAM, and QPSK modulated through a multiplexer or re-multiplexer multiplexed stream as a MPEG-2 Transport Stream after content encoding. The new system proposed in this paper transmits an emergency alert message without scrambling after replacing the PID and payload of the -packet with the message prototype in the TS stream from the multiplexer. If we need to transmit an emergency alert message under digital TV broadcasting services, then the receiver first checks the PID of each packet in the TS stream for the emergency alert message. Next, if a packet is determined to be an emergency alert message, then the set-top box displays the message on the TV screen using its function of On Screen Display, or the PC based software displays the message on the monitor screen using its function of overlay with user interface if the packet is found to be an emergency alert message. We have designed an emergency alert message protocol and a system model. By experiments and analysis of the system, we concluded that the system achieved efficiency and the ability to send and receive emergency alert messages using the system under different digital TV broadcasting service environments.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.7B
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• pp.706-718
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• 2002

In bluetooth system, there are two kinds of interference. One is the frequency static interference, for example 802.11 direct sequence, the interferer uses fixed frequency band. Another is frequency dynamic interference, for example other piconets or 802.11 frequency hopping, the interferer uses dynamic frequency channel and cant be estimated. In this paper we introduce a novel solution of hybrid method of Listen-Before-Talk (LBT) and Adaptive Frequency Hopping (AFH) to address the coexistence of bluetooth and Direct Sequence of wireless local area network (WLAN). Before any bluetooth packet transmission, in the turn around time of the current slot, both the sender and receiver sense the channel whether there is any transmission going on or not. If the channel is busy, packet transmission is withdrawn until another chance. This is the LBT in Bluetooth. Because of asymmetry sense ability of WLAN and bluetooth, AFH is introduced to combat the left front-edge packet collisions. In monitor period of AFH, LBT is performed to label the channels with static interference. Then, all the labeled noisy channels are not used in the followed bluetooth frequency hopping. In this way, both the frequency dynamic and frequency static interference are effectively mitigated. We evaluate the solution through packet collision analysis and a detail realistic simulation with IP traffic. It turns out that the hybrid method can combat both the frequency dynamic and frequency static interference. The packet collision analysis shows it almost doubles the maximal system aggregate throughput. The realistic simulation shows it has the least packet loss.

• Proceedings of the IEEK Conference
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• 2002.06e
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• pp.83-86
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• 2002

In this paper while a router is routing all packet to the next hop, it inspects whether there is congestion on this current hop router or not and if the router discovers that it has some congestion, it informs that the packet is experienced to congestion. The packet arrived to next hop including some information about the congestion is processed first and it has wider bandwidth than another packet The amount of congestion is recorded to the DS field of IP header by congestion experience level. In the next hop when the packet including the congestion information is routed, the standard packet dropping ratio of the current router is changed in proportion to congestion experience that is recorded in IP header on of that. When the packet that has experienced congestion before is arrived, the router extends the drop threshold value not to drop the packet. It mean that transferring the audio or video stream, if the packet is already experienced the congestion in another hop, the router can provide the better service quality about 15∼25% than another.