• Proceedings of the Korean Information Science Society Conference
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• 1998.10a
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• pp.229-231
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• 1998

본 논문에서 제안하는 매체접근제어 프로토콜은 활성 단말의 개수를 통계적 특성에 의하여 예측하고 이를 상향 요구 슬롯의 액세스 확률에 적용하여 요구슬롯의 수율(Throughput)을 높이도록 하였으며, 단말에세 슬롯을 할당하는 방식으로 고정할당법과 동적 할당법을 동시에 사용하였다. 고정 할당법은 모든 단말에게 최송한 제공될 수 있는 서비스 품질(QoQ: Quality of Server)을 보장하는 방식이며 동적할당법은 음성 단말에 대하여 버퍼의 상태에 따라 지정된 서비스 품질을 제공하기 위하여 각 단말에게 동적으로 슬롯을 할당하는 방법이다. 제안된 프로토콜에 대해 시뮬레이션을 통하여 음성 및 데이터 단말의 수율과 패킷 전송 지연을 구하고 음성 단말의 패킷 손실률을 분석하여 동적 할당법의 효율에 대하여 알아본다.

• The Journal of Natural Sciences
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• v.8 no.2
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• pp.93-99
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• 1996

In this study, we compare performance of priority schemes of FDDI and Expressnet for integrated voice and data traffic through simulation. The voice capacity of FDDI is higher than that of Expressnet for all cases considered. When compared to Expressnet, FDDI achieves a higher data throughput for file transfer traffic but it incurs a longer delay for interactive traffic.

• Journal of Korea Multimedia Society
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• v.8 no.7
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• pp.1012-1021
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• 2005

This paper describes the design and implementation of an efficient multi-point multimedia conference system using IP grouping. Existing multi-point multimedia conference systems are difficult for multi-user to perform efficient cooperation due to bandwidth limitation for data transmission of video, audio and documentation. In the case that multi-user uses limited bandwidth, smooth cooperation does not accomplish due to transmission delay for the real-time transmission of image and speech data. A hybrid transfer method which is mixed with distributed and centralized methods is used for smooth cooperation, and the network bandwidth is reduced by forming multi-user conference systems of IP grouping in this paper. Also, adaptive image frame variations are used to solve bottleneck effect according to the number of users. An efficient multi-user conference system is designed to support audio quality.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.2A
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• pp.95-104
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• 2010

In wireless communication system, a variety of retransmission algorithms are used in order to improve the quality of service of users. But the system may be inefficient because retransmission algorithms operate independently with other layers. Also, the quality of service can be degraded due to the unnecessary retransmission of packets. To solve these problems, the study on the cross-layer retransmission schemes have been widely performed. However, in order to apply cross-layer retransmission schemes to wireless communication system, whether the performance of cross-layer retransmission schemes meets QoS requirements of each service class has to be verified. Thus, this paper proposes the mathematical model for analyzing the performance of the cross-layer retransmission schemes and derives both the suitable retransmission scheme and the optimal retransmission parameter on each service class. The proposed mathematical model selects the MCS level based on channel state information and The performance analysis is comparatively easy in case that HARQ, ARQ, and AMC schemes are combined. The proposed mathematical model also enables the analysis of the packet transmission delay. To utilize the analytical model, this paper derives the suitable retransmission scheme and the optimal retransmission parameter for delay sensitive services in WiMAX system. Also, the proposed analytical model can be used to analyze the performance of wireless communication system such as LTE and WLAN.

• Proceedings of the Korean Information Science Society Conference
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• 2001.10c
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• pp.670-672
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• 2001

최근 인터넷에서는 메일, 파일 전송 등의 비실시간 응용 뿐만 아니라 음성, 비디오 등의 실시간 응용 서비스를 이용하려는 요구가 증가하고 있다. 이를 위해 광대역 서비스 뿐만 아니라 사용자별 차별화된 QoS의 보장이 요구된다. 네트워크를 통해 전달되는 패킷의 지연 시간과 데이터 처리율, 그리고 손실은 서비스에 따른 요구사항 등이 QoS의 주요 내용인데, 현재의 인터넷은 모든 패킷을 동일하게 전달하는 최선의 노력(Best-Effort)만을 제공하고 있기 때문에 서비스에 따른 패킷의 전달 지연과 지연 변이에 대한 요구사항을 보장해 주지 못하고 있다. 본 연구에서는 분산된 네트워크 환경에서 효율적인 QoS 관리를 위해 이동 에이전트(Mobile Agents)를 이용한 모니터링 시스템을 설계 및 구현하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.1A
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• pp.96-103
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• 2000

The voice traffic is one of the most important real-time objects in WATM(Wireless Asynchronous Transfer Mode) networks. In this paper, we propose a new MAC(Medium Access'Control) protocol for the efficienttransmission of voice traffic over WATM networks in the multimedia environment and compare the performanceto existing similar protocols. The new protocol separates the reservation slot period for voice and that for data toguarantee some level of QoS(Quality of Service) in voice traffic. This is denoted by a slot assignment functiondepending on the frame size. According to the characteristics of voice traffic which is repeatedly in silent states,the protocol allocates voice reservation request slots dynamically with respect to the number of silent(off state)voice sources and also sends the first block of talkspurt restarted after silent period with a reservation requestslot to reduce the access delay.The simulation results show that the proposed protocol has better performance than Slotted Aloha in bandwidthefficiency, and can serve a certain level of QoS by the given slot assignment function even when the number ofvoice terminals varies dynamically. This means we can observe that the new MAC protocol is much better thanthe NC-PRMA(None Collision-Packet Reservation Multiple Access) protocol.

• The KIPS Transactions:PartC
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• v.16C no.5
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• pp.661-668
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• 2009

The Extended real-time Polling Service (ErtPS) is proposed tosupport QoS of VoIP service with silence suppression which generates variable size data packets in IEEE 802.16e systems. If the silence is suppressed, VoIP should support Comfort Noise Generation (CNG) which generates comfort noise for receiver's auditory sense to notify the status of connection to the user. CNG mode in silent-period generates a data with lower bit rate at long packet transmission intervals in comparison with talk-spurt. Therefore, if the ErtPS, which is designed to support service flows that generate data packets on a periodic basis, is applied to silent-period, resources of the uplink are used inefficiently. In this paper, we proposed the Improved ErtPS algorithm for efficient resource utilization of the silent-period in VoIP traffic supporting CNG. In the proposed algorithm, the base station allocates bandwidth depending on the status of voice at the appropriate interval by havingthe user inform the changes of voice status. The Improved ErtPS utilizes the Cannel Quality Information Channel (CQICH) which is an uplink subchannel for delivering quality information of channel to the base station on a periodic basis in 802.16e systems. We evaluated the performance of proposed algorithm using OPNET simulator. We validated that proposed algorithm improves the bandwidth utilization of the uplink and packet transmission latency

• The Transactions of the Korea Information Processing Society
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• v.7 no.4
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• pp.1211-1216
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• 2000

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rateot send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link. An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used fro feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.10a
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• pp.693-697
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• 2007

The SSB modem performs the modulation process which converts the digital voltage level to the audible frequency band signal and the demodulation process which converts reversely the audible frequency signal to the digital voltage level. The modulator and the demodulator are implemented with a single DSP chip. Because of the SSB specific character, the distortion occurs when the frequency is changed. This distortion has no effect on voice communication, but it has an significant effect on data communication. In other words, it is impossible to send data stream with adjacent 2 periods. Therefore, in case of using 2-tone FSK, it is needed to send at least 3 periods to transmit 1 bit. Therefore we implemented the modem using modified phase-delay shift keying to transmit 1 tone signal for high speed transmission. In the 1200[bps] mode, it generates 0, $187{\mu}s$ delay time at 1.3kHz symbol frequency, and in the 2400[bps] mode, 0, $70{\mu}s$, $130{\mu}s$, $200{\mu}s$ delay time at 1.5kHz symbol frequency. Finally, in the maximum 3600[bps] mode, it generates 0, $100{\mu}s$, $160{\mu}s$, $250{\mu}s$ delay time at 2.0kHz symbol frequency. The measured results of the implemented SSB modem shows a good transfer functional characteristic by spectrum analyzer, almost same bandwidth in pass band and 20dB higher SNR comparing the German PACTOR and American CLOVER and in the experimental transmitting test, we verified the transmitted data is received correctly in platform.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.11 no.10
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• pp.1852-1857
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• 2007

The SSB modem performs the modulation process which converts the digital voltage level to the audible frequency band signal and the demodulation process which converts reversely the audible frequency signal to the digital voltage level. The modulator and the demodulator are implemented with a single DSP chip. Because of the SSB specific character, the distortion occurs when the frequency is changed. This distortion has no effect on voice communication but it has an significant effect on data communication. In other words, it is impossible to send data stream with adjacent 2 periods. Therefore, in case of using 2-tone FSK, it is needed to send at least 3 periods to transmit 1 bit. Therefore we implemented the modem using modified phase-delay shift keying to transmit 1 tone signal for high speed transmission. In the 1200[bps] mode, it generates 0, $187{\mu}s$, delay time at 1.3kHz symbol frequency, and in the 2400[bps] mode, 0, $70{\mu}s\;130{\mu}s\;200{\mu}s$, delay time at 1.5kHz symbol frequency. Finally, in the maximum 3600[bps] mode, it generates 0, $100{\mu}s\;160{\mu}s\;250{\mu}s$ 2.0kHz symbol frequency. The measured results of the implemented SSB modem shows a good transfer functional characteristic by spectrum analyzer, almost same bandwidth in pass band and 20dB higher SNR comparing the emu FACTOR and American CLOVER and in the experimental transmitting test, we verified the transmitted data is received correctly in platform.