• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.21 no.11
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• pp.1269-1274
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• 2010

Wireless communication channels with the most severe multipath fading phenomenon that appears each time a different delay is caused by the frequency selective fading. At this time, ISI due to the performance degradation of wireless communication channels and data transfer speed gives the constraints. OFDM technique can remove ISI inserting longer guard interval than channel delay spread of channel between symbol. However, the multi-path delay of the channel to be serious with the guard interval can not eliminate ISI. In this case, using the equalizer must compensate. Especially, use of equalizer is need absolutely as data rate becomes high speed. In this paper, we analyze the BER performance with pre-equalization for MIMO-OFDM over fading channel. The results of this paper can be applied to MIMO-OFDM system with equalization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.6A
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• pp.506-514
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• 2009

To minimize energy consumption, most of MAC Protocols in WSNs exploit low duty cycling. Among those, RMAC [4] allows a node to transmit a data packet for multiple hops in a single duty cycle, which is made possible by exploiting a control frame named Pioneer (PION) in setting up the path. In this paper, we present a MAC Protocol called Hop Extended MAC (HE-MAC) that transmits the data packet for more multiple hops in a single duty cycle. It employs an EXP (Explorer) frame to set up the multiple hop transmission, which contains the information of the maximum hop that a packet can be transmitted. With the use of the information in EXP and an internal state of Ready to Receive (RTR), HEMAC extends the relay of the packet beyond the termination of the data period by two more hops compared to RMAC. Along with our proposed adaptive sleeping method, it also reduces power consumption and handles heavy traffic efficiently without experiencing packet inversion observed in RMAC. We analytically obtain the packet delivery latency in HE-MAC and evaluate the performance through ns-2 simulations. Compared to RMAC, HE-MAC achieves 14% less power consumption and 20% less packet delay on average for a random topology of 300 nodes.

• Annual Conference of KIPS
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• 2014.11a
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• pp.160-163
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• 2014

IP 네트워크를 통해 전송되는 실시간 스트라밍은 네트워크 상태에 따라 데이터 손실 및 전송 지연의 문제가 발생 할 수 있다. 이에 멀티미디어 네트워킹 애플리케이션은 RTP(Realtime Transport Protocol)와 RTCP(RTP Control Protocol)를 지원하는 RTSP(Realtime Streaming Protocol)를 사용한다. RTSP를 사용하는 송신단은 패킷 손실과 전송지연이 발생할 경우 끊김 없는 멀티미디어 스티리밍을 제공하기 위해 네트워크 상태에 따라 전송률을 변경한다. 하지만 전송률이 낮아지면 멀티미디어 품질이 떨어지게 된다. 만약 짧은 시간 동안 전송률이 빈번하게 변경된다면 일관성 있는 품질의 서비스를 제공하기 어렵다. 본 논문에서는 SDN 환경에서 실시간 스트리밍을 효과적으로 제어하기 위해 중앙 컨트롤러가 RTCP를 수신하고 송신단의 전송률과 현재 링크의 가용 전송율을 추정한다. 추정된 가용 전송율이 송신단의 전송률보다 낮다며 각 스위치의 활성화된 플로우 엔트리가 낮은 스위치로 라우팅 경로를 선택하는 메커니즘을 제안한다.

• Journal of Korea Multimedia Society
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• v.10 no.8
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• pp.1039-1051
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• 2007

In this paper, a new lip synchronization control method is proposed to achieve very low delay in the video communication. The lip control is so much vital in video communication as delay reduction. In a general way, to control the lip synchronization, both the playtime and capture time calculated from RTP time stamp are used. RTP timestamp is created by stream sender and sent to the receiver along the stream. It is extracted from the received packet by stream receiver to calculate playtime and capture time. In this paper, we propose the method of searching most adjacent corresponding frame of the audio signal, which is assumed to be played with uniform speed. Encoding buffer of stream sender is removed to reduce the buffering delay. Besides, decoder buffer of receiver, which is used to correct the cracked packet, is resulted to process only 3 frames. These mechanisms enable us to achieve ultra low delay less than 100 ms, which is essential to video communication. Through simulations, the proposed method shows below the 100 ms delay and controlled the lip synchronization between audio and video.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.112-114
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• 2002

인터넷은 무선 이동통신 환경에서 높은 지연 시간과 낮은 전송률로 인해 대용량의데이터를 전송하기 위한 기존의 TCP/IP, HTTP등의 인터넷 프로토콜은 적합하지 않다. 이러한 문제점을 해결하기 위해서 WAP이 등장하였으며, 성능 개선을 위해 여러 단체에서 연구 수행 중 이다. 본 논문에서는 Prefetching Scheme을 기존 WAP에 적용하여 그에 따른 성능변화를 시뮬레이션을 통해 분석한다.

• Journal of Advanced Marine Engineering and Technology
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• v.41 no.1
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• pp.56-61
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• 2017

This paper presents a data prediction method and its application to haptic-based teleoperation systems. In general, time delays inevitably occur during data transmission in a network environment, which degrades the overall performance of haptic-based teleoperation systems. To address this situation, this paper proposes an autoregressive moving average (ARMA) model-based data prediction algorithm for estimating model parameters and predicting future data recursively in real time. The proposed method was applied to haptic data captured every 5 ms while bilateral haptic interaction was carried out by two users with an object in a virtual space. The results showed that the prediction performance of the proposed method had an error of less than 1 ms when predicting position-level data 100 ms ahead.

• 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.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.15 no.3
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• pp.1711-1717
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• 2014

There have been many studies on IEEE 802.15.4 for home area networks(HAN) in Smart Grids. Existing unslotted or slotted IEEE 802.15.4 has almost not met strict conditions of the U.S. Department Of Energy(DOE). This study proposed a improved algorithm that reduces collisions, delay time and changes in the delay time. For this purpose, numbers were given to nodes to make the transmission in the order of the node numbers. Since the probability of the occurrence of collisions would decrease compared to random transmission if the nodes were given numbers, Backoff time was set at 0. In the proposed Numbered-Unslotted-ZeroBackoff algorithm, when the packet size was 133 octets and less than 180 packets per second occurred, it was found that packet delivery ratio was over 99.99%, and that all the maximum delay, the mean delay and the minimum delay were less than 0.02 seconds. This paper could confirm that the algorithm proposed in this study met the strict conditions of the DOE.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.11B
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• pp.969-978
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• 2006

Recently, there has been explosive growth in Internet service contents, and we have witnessed the emergence of many real-time applications on the Internet. These types of applications require special support from the underlying network such as reliability, timeless, and guaranteed delivery, as well as different levels of service quality. Estimation network status and transcoding data rate are researching for QoS of multimedia service by using the RTP/RTCP. We propose two schemes that the way to calculate the one-way delay by expanding the structure of RTP/RTCP in Internet that provides us 'best effort' service and to estimate the network condition by using this in this paper. And we propose the dynamic RTCP mechanism based on one-way delay to deal with rapidly changed network situation immediately. In simulation, proposed monitoring schemes using the one-way delay has similar in performance to monitoring schemes using the GPS, proposed dynamic RTCP scheme obtain the desired result of decreased packet loss rate by sending RTCP RR packet lower than other schemes.