• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.297-312
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• 2005

The multimedia traffic of continuous video and audio data via streaming service accounts for a significant and expanding portion of the Internet traffic. This streaming data delivery is mostly based on RTP with UDP. However, UDP does not support congestion control. For this reason, UDP causes the starvation of congestion controlled TCP traffic which reduces its bandwidth share during overload situation. In this paper, we propose a new TCP-friendly rate control scheme called 'TF-RTP(TCP-Friendly RTP)'. In the congested network state, the TF-RTP exactly estimates the competing TCP's throughput by using the modified parameters. Then, it controls the sending rate of the video streams. Therefore, the TF-RTP adjusts its sending rate to TCP-friendly and fair share with competing TCP traffics. Through the simulation, we prove that the TF-RTP correctly estimates the TCP's throughput and improves the TCP-friendliness and fairness.

• Journal of KIISE:Computing Practices and Letters
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• v.8 no.3
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• pp.328-335
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• 2002

When we want to transfer video streaming data through computer networks, we will have to be allocated a larger bandwidth compared to a general application. In general, it causes a serious network overload inevitably due to the limited bandwidth. In this paper, in order to resolve the problem, we haute taken a method for transmitting video streaming data using RTP and RTCP. With RR(Receiver Report) packet in RTCP we will test it to check whether the traffic in a network has occurred or not. If it happened, we haute tried to reduce the overall network traffic by dynamically changing the quantization factor of the Motion JPEG that is one of the encoding styles in JMF. When the ratio of the average of transmission for each session to the average of overall transmission is greater than 7%, we should adjust the amount of data to be transmitted for each session to reach the session mean values. The experimental results show that the proposed method taken here reduces the overload effectively and therefore improves the efficiency for transmitting video streaming data.

• The KIPS Transactions:PartA
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• v.16A no.5
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• pp.327-338
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• 2009

Recently, with the advance of computing and networking technique, the high speed internet becomes widespread, however, it is still hard job to do streaming the media which requires high network bandwidth over the internet. Previous researches for streaming over the internet mainly proposed techniques that controls the QoS(Quality of Service) of the media in proportion to the network status. Though, this could be the solution for the service provider while the service user who wants constant QoS may not satisfy with variable QoS. In the paper, we propose a network adaptive prefetching technique, PAP, for guarantee of constant QoS. The PAP prefetches frames by increasing the frame transmission rate while the available network bandwidth is high. The PAP uses the prefetched frames to guarantee the QoS while the available network bandwidth is low and increases the playout interval to prevent buffer underflow. The experiment result shows that the proposed PAP could guarantee the constant QoS by prefetching the frames adaptively to the network bandwidth with the characteristic of video stream.

• Journal of IKEEE
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• v.23 no.2
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• pp.599-605
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• 2019

Ferrite cores are used as a soft magnetic material in the fabrication of couplers for inductive powerline communication (PLC). However, it is difficult to adjust the size freely according to the power-line and power-grid environment. In this paper, we report that a nano-crystalline alloy with higher permeability than ferrite can be used as an inductive coupler for non-contact PLC. Since nano-crystalline are produced in the form of a thin ribbon, the size of the coupler can be freely controlled by the number of ribbons wound on the toroidal core. It was fabricated with induction type coupler and showed to be suitable for non-contact power line communication. Experimental results show that the communication bandwidth is 45 Mbps for 100 m and 8 Mbps for 200 m under the current fluctuation of less than 100 A, and the reception ratio is 100%.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.11
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• pp.77-84
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• 2003

This paper presents a multi-functional microstrip spital antenna that operates in dual frequency bands. Several types of beam shape can be selected by controlling the phase difference of two spiral arms with the phase shifters located on each feed line. It has a normal beam at the lower frequency band, and four different patterns at the higher frequency band: normal beam, conical beam and two types of tilted beam. The antenna exhibits more than 10% of bandwidth at each band. The antenna is fabricated with conductor backed electromagnetic absorber in order to attain unidirectional radiation pattern and confirmed the multi-functionality by measurements.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.1
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• pp.90-96
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• 2014

This paper presents a practical linearization technique for a wide-band bang-bang digital phase locked-loop(BBDPLL) by selecting optimal loop gains. In this paper, limitation of the theoretical design method for BBDPLL is explained, and introduced how to implement practical BBDPLLs with CMOS process. In the proposed BBDPLL, the limited cycle noise is removed by reducing the proportional gain while increasing the integer array and dither gain. Comparing to the conventional BBDPLL, the proposed one shows a small area, low power, linear characteristic. Moreover, the proposed design technique can control a loop bandwidth of the BBDPLL. Performance of the proposed BBDPLL is verified using CppSim simulator.

• Proceedings of the KIEE Conference
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• 2008.10b
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• pp.503-504
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• 2008

최근 정보통신 기술들이 빠르게 발전하고 있다. 다양한 통신 기술들 중에서도 업무의 효율을 높이고자 회사 및 가정, 학교 등에서 자주 사용되고 있는 영상채팅시스템을 구현해보고자 한다. 쿼타임 코덱 중 가장 보편적인 코덱으로 인코딩이 쉽고 저사양의 CPU만으로도 실시간 스트리밍이 가능한 H.263 코덱을 사용하여 영상채팅시스템을 Visual C++로 구현을 하였다. 전송로의 지연을 줄이기 위하여 영상, 음성, 텍스트 등을 압축하고 복원하는 데 걸리는 시간을 최소화기 위하여 데이터의 전송대역폭을 적절히 조절하는 알고리듬을 제안하여 전송지연을 최소화하였다. 또한 P2P 방식을 사용하여 다양한 영상 환경에 대하여 영상 및 텍스트 데이터의 안정성과 화질이 우수함을 보였으며, 실시간 가스안전관리 상담에 이용하여 업무의 효율을 높이고자 한다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.8
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• pp.1647-1653
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• 2004

In a contention-based wireless LAN protocol, maximization of channel utilization and fair bandwidth allocations are main topics to deal with. But it is very difficult to achieve these two goals simultaneously. Many studies have been done to achieve these goals. In this paper we propose a control mechanism to support fair transmissions among traffic classes in IEEE 802.11e Wireless LAN. The proposed algorithm uses short-term and long-term transmission times of each traffic classes to control their $CW_{min}$ for fairness. The proposed algorithm don't need to know the exact number of nodes in the networks to support fairness as other studies do. Furthermore any modifications in AP and mobile hosts are not required.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2011.07a
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• pp.188-191
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• 2011

본 논문에서는 시간적 움직임 벡터 성분 값을 절삭하여 시간적 움직임 벡터를 저장하는데 필요한 메모리 요구량을 감소하는 방법에 대해서 제안한다. HEVC 에서 시간적 움직임 벡터는 움직임 병합 모드와 향상된 움직임 벡터 예측 방법에서 부호화 효율 향상을 위해서 사용되고 있다. 하지만 부호화해야 할 원본 영상의 공간적 해상도 및 비트 심도가 증가함에 따라 메모리에 저장할 시간적 움직임 벡터의 데이터뿐만 아니라 메모리 접근 대역폭도 크게 증가한다. 따라서 본 논문에서는 시간적 움직임 벡터 성분의 비트 심도를 조절한 후 메모리에 저장하는 방법에 대해서 제안한다. 제안하는 방법을 HM 2.0 에 구현하고 HEVC 표준화에서 사용되는 임의접근 및 저지연 실험 조건에서 실험했을 때, 평균 0.2%의 비트율 손실을 보이지만, 필요한 메모리 요구량을 약 반 정도 줄일 수 있다.

• The Journal of the Korea Contents Association
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• v.8 no.2
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• pp.67-73
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• 2008

Recently, due to the rapid advances of mobile communication, multimedia service can be provided by mobile devices. Cellular network tends to uses SIP as a call setup protocol in order to provide various multimedia service to consumers. Cellular network holds the characteristic of higher BER(Bit Error Rate), narrower bandwidth than the wired network. The value of SIP RTI(Retransmission Time interval) based on the wired network decreases a network efficiency and increases a call setup delay over cellular network. By using NS-2 simulator, we show new SIP RTI value adequate over cellular network. We design and implement home network by using the modified SIP that is suitable for cellular network.