• Proceedings of the Korean Information Science Society Conference
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• 2002.04a
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• pp.241-243
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• 2002

본 논문은 무선 랜 상에서 전송 지연을 보장하기 위한 백오프 방식을 제안한다. 이 방식은 전송되는 트래픽의 종류에 따라 백오프 시간을 할당하도록 설계되었으며, 기존의 IEEE 802.11 매체접근제어 방식을 그대로 유지하며, 보다 향상된 서비스 품질을 지원한다. 제안된 백오프 방식에서는 모든 트래픽을 실시간 트래픽과 비실시간 트래픽으로 구분하여 처리된다. 실시간 트래픽은 제안된 백오프 방식으로 처리되고, 비실시간 트래픽은 기존의 IEEE 802.11 방식에 의해 처리된다. 본 논문에서는 컴퓨터 시뮬레이션을 사용하여 제안된 백오프 방식을 성능 평가하였으며, 시뮬레이션 결과는 전송지연 관점에서 기존의 IEEE 802.11e 방식보다 성능이 향상됨을 보여준다.

• KIPS Transactions on Computer and Communication Systems
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• v.10 no.2
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• pp.29-38
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• 2021

The Internet including mobile networks is developing to overcoming the limitation of physical distance and providing or acquiring information from remote locations. However, the systems that use video as primary information require higher bandwidth for recognizing the situation in remote places more accurately through high-quality video as well as lower latency for faster interaction between devices and users. The emergence of the 5th generation mobile network provides features such as high bandwidth and precise location recognition that were not experienced in previous-generation technologies. In addition, the Mobile Edge Computing that minimizes network latency in the mobile network requires a change in the traditional system architecture that was composed of the existing smart device and high availability server system. However, even with 5G and MEC, since there is a limit to overcome the mobile network state fluctuations only by enhancing the network infrastructure, this study proposes a high-definition video streaming system in ultra-low latency based on the SRT protocol that provides Forward Error Correction and Fast Retransmission. The proposed system shows how to deploy software components that are developed in consideration of the nature of 5G and MEC to achieve sub-1 second latency for 4K real-time video streaming. In the last of this paper, we analyze the most significant factor in the entire video transmission process to achieve the lowest possible latency.

• Proceedings of the Korean Information Science Society Conference
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• 2012.06d
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• pp.404-406
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• 2012

최근 네트워크 기술의 발달과 모바일 기기 성능의 향상으로 대화형 멀티미디어 서비스에 대한 요구가 증가하고 있다. 대화형 멀티미디어 서비스는 음성과 영상을 실시간으로 주고받는 서비스로 지연에 민감한 특성을 가진다. 따라서 대화형 멀티미디어 서비스의 품질을 보장하기 위해서는 재생 지연한계를 고려해야 한다. 본 논문에서는 대화형 멀티미디어 서비스의 단대단 지연한계를 보장하기 위해서 네트워크와 송신버퍼 그리고 수신버퍼에서 발생하는 지연을 고려하여 전송률을 조절하는 기법을 제안한다. 제안하는 기법은 송신버퍼와 수신버퍼에 지연한계를 설정하고 이를 보장하기 위해 전송률을 조절함으로써 지연한계와 재생의 연속성을 보장하였다. 실험을 통해 본 논문이 대화형 멀티미디어 서비스의 단대단 지연한계를 보장함을 보였다.

• The Journal of the Korea Contents Association
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• v.8 no.1
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• pp.177-185
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• 2008

Research on media access control (MAC) scheme for Wireless Sensor Network (WSN) has been mainly focused on energy efficiency improvement, while interest on latency is relatively weak. However, end-to-end latency could be a critical limitation specifically in the multi-hop network such as wireless multimedia sensor networks. In this paper we propose a media access control scheme with distributed transmission power control to Improve end-to-end transmission latency as well as reduce power consumption in multi-hop wireless sensor networks. According to the simulation results, the proposed scheme is turned out to be an energy efficient scheme with improved end-to-end transmission latency.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.9A
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• pp.2305-2312
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• 1998

This paper analyzed the delay characteristics of speech packets in virtual cellular network(VCN). The probability distribution of packet delay is obtained using the markov chain model when periodic speech packets are transmitted by slotted-ALOHA protocol. The effects of probility of capture and retransmission policy on the performance were also analyzed. At first, the probability cumulative function of packet delay is calculated from the probability of capture as a function of location of mobile terminal. In order to investigate the effects of backoff delay, we defined a parameter NPr, where N is the period (frame size) of the speech packets and Pr is the retransmission probability for each speech packet. We also obtained the 1% outage delay for various frame size N.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8A
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• pp.600-609
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• 2009

The traditional carrier sense multiple access (CSMA) protocol like IEEE 802.11 Distributed Coordination Function (DCF) does not handle the constraints adequately, leading to degraded delay latency and throughput as the network scales are enlarged. We present more efficient method of a medium access for real-time wireless sensor networks. Proposed MAC protocol is like the randomized CSMA protocol, but unlike previous legacy protocols, it does not use a time-varying contention window from which a node randomly picks a transmission slot. To reduce the latency for the delivery of event reports, we carefully decide to select a fixed-size contention window with non-uniform probability distribution of transmitting in each slot. We show that the proposed method can offer up to severaansimes latency reduction compared to legacy of IEEE 802.11 as the size of the sensor network scales up to 256 nodes using widely using network simulation package,caS-2. We finally show that proposed MAC scheme comes close to meet bounds on the best latency being achieved by a decentralized CSMA-based MAC protocol for real-time wireless sensor networks which is sensitive to delay latency.

• Journal of the Institute of Electronics and Information Engineers
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• v.49 no.11
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• pp.9-16
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• 2012

In this paper, we focus on end-to-end transfer delay improvement by using network coding and propose a scheme where a message is divided into several packets which are network coded generating additional redundancy with the results that the number of transmitted packets increases. In networks with high coefficient of variation of transfer time, increased number of packets could reduce the transfer time of the message to a destination. For the proposed scheme, we investigate the optimum number of divided packets and redundancy considering transfer delay reduction and additional transmission cost caused by using network coding under the restriction of maximum transmission packet size. From the results of the investigation, we see that the proposed scheme is effective in networks having high variability of transfer time and would be very useful and practical especially for the case that expedited deliveries of messages are needed.

• Proceedings of the Korean Information Science Society Conference
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• 2005.11a
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• pp.223-225
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• 2005

센서네트워크에서 이동물체의 경로 추적과 같이 실시간으로 패킷이 목적노드에 도착하여야 데이터가 유효한 응용의 경우에는 패킷 전송의 실시간을 보장할 수 있는 전송경로설정이 필요하다. 본 논문에서는 패킷을 전송하는데 있어서 지연시간을 증가시키는 요소인 혼잡상황과 낮은 링크 퀄리티를 가지는 경로를 회피함으로 전송지연시간을 최소로 줄이는 전송경로 설정 기법을 제안한다 네트워크의 위상에 따라 노드마다 다른 양상을 보이는 채널 부하도를 효과적으로 측정하고 네트워크의 혼잡상황을 판단하는 기준으로 삼는다. 이와 함께 링크의 전송 성공률을 이용하여 이웃 노드들의 전송경로 선택의 적합도를 계산하고 확률적으로 경로를 선택한다.

• Proceedings of the Korean Information Science Society Conference
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• 2007.10d
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• pp.597-601
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• 2007

본 논문에서는 휴면 지연에 의한 전송지연을 줄이는 효과적인 방법으로 staggered schedule 방식에 기반하여 동작하는 SPEED-MAC 프로토콜을 제안하고자 한다. SPEED-MAC은 이벤트 발생에 따른 정보를 발생사실과 세부정보로 나누어 각각 이벤트 공지 구간과 데이터 전송 구간에 싱크 노드로 전송함으로써 이벤트 발생 사실을 싱크 노드에게 최단시간에 전송할 수 있다. 또한 제어 패킷의 오버헤드를 줄이기 위해 이벤트 공지 구간에서 signal 패킷을 사용함으로써 데이터 패킷을 전송할 때 발생 가능한 충돌을 예측하였고 제어 패킷을 동적으로 사용하여 노드의 에너지 효율성을 추가로 개선할 수 있게 되었다. 본 논문에서는 SPEED-MAC을 MICA MOTE에서 구현하였으며 기존의 MAC 프로토콜에 비하여 메시지 전송 지연시간을 대폭 단축시킬 수 있었으며 또한 저 전력을 구현할 수 있는 것을 확인하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.4
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• pp.940-949
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• 1998

The paper proposes an effective mechanism hiring the adaptive buffer control schemes on the mobile seceiver's side that can resolve the problems of delay spikes caused by the handover process onthe cell boundaries when they transmit real-time moving images through the high speed wireless channels. to confirm the effectiveness of the proposed schemes, we have modeled the characteristics of the random errors on the wireless channels, the burst errors and that of the transmission delay by the handover. We also compared the rate of the delay-adaptation of the receive buffers caused by the delay and delay spikes. the comparison was done by testing the suggested schemes against the existing schemes by applying tehm to the packet delay model. We also could identify the noticable reduction of the casesof buffer delay and overflow compared with the conventional schemes, by applying the suggested algorithm to the video image compressed by H.263.