• Title/Summary/Keyword: 제어 패킷

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Wakeup period Control Mechanism for traffic Change Environment (트래픽 변화 환경을 위한 Wakeup 주기 제어 메커니즘)

  • Jeon, Jun-Heon;Kim, Seong-Cheol;Kim, Yeong-Joon;Kim, Hye-Yun;Kim, Joong-Jae;Kim, Hye-Yun
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2013.10a
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    • pp.519-521
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    • 2013
  • 무선 센서 네트워크에서는 에너지가 제한된 배터리로 네트워크를 구성하기 때문에 에너지 효율적 사용에 대한 연구가 주요한 이슈이다. 본 논문에서는 효율적인 에너지 사용을 위하여 트래픽 변화 환경을 위한 Wakeup 주기 제어 메커니즘을 제안한다. 제안되는 MAC 프로토콜은 수신자의 제어 신호로 데이터 전송이 시작된다. 송신 노드는 트래픽이 변화를 데이터 패킷 프레임에 플래그(flag)를 추가하여 수신노드에 전달한다. 수신 노드는 이를 통해 수신 노드의 Wakeup 주기를 제어한다. 제안되는 MAC 프로토콜은 트래픽이 적을 경우 수신 노드의 sleep 구간의 증가를 통하여 에너지가 절약된다. 또한 트래픽이 높은 경우 수신 노드의 Wakeup 주기를 줄여 송신 노드의 idle listening으로 발생하는 에너지 소모를 감소시킨다. 제안되는 MAC 프로토콜은 기존의 프로토콜과 비교하여 빠르게 Wakeup 주기를 조절함으로서 에너지 효율적면에서 더 좋은 성능을 보여준다.

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Wireless Measurement based TFRC for QoS Provisioning over IEEE 802.11 (IEEE 802.11에서 멀티미디어 QoS 보장을 위한 무선 측정 기반 TFRC 기법)

  • Pyun Jae young
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.30 no.4B
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    • pp.202-209
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    • 2005
  • In this paper, a dynamic TCP-friendly rate control (TFRC) is proposed to adjust the coding rates according to the channel characteristics of the wireless-to-wired network consisting of wireless first-hop channel. To avoid the throughput degradation of multimedia flows traveling through wireless lint the proposed rate control system employs a new wireless loss differentiation algorithm (LDA) using packet loss statistics. This method can produce the TCP-friendly rates while sharing the backbone bandwidth with TCP flows over the wireless-to-wired network. Experimental results show that the proposed rate control system can eliminate the effect of wireless losses in flow control of TFRC and substantially reduce the abrupt quality degradation of the video streaming caused by the unreliable wireless link status.

A Performance Study on The Advanced Peer-to-Peer Network for Broadband Communications (Advanced Peer-to-Peer Network에서의 초고속 통신망의 성능연구)

  • 황명상;류제영;주기호;박두영
    • Proceedings of the Korea Institute of Convergence Signal Processing
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    • 2000.12a
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    • pp.9-12
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    • 2000
  • In this paper, we carry out a performance study related to the Advanced Peer-to-Peer Network(APPN). For this particular network, it has been proposed to use the leaky bucket as a way of controlling congestion within the network. On the top of leaky bucket type rate based congestion control scheme for high speed networks, a user will typically operate an error control scheme for retransmitting lost and erroneous packets. We propose a Perform ance model in order to study the Interaction between a user's error control scheme and the leaky bucket congestion control scheme for high speed networks. Simulation results show that parameters such as the window size and the token generation rate in the leaky bucket are key factors affecting the end-to-end delay.

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Reliable Asynchronous Image Transfer Protocol In Wireless Multimedia Sensor Network (무선 멀티미디어 센서 네트워크에서의 신뢰성 있는 비동기적 이미지 전송 프로토콜)

  • Lee, Joa-Hyoung;Seon, Ju-Ho;Jung, In-Bum
    • The KIPS Transactions:PartC
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    • v.15C no.4
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    • pp.281-288
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    • 2008
  • Recently, the advance of multimedia hardware has fostered the development of wireless multimedia sensor network which is able to ubiquitously obtain multimedia content such as image or audio from the environment. The multimedia data which has several characteristics such as large size and correlation between the data requires reliability in transmission. However, the existing solution which take the focus on the efficiency of network mainly, is not appropriate to transmit the multimedia data. In the paper, we proposes a reliable asynchronous image transfer protocol, RAIT. RAIT applies double sliding window method in node-to-node image tansfer to prevent the packet loss caused by network congestion. The double sliding window consists of one sliding window for the receiving queue, which is used for prevention of packet loss caused by communication failure between nodes and the other sliding window for the sending queue which prevents the packet loss caused by network congestion. the routing node prevents the packet loss and guarantees the fairness between the nodes by scheduling the packets based on the image non-preemptively. The RAIT implements the double sliding window method by cross layer design between RAIT layer, routing layer, and queue layer. The experiment shows that RAIT guarantees the reliability of image transmission compared with the existing protocol.

An Experimental Study on the Application of NTCIP to Korean Traffic Signal Control System (교통신호제어시스템 NTCIP 통신규약 적용성 실험 연구)

  • Go, Gwang-Yong;Jeong, Jun-Ha;Lee, Seung-Hwan;An, Gye-Hyeong
    • Journal of Korean Society of Transportation
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    • v.24 no.5 s.91
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    • pp.19-33
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    • 2006
  • This paper presents the results of an experimental study on the application of NTCIP protocol to Korean traffic signal control system. For this study the communication Protocol of the existing traffic signal control system was adjusted to meet NTCIP standard. Management information base for Korea real-time traffic signal control system, message library of OER, traffic control center management software supporting SNMP/SFMP Protocol, and agent softwares for local controllers were developed during the experimental study. The applicability test of the adjusted system by NTCIP standard was performed. Fifty eight Percent of communication packets were lost at 2.400bps communication speed, which made the operation impossible. The experimentations with communication speeds 4,800bps and 9,600bps did not cause problems. In conclusion, to apply the NTCIP standard to domestic real-time traffic control system, communication environments need to be upgraded to 4,800bps or higher.

Design and Implementation of MPEG-4 Streaming System with Prioritized Adaptive Transport (우선순위화 기반 적응형 전송 기능을 가진 MPEG-4 스트리밍 시스템의 설계 및 구현)

  • 박상훈;장혜영;권영우;김종원;유웅식;권오형
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.29 no.8A
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    • pp.859-867
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    • 2004
  • To provide high-quality media streaming service over the best-effort Internet, a streaming methodology is required to response to the dynamic fluctuation of underlying networks. In this paper, we implement the MPEG-4 streaming system with adaptive transport based on priorities of media packets. The implemented system is composed of the common MPEG-4 streaming components such as elementary stream provider, sync and DMIF layer, and adaptive transport module including data prioritization and FEC control. More specifically, the prioritized sync layer packets (based on object level) are delivered to the transport module and then are encoded by an adaptive FEC encoder to help reliable transport. The FEC combination is dynamically adjusted by the feedback information from the receiver. In addition, low priority packets are selectively dropped to meet the limitation of available bandwidth. The experimental results over the emulation-based testbed show that the Proposed system can mitigate the impact of network fluctuation and thus improve the quality of streaming.

Preceding Error Recovery Algorithm for Multimedia Stream in the Tree-based Multicast Environments (트리기반 멀티캐스트 환경에서 멀티미디어 스트림을 위한 선행에러복구 방안)

  • Kim, Ki-Young;Yoon, Mi-Youn;Shin, Young-Tae
    • The KIPS Transactions:PartC
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    • v.10C no.3
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    • pp.345-354
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    • 2003
  • IP Multicast is required of more little network resources than one in unicast. Furthermore, reliable multicast has been researched for supporting reliability at IP Multicast mechanism. Although these studies are carried out, they only have focused on general data. In other words, in case that realtime packet, they can not support reliability since they do not consider realtime properties such as dependency of interframe and playback in time. Besides, we also request to support scalability because we are based on Mobile IP network together with internet. Thus, we need a mechanism to guarantee reliability and scalability of realtime stream data. In this paper, we propose PER (Preceding Error Recovery) that reflect characteristics of the realtime data, especially for H.323. PER provides scalable reliability because it is based on tree-based multicast basically and helps to support scalable relibility as reducing control packet and recovers stream buffer space from underflow status as soon as possible. PER shows much better scalable and reliable than existing works.

Implementation of TCP Retransmitted Packet Loss Recovery using ns-2 Simulator (ns-2 시뮬레이터를 이용한 TCP 재전송 손실 복구 알고리듬의 구현)

  • Kim, Beom-Joon
    • The Journal of the Korea institute of electronic communication sciences
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    • v.7 no.4
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    • pp.741-746
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    • 2012
  • Transmission control protocol(TCP) widely used as a transport protocol in the Internet includes a loss recovery function that detects and recovers packet losses by retransmissions. The loss recovery function consists of the two algorithms; fast retransmit and fast recovery. There have been researches to avoid nonnecessary retransmission timeouts (RTOs), which leads to selective acknowledgement (SACK) option and limited transmit scheme that are standardized by IETF (Internet Engineering Task Force). Recently, a method that covers the case in which a retransmitted packet is lost again has been propsed. The method, however, is not proved in terms of the additive increase multiplicative decrease (AIMD) principle of TCP congestion control. In this paper, therefore, we analyzed the method in terms of the principle by ns-simulations.

A Steady State Analysis of TCP Rate Control Mechanism on Packet loss Environment (전송 에러를 고려한 TCP 트래픽 폭주제어 해석)

  • Kim, Dong-Whee
    • Journal of Korea Society of Industrial Information Systems
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    • v.22 no.1
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    • pp.33-40
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    • 2017
  • In this Paper, Analyse the Steady State Behavior of TCP and TFRC with Packet Error when both TCP and TFRC Flows Co-exist in the Network. First, Model the Network with TCP and TFRC Connections as a Discrete Time System. Second, Calculate Average Round Trip Time of the Packet Between Source and Destination on Packet Loss Environment. Then Derive the Steady State Performance i.e. Throughput of TCP and TFRC, and Average Buffer Size of RED Router Based on the Analytic Network Model. The Throughput of TCP and TFRC Connection Decrease Rapidly with the Growth of Sending Window Size and Their Transmission Rate but Their Declines become Smoothly when the Number of Sending Window Arrives on Threshold Value. The Average Queue Length of RED Router Increases Slowly on Low Transmission Rate but Increases Rapidly on High Transmission Rate.

A Secondary MAP Scheme for Decreasing a Handover Delay and Packet Loss in an HMIPv6 (HMIPv6에서 핸드오버 지연 및 패킷 손실 감소를 위한 2차 MAP 이용 기법)

  • Jang Seong Sik;Lee Won Yeoul;Park Sun Young;Byun Tae Young;Han Ki Jun
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.42 no.2 s.332
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    • pp.39-48
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    • 2005
  • An HMIPv6 provides micro mobility management using MAP for decreasing handover delay and network load in a mobile IP networks. An HMIPv6 uses distance based algorithm for MAP selection when a mobile host enters a new network domain. However, since every mobile hosts select a farthest router as a MAP, a handover delay and packet loss will be increased. A new MAP selection scheme is herein proposed to solve the problems caused by the distance based MAP selection algorithm by using secondary MAP. We executed the performance evaluation by simulation about handover delay and packet loss of an HMIPv6 and our proposed scheme. The simulation results show that the performance of our proposed scheme is better than that of HMIPv6.