• Title/Summary/Keyword: Packet Service Time

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Dynamic Load-Balancing Algorithm Incorporating Flow Distributions and Service Levels for an AOPS Node

  • Zhang, Fuding;Zhou, Xu;Sun, Xiaohan
    • Journal of the Optical Society of Korea
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    • v.18 no.5
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    • pp.466-471
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    • 2014
  • An asynchronous optical packet-switching (AOPS) node with load-balancing capability can achieve better performance in reducing the high packet-loss ratio (PLR) and time delay caused by unbalanced traffic. This paper proposes a novel dynamic load-balancing algorithm for an AOPS node with limited buffer and without wavelength converters, and considering the data flow distribution and service levels. By calculating the occupancy state of the output ports, load state of the input ports, and priorities for data flow, the traffic is balanced accordingly. Simulations demonstrate that asynchronous variant data packets and output traffic can be automatically balanced according to service levels and the data flow distribution. A PLR of less than 0.01% can be achieved, as well as an average time delay of less than 0.46 ns.

Congestion Control Mechanism for Efficient Network Environment in WMSN (무선 멀티미디어 센서 네트워크에서 효율적인 네트워크 환경을 위한 혼잡 제어 메커니즘)

  • Park, Jeong-Hyeon;Lee, Sung-Keun;Oh, Won-Geun
    • The Journal of the Korea institute of electronic communication sciences
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    • v.10 no.2
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    • pp.289-296
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    • 2015
  • Wireless multimedia sensor network senses and transfers mass multimedia data. Also, it is sensitive to latency. This thesis proposes a routing technique based on traffic priority in order to improve the network efficiency by minimizing latency. In addition, it proposes a congestion control mechanism that uses packet service time, packet inter-arrival time, buffer usage, etc. In this thesis, we verified the reduction of packet latency in accordance with the quality level of packet as a result of the performance analysis through the simulation method. Also, we verified that the proposed mechanism maintained a reliable network state by preventing packet loss due to network overload.

Fuzzy-based Dynamic Packet Scheduling Algorithm for Multimedia Cognitive Radios (멀티미디어 무선인지 시스템을 위한 퍼지 기반의 동적 패킷 스케줄링 알고리즘)

  • Tung, Nguyen Thanh;Koo, In-Soo
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.12 no.3
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    • pp.1-7
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    • 2012
  • Cognitive radio, a new paradigm for wireless communication, is being recently expected to support various types of multimedia traffics. To guarantee Quality of Service (QoS) from SUs, a static packet priority policy can be considered. However, this approach can easily satisfy Quality of Service of high priority application while that of lower priority applications is being degraded. In the paper, we propose a fuzzy-based dynamic packet scheduling algorithm to support multimedia traffics in which the dynamic packet scheduler modifies priorities of packets according to Fuzzy-rules with the information of priority and delay deadline of each packet, and determines which packet would be transmitted through the channel of the primary user in the next time slot in order to reduce packet loss rate. Our simulation result shows that packet loss rate can be improved through the proposed scheme when overall traffic load is not heavy.

Exploiting Packet Semantics in Real-time Multimedia Streaming

  • Hong, Sung-Woo;Won, You-Jip
    • Proceedings of the Korean Society of Broadcast Engineers Conference
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    • 2009.01a
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    • pp.118-123
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    • 2009
  • In this paper, we propose packet selection and significance based interval allocation algorithm for real-time streaming service. In real-time streaming of inter-frame (and layer) coded video, minimizing packet loss does not imply maximizing QoS. It is true that packet loss adversely affects the QoS but one single packet can have more impact than several other packets. We exploit the fact that the significance of each packet loss is different from the frame type it belongs to and its position within GoP. Using packet dependency and PSNR degradation value imposed on the video from the corresponding packet loss, we find each packet's significance value. Based on the packet significance, the proposed algorithm determines which packets to send and when to send them. The proposed algorithm is tested using publicly available MPEG-4 video traces. Our scheduling algorithm brings significant improvement on user perceivable QoS. We foresee that the proposed algorithm manifests itself in last mile connection of the network where intervals between successive packets from the source and to the destination are well preserved.

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A Construction and Operation Analysis of Group Management Network about Control Devices based on CIM Level 3 (CIM 계층 3에서 제어 기기들의 그룹 관리 네트워크 구축과 운영 해석)

  • 김정호
    • The Journal of Society for e-Business Studies
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    • v.4 no.1
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    • pp.87-101
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    • 1999
  • To operate the automatic devices of manufacturing process more effectively and to solve the needs of the resource sharing, network technology is applied to the control devices located in common manufacturing zone and operated by connecting them. In this paper, functional standard of the network layers are set as physical and data link layer of IEEE 802.2, 802.4, and VMD application layer and ISO-CIM reference model. Then, they are divided as minimized architecture, designed as group objects which perform group management and service objects which organizes and operates the group. For the stability in this network, this paper measures the variation of data packet length and node number and analyzes the variated value of the waiting time for the network operation. For the method of the analysis, non-exhausted service method are selected, and the arrival rates of the each data packet to the nodes that are assumed to form a Poission distribution. Then, queue model is set as M/G/1, and the analysis equation for waiting time is found. For the evalution of the performance, the length of the data packet varies from 10 bytes to 100 bytes in the operation of the group management network, the variation of the wating time is less than 10 msec. Since the waiting time in this case is less than 10 msec, response time is fast enough. Furthermore, to evaluate the real time processing of the group management network, it shows if the number of nodes is less than 40, and the average arrival time is less than 40 packet/sec, it can perform stable operation even taking the overhead such as software delay time, indicated packet service, and transmissin safety margin.

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Bayes Rule for MAC State Sojourn Time Supporting Packet Data Service in CDMA Wireless Celluar Networks

  • Park, Cheon-Won;Kim, Dong-Joon;Shin, Woo-Cheol;Ju, Jee-Hwan
    • Proceedings of the IEEK Conference
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    • 2002.07c
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    • pp.1606-1609
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    • 2002
  • MAC state models appeared with an effort to overcome technical demerits of CDMA in provisioning packet data service. In the scenario of sojourn and transition on MAC states, the design of state sojourn time is a critical issue for an efficient utilization of limited recource; a longer sojourn time leads to more resource being preserved for inactive stations, while more connection components should be recovered with a shorter sojourn time. Thus, the sojourn time at each MAC state must be optimized in consideration of these two conflicting arguments. In this paper, we first present a generic MAC state model. Secondly, based on the generic model, we reveal a relation of inactive period and the delay time of the last packet served in pre- ceding active period and specify a loss function reflect-ing two antinomic features that result from a change of state sojourn time. Using the proposed loss function, we construct a decision problem to find an optima3 rule for state sojourn times. Finally, we present a way of computing Bayes rule by use of the posterior distribution of inactivity duration for given observation on the delay time of last packet. Furthermore, Bayes rules are explicitly expressed for special arrival processes and investigated with respect to traffic load and loss parameters.

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Space and Time Priority Queues with Randomized Push-Out Scheme (확률적 밀어내기 정책을 가지는 공간-시간 우선순위 대기행렬)

  • Kilhwan Kim
    • Journal of Korean Society of Industrial and Systems Engineering
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    • v.46 no.2
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    • pp.57-71
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    • 2023
  • In this study, we analyze a finite-buffer M/G/1 queueing model with randomized pushout space priority and nonpreemptive time priority. Space and time priority queueing models have been extensively studied to analyze the performance of communication systems serving different types of traffic simultaneously: one type is sensitive to packet delay, and the other is sensitive to packet loss. However, these models have limitations. Some models assume that packet transmission times follow exponential distributions, which is not always realistic. Other models use general distributions for packet transmission times, but their space priority rules are too rigid, making it difficult to fine-tune service performance for different types of traffic. Our proposed model addresses these limitations and is more suitable for analyzing communication systems that handle different types of traffic with general packet length distributions. For the proposed queueing model, we first derive the distribution of the number of packets in the system when the transmission of each packet is completed, and we then obtain packet loss probabilities and the expected number of packets for each type of traffic. We also present a numerical example to explore the effect of a system parameter, the pushout probability, on system performance for different packet transmission time distributions.

An Analysis Tool for Deriving Bounds on Delay and Buffer Size in Packet Networks (패킷 네트워크에서 지연과 버퍼 크기 한계를 추출할 수 있는 분석 도구)

  • 편기현;송준화;이흥규
    • Journal of KIISE:Information Networking
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    • v.30 no.5
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    • pp.641-648
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    • 2003
  • In this paper, we propose a general analysis tool that derives guaranteed delay bounds for real-time sessions when those sessions pass through heterogeneous schedulers in general packet networks. This tool characterizes each different scheduler by a homogeneous service curve server. We show that service curve servers can characterize a strictly larger class of schedulers than LR servers. That is, we can characterize not only all LR servers but also RC-EDF and SC scheduler by service curve servers. Even with such advantage over LR servers, service curve servers provide accurate analysis results. We prove mathematically that if we analyze a network that can be analyzed by both LR servers and service curve servers,,both cases yield the same delay bound.

Packet Scheduling Algorithm Considering Maximum Delay Tolerance for HSDPA System

  • Hur, Soojung;Jakhongil, Narzullaev;Park, Yong-Wan
    • IEMEK Journal of Embedded Systems and Applications
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    • v.8 no.6
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    • pp.311-318
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    • 2013
  • In this paper, we consider a new packet scheduling algorithm for real-time traffic in the HSDPA system that has been introduced for the WCDMA system, in order to provide high transmission rates. The objective of the design is to meet the maximum tolerable delay and consider channel assignment based on the received SIR for real-time traffic users. The proposed scheduling algorithm shows that the users are ranked by the ratios of the bits in the buffer to the residual time for transmission as priority order; then the ranked users are assigned certain number of channels based on the SIR value table. The simulation results show that the proposed algorithm can provide a lower packet drop rate, and satisfy real time quality of service (QoS) requirements.

FEAPT Algorithm to compensate Jitter in Internet Phone (인터넷전화에서 지터보상을 위한 Frame Extension for Adaptive Playout Time(FEAPT) 알고리즘)

  • 남재현
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.7 no.6
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    • pp.1168-1176
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    • 2003
  • Internet telephony service is very cheap and very easy to introduce the value-added service than the POTS, but is difficult to the QoS of telephone service. The existing internet typically offers "best effort" services only, which do not make any commitment about delay, packet loss and jitter. This paper compensates the low quality of the speech for packet loss or delay using FEAPT algorithm in Internet phone. In the FEAPT algorithm, the receiver expands the received packet under resonable threshold, and hence compensates the QoS of speech.of speech.