• Title/Summary/Keyword: Packet Loss

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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.

A Packet Loss Concealment Algorithm Based on Multiple Adaptive Codebooks Using Comfort Noise (Comfort Noise를 이용한 다중 적응 코드북 기반 패킷 손실 은닉 알고리즘)

  • Park, Nam-In;Kim, Hong-Kook
    • Proceedings of the IEEK Conference
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    • 2008.06a
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    • pp.873-874
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    • 2008
  • In this paper, we propose a packet loss concealment (PLC) algorithm for CELP speech coders, which is based on multiple adaptive codebooks by using comfort noise for the lost packet recovery. The multiple adaptive codebooks are composed of a conventional adaptive codebook to model periodic excitation of speech and another adaptive codebook to provide a better estimate of excitation when packets are lost in the speech onset region. The performance of the proposed PLC algorithm is evaluated by implementing it into the G.729 decoder and compared with that of the PLC algorithm employed in the G.729 decoder by means of perceptual evaluation of speech quality (PESQ). It is shown from the experiments under different burstiness of packet loss rates of 3% and 5% that the proposed PLC algorithm provides higher PESQ scores than the G.729 PLC algorithm.

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A Study of Performance Improvement of Two Dimensional FEC Schemes For Data Security (데이터보안을 위한 2차원 FEC기법의 성능 향상에 관한 연구)

  • Min, Sun-Ho
    • Journal of the Korea Institute of Information Security & Cryptology
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    • v.23 no.5
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    • pp.957-962
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    • 2013
  • This paper proposes the new enhanced 2-D(2-Dimension) FEC scheme. It analyzes the probability of entire packet loss rate of the existing 2-D FEC by mathematical modeling, finds the problem of the existing 2-D FEC, and deduces the new enhanced 2-D FEC scheme that reduces the entire packet loss probability.

Modeling and Analysis of Burst Switching for Wireless Packet Data (무선 패킷 데이터를 위한 Burst switching의 모델링 및 분석)

  • Park, Kyoung-In;Lee, Chae Young
    • Journal of Korean Institute of Industrial Engineers
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    • v.28 no.2
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    • pp.139-146
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    • 2002
  • The third generation mobile communication needs to provide multimedia service with increased data rates. Thus an efficient allocation of radio and network resources is very important. This paper models the 'burst switching' as an efficient radio resource allocation scheme and the performance is compared to the circuit and packet switching. In burst switching, radio resource is allocated to a call for the duration of data bursts rather than an entire session or a single packet as in the case of circuit and packet switching. After a stream of data burst, if a packet does not arrive during timer2 value ($\tau_{2}$), the channel of physical layer is released and the call stays in suspended state. Again if a packet does not arrive for timerl value ($\tau_{1}$) in the suspended state, the upper layer is also released. Thus the two timer values to minimize the sum of access delay and queuing delay need to be determined. In this paper, we focus on the decision of $\tau_{2}$ which minimizes the access and queueing delay with the assumption that traffic arrivals follow Poison process. The simulation, however, is performed with Pareto distribution which well describes the bursty traffic. The computational results show that the delay and the packet loss probability by the burst switching is dramatically reduced compared to the packet switching.

A Handover Mechanism for QoS Guarantee in WiBro (초고속 휴대 인터넷 망에서 서비스 품질 보장을 위한 핸드오버 메커니즘)

  • Yeom Hong-Ju;Kim Hwa-Sung
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.31 no.7A
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    • pp.659-665
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    • 2006
  • When using the Mobile IP which is the representative technology to secure the mobility in general IP networks, the packet loss during the handover is inescapable. To remedy the packet loss problem, the smooth handover was introduced. However, the smooth handover causes the packets sequence disruption during the packet forwarding procedure and it may result in the degradation of the network performance. The same problem also occurs in the WiBro (High-speed Portable Internet) system that is the next generation portable IP service system. The WiBro system, which provides the high speed data service just like xDSL and leased line in wired internet, aims to guarantee the portability, mobility, and the differentiated service based on IEEE 802.16. So, the handover mechanisms that solve the problems of packet loss and packet sequence distribution are required in the WiBro system. In this paper, we propose the handover mechanism and the packet sequence control algorithm that provide the reliability and the differentiated service for the unicast service in the WiBro system.

BS-PLC(Both Side-Packet Loss Concealment) for CELP Coder (CELP 부호화기를 위한 양방향 패킷 손실 은닉 알고리즘)

  • Lee In-Sung;Hwang Jeong-Joon;Jeong Gyu-Hyeok
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.42 no.12
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    • pp.127-134
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    • 2005
  • Lost packet robustness is an most important quality measure for voice over IP networks(VoIP). Recovery of the lost packet from the received information is crucial to realize this robustness. So, this paper proposes the lost packet recovery method from the received information for real-time communication for CELP coder. The proposed BS-PLC (Both Side Packet Loss Concealment) based WSOLA(Waveform Shift OverLab Add) allow the lost packet to be recovered from both the 'previous' and 'next' good packet as the LP parameter and the excitation signal are respectively recovered. The burst of packet loss is modeled by Gilbert model. The proposed scheme is applied to G.729 most used in VoIP and is evaluated through the SNR(signal to noise) and the MOS(Mean Opinion Score) test. As a simulation result, The proposed scheme provide 0.3 higher in Mean Opinion Score and 2 dB higher in terms of SNR than an error concealment procedure in the decoder of G.729 at $20\%$ average packet loss rate.

Closed-loop Feedback Control for Enhancing QoS in Real-time communication Networks

  • Kim, Hyung-Seok;Kwon, Wook-Hyun
    • 제어로봇시스템학회:학술대회논문집
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    • 2001.10a
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    • pp.40.1-40
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    • 2001
  • In this paper, control theoretic approaches are proposed to guarantee QoS (Quality of Series) such as packet delay and packet loss of real-time traffic in high-speed communication network. Characteristics of variable rate real-time traÆc in communication networks are described. The mathematical model describing networks including source and destination nodes are suggested. By a traffic control mechanism, it is shown that worst-case end-to-end transfer delay of traffic can be controlled and packet loss can be prevented. The simulation shows results of delay control and buer level control to raise QoS in realtime traffic.

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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.

Threshold-based Filtering Buffer Management Scheme in a Shared Buffer Packet Switch

  • Yang, Jui-Pin;Liang, Ming-Cheng;Chu, Yuan-Sun
    • Journal of Communications and Networks
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    • v.5 no.1
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    • pp.82-89
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    • 2003
  • In this paper, an efficient threshold-based filtering (TF) buffer management scheme is proposed. The TF is capable of minimizing the overall loss performance and improving the fairness of buffer usage in a shared buffer packet switch. The TF consists of two mechanisms. One mechanism is to classify the output ports as sctive or inactive by comparing their queue lengths with a dedicated buffer allocation factor. The other mechanism is to filter the arrival packets of inactive output ports when the total queue length exceeds a threshold value. A theoretical queuing model of TF is formulated and resolved for the overall packet loss probability. Computer simulations are used to compare the overall loss performance of TF, dynamic threshold (DT), static threshold (ST) and pushout (PO). We find that TF scheme is more robust against dynamic traffic variations than DT and ST. Also, although the over-all loss performance between TF and PO are close to each other, the implementation of TF is much simpler than the PO.

Packet Loss Concealment Algorithm Based on Speech Characteristics (음성신호의 특성을 고려한 패킷 손실 은닉 알고리즘)

  • Yoon Sung-Wan;Kang Hong-Goo;Youn Dae-Hee
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.31 no.7C
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    • pp.691-699
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    • 2006
  • Despite of the in-depth effort to cantrol the variability in IP networks, quality of service (QoS) is still not guaranteed in the IP networks. Thus, it is necessary to deal with the audible artifacts caused by packet lasses. To overcame the packet loss problem, most speech coding standard have their own embedded packet loss concealment (PLC) algorithms which adapt extrapolation methods utilizing the dependency on adjacent frames. Since many low bit rate CELP coders use predictive schemes for increasing coding efficiency, however, error propagation occurs even if single packet is lost. In this paper, we propose an efficient PLC algorithm with consideration about the speech characteristics of lost frames. To design an efficient PLC algorithm, we perform several experiments on investigating the error propagation effect of lost frames of a predictive coder. And then, we summarize the impact of packet loss to the speech characteristics and analyze the importance of the encoded parameters depending on each speech classes. From the result of the experiments, we propose a new PLC algorithm that mainly focuses on reducing the error propagation time. Experimental results show that the performance is much higher than conventional extrapolation methods over various frame erasure rate (FER) conditions. Especially the difference is remarkable in high FER condition.