• 한국통신학회논문지
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• 제24권10B호
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• pp.1824-1831
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• 1999

본 논문에서는 연속 패킷 손실 환경에서 systematic erasure code를 적용하였을 경우의 손실 복구율에 관하여 분석하고 손실 특성에 따른 부가 전송량의 추정방법에 대하여 제시한다. 연속 패킷 손실환경은 Gilbert 모델로 설정하였고, 기존의 연속 손실 환경에서의 erasure code의 손실 복구율 분석방안을 이용하여 systematic erasure code를 사용하였을 경우의 성능을 분석하고, 평균 패킷 손실율, 손실의 평균 길이 등의 주어진 패킷 손실 특성으로부터 주어진 손실 특성을 만족시키는 부가 전송량의 추정 방법을 제시한다.

• 정보보호학회논문지
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• 제26권6호
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• pp.1571-1582
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• 2016

단방향 전송 기술에서 해결해야 할 이슈들 중에 한 가지는 TCP 기반의 데이터 전송에서 발생하는 패킷 손실을 줄이는 것이다. 잘 알려진 에러 수정 기법들을 활용해서 패킷 손실을 줄일 수 있다. 하지만, 기존의 여러 기법들을 활용한다고 하더라도, 링크 에러와 버퍼 오버플로우에 의한 패킷 손실은 여전히 발생할 수 있다. 본 논문에서는 신뢰성 있는 단방향 데이터 전송 시스템(RED, REliable Data diode)을 제안한다. RED는 기존의 단방향 전송 기술과 마찬가지로 TCP 기반의 데이터 전송을 지원하기 위해 TCP 프록시 기법을 활용한다. RED 송신시스템은 TCP 패킷의 지연 전송을 활용하여 버퍼 오버플로우에 의한 패킷 손실을 줄일수 있다. 또한, RED 송신시스템은 패킷의 중요도 및 여유 자원을 고려하여, RED 수신시스템에게 다수의 동일한 패킷을 복제 및 전송함으로써, 단방향 전송 링크에서의 링크 에러에 의한 패킷 손실을 줄일 수 있다.

• ETRI Journal
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• 제38권6호
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• pp.1064-1073
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• 2016

An adaptive speech streaming method to improve the perceived speech quality of a software-based multipoint control unit (SW-based MCU) over IP networks is proposed. First, the proposed method predicts whether the speech packet to be transmitted is lost. To this end, the proposed method learns the pattern of packet losses in the IP network, and then predicts the loss of the packet to be transmitted over that IP network. The proposed method classifies the speech signal into different classes of silence, unvoiced, speech onset, or voiced frame. Based on the results of packet loss prediction and speech classification, the proposed method determines the proper amount and bitrate of redundant speech data (RSD) that are sent with primary speech data (PSD) in order to assist the speech decoder to restore the speech signals of lost packets. Specifically, when a packet is predicted to be lost, the amount and bitrate of the RSD must be increased through a reduction in the bitrate of the PSD. The effectiveness of the proposed method for learning the packet loss pattern and assigning a different speech coding rate is then demonstrated using a support vector machine and adaptive multirate-narrowband, respectively. The results show that as compared with conventional methods that restore lost speech signals, the proposed method remarkably improves the perceived speech quality of an SW-based MCU under various packet loss conditions in an IP network.

• 대한산업공학회지
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• 제28권2호
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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.

• ETRI Journal
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• 제25권5호
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• pp.356-368
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• 2003

This paper analyzes the characteristics of packet losses in mobile computing environments based on the Gilbert model and then describes a mechanism that can recover the lost audio packets using redundant data. Using information periodically reported by a receiver, the sender dynamically adjusts the amount and offset values of redundant data with the constraint of minimizing the bandwidth consumption of wireless links. Since mobile computing environments can be often characterized by frequent and consecutive packet losses, loss recovery mechanism need to deal efficiently with both random and consecutive packet losses. To achieve this, the suggested mechanism uses relatively large, discontinuous exponential offset values. That gives the same effect as using both the sequential and interleaving redundant information. To verify the effectiveness of the mechanism, we extended and implemented RTP/RTCP and applications. The experimental results show that our mechanism, with an exponential offset, achieves a remarkably low complete packet loss rate and adapts dynamically to the fluctuation of the packet loss pattern in mobile computing environments.

• 대한임베디드공학회논문지
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• 제6권3호
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• pp.149-156
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• 2011

Transmission delay and packet loss induced by a communication network can degrade the control performance and, even make the system unstable. This paper presents a method for compensating transmission delay and packet loss in a networked control system for unmanned underwater vehicle. The proposed method is based on Lagrange interpolation in order to satisfy the requirements of simplicity and model-independency. In this work, the lost/delayed data are estimated in real time by only using the past data without requiring any mathematical model of the controlled system. Consequently, the proposed method can be implemented independent of the controlled system, and also it can achieve fast and accurate compensation performance. The performance of the proposed technique is evaluated by numerical simulations with an unmanned underwater vehicle.

• Journal of information and communication convergence engineering
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• 제9권5호
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• pp.551-555
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• 2011

Optical packet switching is a promising technology, which can integrate both data and optical network. In this paper, we present a comparative study of various traffic arrival approaches in WDM packet switches. The comparison is made based on packet loss rate and average delay under uniform and self-similar Pareto traffic. Computer simulations are performed in order to obtain the switch performance metrics. Study shows that burstiness of data traffic has a strong negative impact in the performance of WDM packet switches.

• 전자공학회논문지S
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• 제35S권7호
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• pp.1-5
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• 1998

In this paper, recovery method with high-bitrate and low-bitrate coder was implemented in order to recover consecutive packet loss over the Internet. LPC was used as redundant audio data for recover of lost packets and RTP parcket format was modified for accommodation of redundant data. In measuring results using random packet loss rate with three redundant datra in every packet, it has shown that recovery rate was 80% in los rate of 50%. Since the processing delay for recovery of the lost packet was 200ms, this recovery method can be applied to real-time Internet sevice such as Internet phone.

• Journal of Communications and Networks
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• 제16권3호
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• pp.322-334
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• 2014

Multimedia content is very sensitive to packet loss and therefore multimedia streams are typically protected against packet loss, either by supporting retransmission requests or by adding redundant forward error correction (FEC) data. However, the redundant FEC information introduces significant additional bandwidth requirements, as compared to the bitrate of the original video stream. Especially on wireless and mobile networks, bandwidth availability is limited and variable. In this article, an adaptive FEC (A-FEC) system is presented whereby the redundancy rate is dynamically adjusted to the packet loss, based on feedback messages from the client. We present a statistical model of our A-FEC system and validate the proposed system under different packet loss conditions and loss probabilities. The experimental results show that 57-95%bandwidth gain can be achieved compared with a static FEC approach.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제12권7호
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• pp.3150-3171
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• 2018

In this paper, a heuristic method that leverages packet traces captured at the entire boarder of an ISP to distinguish and estimate the overall packet loss within an ISP's management domain (Intra_Path_Loss) and that in the outside Internet (Inter_Path_Loss) is proposed. Our method is inspired by that packet losses happened at different locations will cause different TCP sequence number patterns at the border of an ISP. Thereby, we leverage these TCP sequence number patterns to build a series of heuristic rules to estimate Intra_Path_Loss and Inter_Path_Loss, respectively. We do this work with an eye towards showing that the overall packet losses defined and estimated in this paper can provide the operators with some valuable information to help them precisely grasp the overall performance of network paths and narrow down the range of network anomalies. The proposed method is rigorously validated with simulations, and finally the results from a regional academic network JSERNET verify its effectiveness and practicability.