• 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.

• Proceedings of the KAIS Fall Conference
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• 2007.11a
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• pp.198-200
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• 2007

본 논문에서는 블록 기반으로 압축된 이미지의 전송방법에서 EREC 기법들의 패킷 손실 내성을 평가하고 패킷 손실 환경에서 현존하는 EREC의 성능을 비교한다. 이 논문에서는 기존의 EREC 기법들이 임의의 비트에 대한 에러 내성을 제공하는 것과는 대조적으로 패킷 손실 환경에서 EREC방법의 검증을 그 목적으로 한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.10C
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• pp.950-958
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• 2007

This paper presents an efficient packet loss resilient scheme for real-time video transmission over the Internet. By analyzing the temporal and spatial dependencies in inter- and intra-frames, we assign forward error correction codes (FEC) across video packets for minimizing the effect of error concealment and error propagation from packet loss. To achieve optimal allocation of FEC codes, we formulate the effect of packet loss on video quality degradation as packet distortion model. Then we propose an unequal FEC assignment scheme with low complexity based on packet correction rate, which uses the packet distortion model and includes channel status information. Simulation results show that the proposed FEC assignment scheme gives substantial improvement for the received video quality in packet lossy networks. Furthermore the proposed scheme achieves relatively smaller degradation of video quality with higher packet loss rates.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1997.11a
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• pp.19-22
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• 1997

음성이나 동화상 같은 실시간 정보를 인터넷상으로 전송하는 인터넷방송 시스템은 인터넷 서비스의 중요한 이슈중에 하나지만 구현시에 가장 커다란 문제가 되는 것은 패킷 손실이다. 인터넷라디오나 AOD에서 발생하는 손실된 패킷을 복구하기 위해 여러 가지 방법이 제안되고 있으나, 현재 인터넷폰 등을 위해서는 잉여 오디오 정보를 이용한 패킷 복구 알고리즘이 가장 효과적으로 사용되고 있으나 단방향 통신이면서 음질이 가장 중요한 요소인 방송시스템에서는 그대로 적용하기에 문제가 있다. 본 논문에서는 잉여 오디오 정보를 이용한 패킷 복구 알고리즘에 사용될 수 있는 인터리빙 방법을 제안하였다. 연구결과, 인터리빙 처리를 위하여 추가적인 지연이 발생하는 단점은 있지만 연집 에러를 효과적으로 분산시켜 패킷 손실 복구율을 크게 개선시킬 수 있다.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.218-223
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• 2008

In this paper, we implement several methods to improve an audio collaboration system over IP networks, and then evaluate the performance of the implemented methods. In general, speech and audio quality degrades depending on the characteristics of IP networks such as jitter and packet loss. In order to reduce this quality degradation, we propose a lower bit rate audio delivery scheme using the MPEG-2 AAC (Advanced Audio Coding) audio codec in a viewpoint that a packet loss rate could be reduced by a smaller packet size. In addition, iLBC (Internet Low-Bitrate Codec) and the G.711 packet loss concealment algorithm defined by IEFT and ITU-T, respectively, are applied to a audio collaboration system. RAT (Robust-Audio Tool)[7] is used as a baseline platform for the implementation of the proposed methods. It is shown from the implementation that the implemented MPEG-2 AAC audio codec with a bitrate of 256 kbit/s performs as similar as the uncompressed audio quality with a bitrate of 512 kbit/s, and that iLBC and the G.711 packet loss concealment algorithm can improve speech quality when a packet loss rate is 2~10%.

• The Journal of the Acoustical Society of Korea
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• v.32 no.2
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• pp.157-166
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• 2013

This paper proposes a non-intrusive speech quality estimation method considering the effects of packet loss to perceptual quality. Packet loss is a major reason of quality degradation in a packet based speech communications network, whose effects are different according to the input speech characteristics or the performance of the embedded packet loss concealment (PLC) algorithm. For the quality estimation system that involves packet loss effects, we first observe the packet loss of G.729 codec which is one of narrowband codec in VoIP system. In order to quantify the lost packet affects, we design a classification algorithm only using speech parameters of G.729 decoder. Then, the degradation values of each class are iteratively selected that maximizes the correlation with the degradation PESQ-LQ scores, and total quality degradation is modeled by the weighted sum. From analyzing the correlation measures, we obtained correlation values of 0.8950 for the intrusive model and 0.8911 for the non-intrusive method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.7B
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• pp.477-487
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• 2008

Congestion control of the TCP reduces transmission rate when it detects packet loss because packet loss origines from congestion in the wired network. In the wireless network, packet loss comes from channel errors. Wired TCP degrades performance when there are wireless losses because it does not classify type of loss. These day, there are many researches which classify type of loss between congestion loss and wireless loss for wired-wireless hybrid network. For wireless TCP, many of existing algorithms are based on the estimated bandwidth or variations of packet arrival time. In this paper, we propose a new TCP scheme to distinguish the wireless packet losses from the congestion packet losses using MIB of the IEEE 802.11 MAC. We perform excessive simulations using the NS-2 network simulator and analyze the simulation results to compare the performance of the proposed algorithm to other well-known algorithms. From simulation results, we know that proposed algorithm improves performance about 12% and 32% compared with Spike algorithm and mBiaz algorithm, respectively.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1999.11b
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• pp.225-228
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• 1999

Network의 QoS를 결정하는 인자에는 손실과 지연, jitter 등이 있다. 그 중에서 손실의 양을 최소화하기 위하여 패킷의 크기를 될 수 있는 한 작게 보냄으로서 라우터에서 생기는 손실을 최소화하는 방법이 있다. 본 연구에서는 비디오 데이터의 전송에 있어서 패킷의 크기를 현재 망의 상황에 고려하여 여러 개의 소켓으로 나누어 보냄으로서 손실의 양을 최소화한다.

• Proceedings of the Korean Information Science Society Conference
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• 2000.10c
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• pp.514-516
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• 2000

현재 가장 널리 쓰이는 수송계층 프로토콜인 TCP는 패킷 손실의 원인을 망의 혼잡 때문에 일어난다고 가정하고 있으므로 기존의 유선망과 고정 호스트로 이루어진 전통적인 네트워크에 적합하다. 그러나 무선 링크에서의 패킷 손실은 대부분 혼잡에 의해서가 아니라 높은 에러율과 핸드오프에 의해 발생하게 되므로 기존의 TCP를 그대로 사용하면 불필요한 혼잡제어 메커니즘의 호출로 성능의 저하를 가져온다. 현재까지 무선환경에 적합한 TCP를 위한 많은 방안이 제시되고 있지만 근본적인 해결책을 제시하지 못하고 있다. 본 논문에서 제안하는 기법은 패킷손실이 유선링크에서 일어나는 것인지, 무선링크에서 일어나는 것인지를 판별하여 패킷손실이 유선링크에서 일어난 경우는 기존의 혼잡제어 메커니즘을 호출하여 재전송하고, 무선링크에서 일어난 경우는 혼잡제어 메커니즘을 호출하지 않고 재전송 하여 성능을 개선한다.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.11 no.6
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• pp.89-94
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• 2012

In this paper, we considered the performances of PMIPv6 and DMM in the viewpoints of traffic load change and packet delivery latency time, where a new buffering in the previous MAAR is proposed to reduce packet loss during handover. To show the superiority of the DMM and to validate the operation of the buffering scheme, we accomplished its simulation under the typical handover. Our performance comparison results show that the DMM is better than the performance of traffic load change and packet delivery latency time of PMIPv6. We can see that the proposed buffering scheme is better than the existing one in terms of packet loss.