• Journal of Communications and Networks
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• 제6권3호
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• pp.235-244
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• 2004

There have been a lot of approaches to evaluate and predict transmission control protocol (TCP) performance in a numerical way. Especially, under the recent advance in wireless transmission technology, the issue of TCP performance over wireless links has come to surface. It is because TCP responds to all packet losses by invoking congestion control and avoidance algorithms, resulting in degraded end-to-end performance in wireless and lossy systems. By several previous works, although it has been already proved that overall TCP performance is largely dependent on its loss recovery performance, there have been few works to try to analyze TCP loss recovery performance with thoroughness. In this paper, therefore, we focus on analyzing TCP's loss recovery performance and have developed a simple model that facilitates to capture the TCP sender's behaviors during loss recovery period. Based on the developed model, we can derive the conditions that packet losses may be recovered without retransmission timeout (RTO). Especially, we have found that TCP Reno can retransmit three packet losses by fast retransmits in a specific situation. In addition, we have proved that successive three packet losses and more than four packet losses in a window always invoke RTO easily, which is not considered or approximated in the previous works. Through probabilistic works with the conditions derived, the loss recovery performance of TCP Reno can be quantified in terms of the number of packet losses in a window.

• 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.

• 한국통신학회논문지
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• 제29권7B호
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• pp.660-666
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• 2004

처리율로 대표되는 전반적인 TCP의 성능은 패킷 손실이 발생했을 때 이를 복구하는 과정의 효율성에 의해 크게 좌우된다. 특히, 무선 링크를 통한 전송 과정에서의 비트 오류로 인해 발생하는 비 혼잡 패킷 손실은 TCP의 손실 복구 성능을 크게 저하시키는 문제점이 있다. 본 논문에서는 무선 채널에 존재하는 다중 경로 페이딩(multipath fading)에 의해서 상호 연관성을 가지고 연속적으로 발생하는 패킷 손실(correlated packet Losses)에 대해 TCP 손실 복구 과정의 성능을 모델링을 통해서 분석한다. Markov 프로세스를 이용한 분석 결과를 바탕으로 전체적인 패킷 손실 확률이 매우 낮더라도 패킷 손실의 연속성이 손실 복구 차원에서의 TCP의 성능을 크게 저하시킬 수 있음을 보인다.

• 한국통신학회논문지
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• 제28권4B호
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• pp.288-297
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• 2003

TCP Reno 는 하나의 윈도우 내에서 다수 개의 패킷 손실이 발생하는 경우 손실된 패킷들을 효율적으로 복구하는 것이 불가능한 문제점을 가지과 있다. 이 문제점을 개선하기 위해서 설계된 TCP New-Reno 는 부분 스인 패킷(partial acknowledgement)를 통해 fast recovery를 연장함으로써 다수 개의 패킷 손실이 발생하더라도 이들을 재전송에 의해 복구하는 것이 가능하다. 그러나 TCP New-Reno 역시 재전송 패킷들이 다시 손질되는 경우 불가피한 RTO(Retransmission Timeout)가 발생한다는 문제점을 가지고 있다. 이런 문제점을 개선하기 위해서 중복 승인 패킷 수를 근거로 재전송 패킷 손실을 감지할 수 있는 DAC(Duplicate Acknowledgement Counting) 알고리듬을 제안한다. TCP Reno. TCP New-Reno 그리고 DAC를 사용하는 경우에 대해 손실 복구 과정을 정확하게 모델링하고 손실된 패킷이 복구되기 위한 조건들을 유도한다. 임의 패킷 손실 확률에 대한 손실 복구 확률을 수학적으로 계산하고 이를 통해 DAC가 TCP New-Reno 의 손실 복구 기능을 향상시킬 수 있다는 것을 보인다.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제2권2호
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• pp.82-102
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• 2008

Packet losses tend to occur during short error bursts separated by long periods of relatively error-free transmission. There is also a significant spatial correlation in loss among the receiver nodes in a multicast session. To recover packet transmission errors at the transport layer, tree-based protocols construct a logical tree for error recovery before data transmission is started. The current tree construction scheme does not scale well because it overloads the sender node. We propose a scalable recovery tree construction scheme considering these properties. Unlike the existing tree construction schemes, our scheme distributes some tasks normally handled by the sender node to specific nodes acting as repair node distributors. It also allows receiver nodes to adaptively re-select their repair node when they experience unacceptable error recovery delay. Simulation results show that our scheme constructs the logical tree with reduced message and time overhead. Our analysis also indicates that it provides fast error recovery, since it can reduce the number of additional retransmissions from its upstream repair nodes or sender node.

• 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.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제6권4호
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• pp.1006-1025
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• 2012

Improving the quality of service in IP networks is a major challenge for real-time voice communications. In particular, packet arrival-delay variation, so-called "jitter," is one of the main factors that degrade the quality of voice in mobile devices with the voice-over Internet protocol (VoIP). To resolve this issue, a receiver-based enhanced timing recovery algorithm combined with active jitter estimation is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. Extensive simulations have shown that the proposed algorithm delivers high voice quality by pursuing an optimal trade-off between average buffering delay and packet loss rate.

• Journal of Ubiquitous Convergence Technology
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• 제1권1호
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• pp.28-34
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• 2007

Tree-based reliable multicast protocols provide scalability by distributing error-recovery tasks among several repair nodes. These repair nodes perform local error recovery for their receiver nodes using the data stored in their buffers. We propose a packet loss patterns adaptive feedback scheduling scheme to manage these buffers in an efficient manner. Under our scheme, receiver nodes send NAKs to repair nodes to request packet retransmissions only when the packet losses are independent events from other nodes. At dynamic and infrequent intervals, they also send ACKs to indicate which packets can be safely discarded from the repair node's buffer. Our scheme reduces delay in error recovery because the requested packets are almost always available in the repair node's buffers. It also reduces the repair node's workload because (a) each receiver node sends infrequent ACKs with non-fixed intervals and (b) their sending times are fairly distributed among all the receiver nodes.

• 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.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2006년도 하계종합학술대회
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• pp.353-354
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• 2006

This paper suggests the packet loss recovery method to communicate in real-time in the Internet. To reduce the effects of packet loss, Forward Error Correction(FEC) that adds redundant information to voice packets can be used. The major cause for speech quality degradation in IP-networks is packet loss. So, We recovered single lossy packet by using FEC method and concealed continued errors. The proposed scheme is evaluated in the Gilbert Internet channel model. The high quality of audio maintained up to 30% packet loss.