• 한국통신학회논문지
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• 제30권6B호
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• pp.396-405
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• 2005

Snoop 프로토콜은 유무선이 혼재된 망에서 무선 링크에서 발생하는 TCP 패킷 손실을 효과적으로 보상하여 TCP 처리율(throughput)을 향상시킬 수 있는 효율적인 프로토콜이다. 하지만, 무선 링크에서 연집한(burst) 패킷 손실이 발생하는 경우에는 지역 재전송을 효과적으로 수행하지 못하여 효율이 떨어진다는 문제점이 있다. 본 논문에서는 이러한 Snoop 프로토콜의 단점을 개선한 Enhanced Snoop(E-Snoop) 프로토콜을 제안한다. E-Snoop 프로토콜은 Snoop 프로토콜과 같이 중복 ACK 패킷 수신과 지역 재전송 타이머 만료에 의해 무선 링크에서의 패킷 손실을 인지할 수 있을 뿐만 아니라, new ACK 패킷 수신을 통해서도 패킷 손실을 인식할 수 있도록 설계되었다. 따라서, 무선 링크상의 연속한 패킷 손실을 빨리 인지하고 신속한 지역 재전송을 수행함으로써 TCP 처리율을 향상시킬 수 있다. 컴퓨터 시뮬레이션 결과 E-Snoop 프로토콜은 기존의 Snoop 프로토콜보다 TCP 처리율을 더 효율적으로 향상시킬 수 있음을 확인할 수 있었고, 특히 패킷 손실율이 높은 무선 링크에서 더 높은 성능 향상을 얻을 수 있었다.

• Journal of information and communication convergence engineering
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• 제9권5호
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• pp.530-538
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• 2011

Performance of TCP can be severely degraded in Mobile IP-based wireless networks where packet losses not related to network congestion occur frequently during inter-subnetwork handoffs by user mobility. To solve such a problem in the networks using Mobile IP, the packet buffering method at a base station(BS) recovers those packets dropped during handoff by forwarding the buffered packets at the old BS to the mobile users. But, when the mobile user moves to a congested BS in a new foreign subnetwork, those buffered packets forwarded by the old BS are dropped and TCP transmission performance of a mobile user degrades severely. In this paper, we propose a Modified BLUE(MBLUE) buffer required at a BS to increase TCP throughput in Mobile IP-based networks. When a queue length exceed a threshold and congestion grows, MBLUE increases its packet drop probability. But, when a TCP connection is added at new BS by a handoff, the old BS marks the buffered packets. And new BS receives the marked packets without dropping. Simulation results show that MBLUE buffer reduces congestion during handoffs and increases TCP throughputs.

• 대한음성학회지:말소리
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• 제57호
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• pp.175-189
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• 2006

Because a packet loss brings about degradation of speech quality, VoIP speech coders have PLC (Packet Loss Concealment) mechanism. G.711, which is a mandatory VoIP speech coder, also has the PLC algorithm based on pitch period replication. However, it is not robust to burst losses. Thus, we propose two methods to improve the performance of the original PLC algorithm in G.711. One adaptively utilizes voiced/unvoiced information of adjacent good frames regarding to the current lost frame. The other is based on adaptive gain control according to energy variation across the frames. We evaluate the performance of the proposed PLC algorithm by measuring a PESQ value under different random and burst packet loss simulating conditions. It is shown from the experiments that the performance of the proposed PLC algorithm outperforms that of PLC employed in ITU-T Recommendation G.711.

• 한국통신학회논문지
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• 제19권1호
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• pp.66-74
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• 1994

고속의 패킷 교환망에서 최적 경로 제어 문제(ORP: Optical Routing Poblem)를 해결하기 위해 gradient projection (GP) 기법을 적용하였따. 망 평균 패킷의 손실율을 최소화하는 ORP는 교환 노드들에서 패킷들이 손실됨으로써 non-convex가 되어, GP 알고리즘에 의해서는 그 경로해를 직접 구할 수가 없다. 따라서, 이 non-convex ORP를 reduced ORP (R-ORP)라 불리는 convex 문제로 변형시킨후, 그 경로해를 구하고자 GP 알고리즘을 이용하였다. 컴퓨터 모의 실험에 의해서는 R-ORP를 통해 얻어진 경로해가 원래 ORP의 최적 경로해에 아주 근사함을 보여준다. 또한 ORP 와 R-ORP들을 통해 얻어진 두 경로해간의 차이에 있어서의 이론적인 상한이 유도된다.

• KSII Transactions on Internet and Information Systems (TIIS)
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• 제13권12호
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• pp.5948-5971
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• 2019

Estimating loss rates with the packet traces captured from some point in the middle of the network has received much attention within the research community. Meanwhile, existing intermediate-point methods like [1] require the capturing system to capture all the TCP traffic that crosses the border of an access network (typically Gigabit network) destined to or coming from the Internet. However, limited to the performance of current hardware and software, capturing network traffic in a Gigabit environment is still a challenging task. The uncaptured packets will affect the total number of captured packets and the estimated number of packet losses, which eventually affects the accuracy of the estimated loss rate. Therefore, to obtain more accurate loss rate, a method of strengthening packet loss measurement from the network intermediate point is proposed in this paper. Through constructing a series of heuristic rules and leveraging the binomial distribution principle, the proposed method realizes the compensation for the estimated loss rate. Also, experiment results show that although there is no increase in the proportion of accurate estimates, the compensation makes the majority of estimates closer to the accurate ones.

• Journal of Communications and Networks
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• 제4권4호
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• pp.266-281
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• 2002

The General Packet Radio Service (GPRS) is being deployed by GSM network operators world-wide, and promises to provide users with “always-on” data access at bandwidths comparable to that of conventional fixed-wire telephone modems. However, many users have found the reality to be rather different, experiencing very disappointing performance when, for example, browsing the web over GPRS. In this paper, we examine the causes, and show how unfortunate interactions between the GPRS link characteristics and TCP/IP protocols lead to poor performance. A performance characterization of the GPRS link-layer is presented, determined through extensive measurements taken over production networks. We present measurements of packet loss rates, bandwidth availability, link stability, and round-trip time. The effect these characteristics have on TCP behavior are examined, demonstrating how they can result in poor link utilization, excessive packet queueing, and slow recovery from packet losses. Further, we show that the HTTP protocol can compound these issues, leading to dire WWW performance. We go on to show how the use of a transparent proxy interposed near the wired-wireless border can be used to alleviate many of these performance issues without requiring changes to either client or server end systems.

• 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.

• 대한전기학회:학술대회논문집
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• 대한전기학회 2004년도 학술대회 논문집 정보 및 제어부문
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• pp.86-88
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• 2004

As the rapid incensement of the number of internet users has occurred recently, many multimedia application services have been emerging. To improve quality of service, traffic can be suggested to be classified with priority in EPON(Ethernet Passive Optical Network), which is supporting the multimedia application services. In this paper, multimedia application services treat bandwidth classifying device in serving both delay sensitive traffic for real-time audio, video and voice data such as VoIP(Voice over Internet Protocol), and nonreal-time traffic such as BE(Best Effort). With looking through existing mechanisms, new mechanism to improve the quality will be suggested. The delay performances and packet losses of traffic achieved by supporting bandwidth allocation of upstream traffic in suggested mechanism will be analyzed with simulation.

• 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.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2006년도 하계종합학술대회
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• pp.59-60
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• 2006

TCP applied to the wireless-wired integrated network is the one that was applied to the existing wired network. In the wireless-wired integrated network, both wireless and congestion loss can occur. When wireless packet losses occur, the congestion control of TCP causes performance degradation by reducing its transmission rate. In this paper, we propose the algorithm to distinguish the wireless packet loss from congestion packet loss using MIB of the 802.11 MAC which has been generally used recently in wireless links.