• The KIPS Transactions:PartC
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• v.12C no.1 s.97
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• pp.121-128
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• 2005

Tho existing TCP(Transmission Control Protocol) is known to be unsuitable for a network with the characteristics of high RDP(Bandwidth-Delay Product) because of the fixed small or large buffer size at the TCP sender and receiver. Thus, some trial cases of adjusting the buffer sizes automatically with respect to network condition have been proposed to improve the end-to-end TCP throughput. ATBT(Automatic TCP fluffer Tuning) attempts to assure the buffer size of TCP sender according to its current congestion window size but the ATBT assumes that the buffer size of TCP receiver is maximum value that operating system defines. In DRS(Dynamic Right Sizing), by estimating the TCP arrival data of two times the amount TCP data received previously, the TCP receiver simply reserves the buffer size for the next arrival, accordingly. However, we do not need to reserve exactly two times of buffer size because of the possibility of TCP segment loss. We propose an efficient TCP buffer tuning technique(called TBT-PLR: TCP buffer tuning algorithm based on packet loss ratio) since we adopt the ATBT mechanism and the TBT-PLR mechanism for the TCP sender and the TCP receiver, respectively. For the purpose of testing the actual TCP performance, we implemented our TBT-PLR by modifying the linux kernel version 2.4.18 and evaluated the TCP performance by comparing TBT-PLR with the TCP schemes of the fixed buffer size. As a result, more balanced usage among TCP connections was obtained.

• Journal of Korea Society of Digital Industry and Information Management
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• v.18 no.4
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• pp.13-21
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• 2022

Recently, in wireless video sensor networks(WVSN), various schemes for efficient video data transmission have been studied. In this paper, a layer based cooperative relaying(LCR) algorithm is proposed for minimizing scalable video transmission distortion from packet loss in WVSN. The proposed LCR algorithm consists of two modules. In the first step, a parameter based error propagation metric is proposed to predict the effect of each scalable layer on video quality degradation at low complexity. In the second step, a layer-based cooperative relay algorithm is proposed to minimize distortion due to packet loss using the proposed error propagation metric and channel information of the video sensor node and relay node. In the experiment, the proposed algorithm showed that the improvement of peak signal-to-noise ratio (PSNR) in various channel environments, compared to the previous algorithm(Energy based Cooperative Relaying, ECR) without considering the metric of error propagation.The proposed LCR algorithm minimizes video quality degradation from packet loss using both the channel information of relaying node and the amount of layer based error propagation in scalable video.

• ETRI Journal
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• v.31 no.1
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• pp.21-30
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• 2009

In low bit-rate video transmission, the payload of a single packet can often contain a whole coded frame due to the high compression ratio in both spatial and temporal domains of most modern video coders. Thus, the loss of a single packet not only causes the loss of a whole frame, but also produces error propagation into subsequent frames. In this paper, we propose a novel whole frame error concealment algorithm which reconstructs the first of the subsequent frames instead of the current lost frame to suppress the effects of error propagation. In the proposed algorithm, we impose a constraint which uses side match distortion (SMD) and overlapped region difference (ORD) to estimate motion vectors between the target reconstructed frame and its reference frame. SMD measures the spatial smoothness connection between a block and its neighboring blocks. ORD is defined as the difference between the correlated pixels which are predicted from one reference pixel. Experimental results show that the proposed algorithm effectively suppresses error propagation and significantly outperforms other conventional techniques in terms of both peak signal-to-noise ratio performance and subjective visual quality.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.2
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• pp.579-603
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• 2018

In this paper we propose a QoS-enhanced intelligent stochastic optimal fair real-time packet scheduler, QUEST, for 4G LTE traffic in routers. The objective of this research is to maximize the system QoS subject to the constraint that the processor utilization is kept nearly at 100 percent. The QUEST has following unique advantages. First, it solves the challenging problem of starvation for low priority process - buffered streaming video and TCP based; second, it solves the major bottleneck of the scheduler Earliest Deadline First's failure at heavy loads. Finally, QUEST offers the benefit of arbitrarily pre-programming the process utilization ratio.Three classes of multimedia 4G LTE QCI traffic, conversational voice, live streaming video, buffered streaming video and TCP based applications have been considered. We analyse two most important QoS metrics, packet loss rate (PLR) and mean waiting time. All claims are supported by discrete event and Monte Carlo simulations. The simulation results show that the QUEST scheduler outperforms current state-of-the-art benchmark schedulers. The proposed scheduler offers 37 percent improvement in PLR and 23 percent improvement in mean waiting time over the best competing current scheduler Accuracy-aware EDF.

• ETRI Journal
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• v.22 no.4
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• pp.40-50
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• 2000

We apply a "sliding-window" Maximum Likelihood(ML) estimator to estimate traffic parameters On-Off source and develop a method for estimating stochastic predicted individual cell arrival rates. Based on these results, we propose a simple Connection Admission Control(CAC)scheme for delay sensitive services in broadband onboard packet switching satellite systems. The algorithms are motivated by the limited onboard satellite buffer, the large propagation delay, and low computational capabilities inherent in satellite communication systems. We develop an algorithm using the predicted individual cell loss ratio instead of using steady state cell loss ratios. We demonstrate the CAC benefits of this approach over using steady state cell loss ratios as well as predicted total cell loss ratios. We also derive the predictive saturation probability and the predictive cell loss ratio and use them to control the total number of connections. Predictive congestion control mechanisms allow a satellite network to operate in the optimum region of low delay and high throughput. This is different from the traditional reactive congestion control mechanism that allows the network to recover from the congested state. Numerical and simulation results obtained suggest that the proposed predictive scheme is a promising approach for real time CAC.

• International Journal of Computer Science & Network Security
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• v.22 no.3
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• pp.1-8
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• 2022

Supporting real-time flows with delay and throughput constraints is an important challenge for future wireless networks. In this paper, we develop an optimal scheduling scheme to optimally choose the packets to transmit. The optimal transmission strategy is based on an observable Markov decision process. The novelty of the work focuses on a priority-based probabilistic packet scheduling strategy for efficient packet transmission. This helps in providing guaranteed services to real time traffic in Heterogeneous Wireless Networks. The proposed scheduling mechanism is able to optimize the desired performance. The proposed scheduler improves the overall end-to-end delay, decreases the packet loss ratio, and reduces blocking probability even in the case of congested network.

• The Journal of the Korea institute of electronic communication sciences
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• v.8 no.1
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• pp.71-76
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• 2013

In this paper, end-to-end performances for some VoIP traffics are measured and analyzed over MANET(Mobile Ad-hoc Networks) with packet aggregation transmission under blackhole attacks. A VoIP simulator based on NS-2 is used for performance measure. In the simulation, MOS, call connection ratio, network delay and packet loss rate are measured for end-to-end transmission performance. With analysis of measured data, characteristics of each traffics for performance parameters of packet aggregation transmission is showed on MANET under blackhole attack, As a results of this paper, some considerations be required for packet aggregation transmission are suggested to implement VoIP services on MANET under blackhole attacks.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.12
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• pp.3257-3274
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• 2012

Broadcast routing has become an important research field for vehicular ad-hoc networks (VANETs) recently. However, the packet delivery rate is generally low in existing VANET broadcast routing protocols. Therefore, the design of an appropriate broadcast protocol based on the features of VANET has become a crucial part of the development of VANET. This paper analyzes the disadvantage of existing broadcast routing protocols in VANETs, and proposes an improved algorithm (namely ODAM-C) based on the ODAM (Optimized Dissemination of Alarm Messages) protocol. The ODAM-C algorithm improves the packet delivery rate by two mechanisms based on the forwarding features of ODAM. The first distance-based mechanism reduces the possibility of packet loss by considering the angles between source nodes, forwarding nodes and receiving nodes. The second mechanism increases the redundancy of forwarding nodes to guarantee the packet success delivery ratio. We show by analysis and simulations that the proposed algorithm can improve packet delivery rate for vehicular networks compared against two widely-used existing protocols.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.7
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• pp.1599-1609
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• 2014

TCP does not ensure the bandwidth and delay bound required for multimedia streaming services in broadband wireless network environments. In this paper, we propose a new congestion control scheme for efficient multimedia transmission, called COLO TCP (Concave Increase Slow Start Logarithmic Increase Congestion Avoidance TCP). The COLO TCP prevents the burst packet loss by applying the concave increase algorithm in slow start phase. In the congestion avoidance phase, COLO TCP uses the logarithmic increase algorithm that quickly recovers congestion window after packet loss. To highly utilize network bandwidth and reduce packet loss ratio, COLO TCP uses additive increase algorithm and adaptive decrease algorithm. Through simulation results, we prove that our COLO TCP is more robust for random loss. It is also possible for efficient multimedia transmission.

• Journal of Broadcast Engineering
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• v.5 no.2
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• pp.227-238
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• 2000

This paper, at first, provides analysis on loss pattern of Internet based on real experiments of the current Internet. Then, we propose an effective adaptive QoS management technique, in which measured loss pattern as well as PLR (Packet Loss Ratio) are used to select titrate of temporal scalability. level of FEC and retransmission This selection is also incorporated to the MPEG-4 error resilience tools and error concealment techniques. In order to minimize effect of packet loss, multimedia stream is segmented in the unit of group of Pictures (GOP) and interleaving and FEC are applied to the segment. Proposed algorithms are applied to build a VOD system.