• Journal of the Korea Society of Computer and Information
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• v.19 no.3
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• pp.45-54
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• 2014

In this paper, an improved two-step P2P traffic classification scheme is proposed to overcome the limitations of the existing methods. The first step is a signature-based classifier at the packet-level. The second step consists of pattern heuristic rules and a statistics-based classifier at the flow-level. With pattern heuristic rules, the accuracy can be improved and the amount of traffic to be classified by statistics-based classifier can be reduced. Based on the analysis of different decision tree algorithms, the statistics-based classifier is implemented with REPTree. In addition, the ensemble algorithm is used to improve the performance of statistics-based classifier Through the verification with the real datasets, it is shown that our hybrid scheme provides higher accuracy and lower overhead compared to other existing schemes.

• Journal of the Korea Society of Computer and Information
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• v.26 no.6
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• pp.73-80
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• 2021

In this paper, we propose a packet processing method capable of handling large-capacity traffic over 20Gbps using multi-core and huge page memory approaches. As ICT technology advances, the global average monthly traffic is expected to reach 396 exabytes by 2022. With the increase in network traffic, cyber threats are also increasing, increasing the importance of traffic analysis. Traffic analyzed as an existing high-cost foreign product simply stores statistical data and visually shows it. Network administrators introduce and analyze many traffic analysis systems to analyze traffic in various sections, but they cannot check the aggregated traffic of the entire network. In addition, since most of the existing equipment is of the 10Gbps class, it cannot handle the increasing traffic every year at a fast speed. In this paper, as a method of processing large-capacity traffic over 20Gbps, the process of processing raw packets without copying from single-core and basic SMA memory approaches to high-performance packet reception, packet detection, and statistics using multi-core and NUMA memory approaches suggest When using the proposed method, it was confirmed that more than 50% of the traffic was processed compared to the existing equipment.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.5
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• pp.995-1003
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• 2004

A number of Internet application services are used with the development of Internet backbone nowadays. Well-known services such as WWW, ]n, email are provided at first time. Tremendous unwell-known services are presented according to the demands of various contents. After analyzing PDU information of the packet using unwell-known port travelling on the internet, searching internet service type and its statistical data is provided with internet traffic analyst as very useful information. This paper presents the mechanism to extract the internet application services operated on (un)well-known port of UDP or TCP used occasionally through netflow and tcpdump method introduced by ethereal and the operation scheme of the service. Afterwards to get the detailed statistics of the analyzed application service, the agent and the server environment, the agent gathering raw data traffics and the server adapting the traffic received from the agent BNF(Backus-Naur Form) method, is also introduced. Adapting the presented mechanism eve. LAN of Andong national university, the internet traffic service type and the detailed statistics of the analyzed application services which provides with internet traffic analyst are presented as very useful information.

• ETRI Journal
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• v.22 no.4
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• pp.1-12
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• 2000

Designing a reliable multicast-based network that scales to the size of a multicast group member is difficult because of the diversity of user demands. The loss inferences of internal nodes by end-to-end measurements do not require the use of complete statistics because of the use of maximum likelihood estimation. These schemes are very efficient and the inferred value converges fast to its true value. In the theoretical analysis, internal delay estimation is possible but the analysis is very complex due to the continuity property of the delay. In this paper, we propose the use of a bottleneck location estimator. This can overcome the analytical difficulty of the delay estimation using the power spectrum of the packet interarrival time as the performance metric. Both theoretical analysis and simulation results show that the proposed scheme can be used for bottleneck location inference of internal links in scalable multicasting.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.3 no.5
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• pp.475-491
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• 2009

This paper is primarily concerned with multi-path routing in Mobile Ad hoc Networks (MANETs). We propose a novel associativity-based on-demand source routing protocol for MANETs that attempts to establish relatively stable path(s) between the source and the destination. We introduce a new notion for gauging the temporal and spatial stability of nodes, and hence the paths interconnecting them. The proposed protocol is compared with other unipath (DSDV and AODV) and multi-path (AOMDV) routing protocols. We investigate the performance in terms of throughput, normalized routing overhead, packet delivery ratio etc. All on-demand protocols show good performance in mobile environments with less traffic overhead compared to proactive approaches, but they are prone to longer end-to-end delays due to route discovery and maintenance.

• ETRI Journal
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• v.34 no.4
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• pp.645-648
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• 2012

An incremental subspace learning scheme to recover lost speech segments online is presented. Our contributions in this work are twofold. First, the recovery problem is transformed into an interpolation problem of the time-varying gains via nonnegative matrix factorization. Second, incremental nonnegative matrix factorization is employed to allow online processing and track the evolution of speech statistics. The effectiveness of the proposed scheme is confirmed by the experiment results.

• Journal of Communications and Networks
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• v.6 no.2
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• pp.133-146
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• 2004

Recently, many active queue management (AQM) algorithms have been proposed to address the performance degradation. of end-to-end congestion control under tail-drop (TD) queue management at Internet routers. However, these AQM algorithms show performance improvement only for limited network environments, and are insensitive to dynamically changing network situations. In this paper, we propose an adaptive queue management algorithm, called PID-controller, that uses proportional-integral-derivative (PID) feedback control to remedy these weak-Dalles of existing AQM proposals. The PID-controller is able to detect and control congestion adaptively and proactively to dynamically changing network environments using incipient as well as current congestion indications. A simulation study over a wide range of IP traffic conditions shows that PID-controller outperforms other AQM algorithms such as Random Early Detection (RED) [3] and Proportional-Integral (PI) controller [9] in terms of queue length dynamics, packet loss rates, and link utilization.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.12
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• pp.9-17
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• 2003

In this paper we proposea method for estimating the bandwidth in next-generation If network. Especially, we concentrate on the edge routers accommodating the VoIP connections as well as a group of data connections. Bandwidth dimensioning is carried out at call level and packet level for voice traffic in the next-generation IP network. The model incorporates the statistical estimation approach at a call level for obtaining the number of voice connections simultaneously in the active mode. The call level model incorporates a statistical technique to compute the statistics of the number of active connections such as the mean and variance of the simultaneously connected calls in the network. The packet level model represents a load map for voice and data traffic by using non-preemptive M/G/1 queuing model with strict priority for voice over data buffer, From the proposed traffic model, we can derive a graph for upper bounds on the traffic load in terms of bandwidth for voice and data connections. Via numerical experiments we illustrate the implication of the work.

• Journal of Broadcast Engineering
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• v.14 no.5
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• pp.573-588
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• 2009

In this paper, we propose a novel video quality degradation monitoring scheme titled VR-VQMS(Visual Rhythm based Video Quality Monitoring Scheme) over an IPTV service prone to packet losses during network transmission. Proposed scheme quantifies the amount of quality degradation due to packet losses, and can be classified into a RR(reduced-reference) based quality measurement scheme exploiting visual rhythm data of H.264-encoded video frames at a media server and reconstructed ones at an Set-top Box as feature information. Two scenarios, On-line and Off-line VR-VQMS, are proposed as the practical solutions. We define the NPSNR(Networked Peak-to-peak Signal-to-Noise Ratio) modified by the well-known PSNR as a new objective quality metric, and several additional objective and subjective metrics based on it to obtain the statistics on timing, duration, occurrence, and amount of quality degradation. Simulation results show that the proposed method closely approximates the results from 2D video frames and gives good estimation of subjective quality(i.e.,MOS(mean opinion score)) performed by 10 test observers. We expect that the proposed scheme can play a role as a practical solution to monitor the video quality experienced by individual customers in a commercial IPTV service, and be implemented as a small and light agent program running on a resource-limited set-top box.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.45 no.6
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• pp.34-51
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• 2008

Today, the Internet is a part of our life. For that reason, we regard revealing characteristics of Internet traffic as an important research theme. However, Internet traffic cannot be easily manipulated because it usually occupy huge capacity. This problem is a serious obstacle to analyze Internet traffic. Many researchers use various sampling techniques to reduce capacity of Internet traffic. In this paper, we compare several famous sampling techniques, and propose efficient sampling scheme. We chose some sampling techniques such as Systematic Sampling, Simple Random Sampling and Stratified Sampling with some sampling intensities such as 1/10, 1/100 and 1/1000. Our observation focused on Traffic Volume, Entropy Analysis and Packet Size Analysis. Both the simple random sampling and the count-based systematic sampling is proper to general case. On the other hand, time-based systematic sampling exhibits relatively bad results. The stratified sampling on Transport Layer Protocols, e.g.. TCP, UDP and so on, shows superior results. Our analysis results suggest that efficient sampling techniques satisfactorily maintain variation of traffic stream according to time change. The entropy analysis endures various sampling techniques well and fits detecting anomalous traffic. We found that a traffic volume diminishment caused by bottleneck could induce wrong results on the entropy analysis. We discovered that Packet Size Distribution perfectly tolerate any packet sampling techniques and intensities.