• Proceedings of the Korean Information Science Society Conference
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• 2001.10c
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• pp.343-345
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• 2001

통신 및 네트워크 사용자의 증가와 실시간 처리를 요하는 멀티미디어 데이터 서비스의 증가의 인하여 네트워크 트래픽이 증가하고 서비스 품질 저하 및 접속 속도의 감소가 초래되고 있다 이러한 문제를 해결하기 위하여 라우터의 용량이 확장되고 있으나, 라우터의 용량확장에 따른 테이블의 크기 증가는 경로 선택을 위한 검색 시간 증가를 초래하므로, 라우팅 테이블의 구성방식에 대한 연구가 필요하다. 본 논문은 TCP/IP 기반의 라우팅 프로토콜을 관리하는 Zebra 소프트웨어를 이용하여, 라우팅 테이블의 구조와 구성방식에 대하여 분석하고, 비교 알고리즘을 제시하여 성능 비교 및 평가를 하고자 한다.

• The KIPS Transactions:PartA
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• v.16A no.2
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• pp.113-124
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• 2009

With increasing multicore system, much effort has been put on the performance improvement of its application. Because multicore system has multiple processing devices in one system, its processing power increases compared to the single core system. However in many cases the advantages of multicore can not be exploited fully because the existing software and hardware were designed to be suitable for single core. When the existing software runs on multicore, its performance improvement is limited by the bottleneck of sharing resources and the inefficient use of cache memory on multicore. Therefore, according as the number of core increases, it doesn't show performance improvement and shows performance drop in the worst case. In this paper we propose a method of performance improvement of multicore system by applying Flow-Level Parallelism to the existing TCP/IP network application and operating system. The proposed method sets up the execution environment so that each core unit operates independently as much as possible in network application, TCP/IP stack on operating system, device driver, and network interface. Moreover it distributes network traffics to each core unit through L2 switch. The proposed method allows to minimize the sharing of application data, data structure, socket, device driver, and network interface between each core. Also it allows to minimize the competition among cores to take resources and increase the hit ratio of cache. We implemented the proposed methods with 8 core system and performed experiment. Experimental results show that network access speed and bandwidth increase linearly according to the number of core.

• KSCI Review
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• v.14 no.2
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• pp.235-244
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• 2006

This paper attacked the unknown DoS which mixed a DoS attack, Worm and the Trojan horse which used IP Source Address Spoofing and Smurf through the SYN Flooding way that UDP, ICMP, Echo, TCP Syn packet operated. the applications that used TCP/UDP in VoIP service networks. Define necessity of a Dynamic Update Engine for a prevention, and measure Miss traffic at RT statistics of inbound and outbound parts in case of designs of an engine at IPS regarding an Self-learning module and a statistical attack spread. and design a logic engine module. Three engines judge attack grades (Attack Suspicious, Normal), and keep the most suitable filtering engine state through AND or OR algorithms at Footprint Lookup modules. A Real-Time Dynamic Engine and Filter updated protected VoIP service from DoS attacks, and strengthened Ubiquitous Security anger, and were turned out to be.

• Journal of the Korea Society of Computer and Information
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• v.11 no.6 s.44
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• pp.165-174
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• 2006

This paper attacked the unknown DoS which mixed a DoS attack, Worm and the Trojan horse which used IP Source Address Spoofing and Smurf through the SYN Flooding way that UDP, ICMP, Echo, TCP Syn packet operated, the applications that used TCP/UDP in VoIP service networks. Define necessity of a Dynamic Update Engine for a prevention, and measure Miss traffic at RT statistics of inbound and outbound parts in case of designs of an engine at IPS regarding an Self-learning module and a statistical attack spread, and design a logic engine module. Three engines judge attack grades (Attack, Suspicious, Normal), and keep the most suitable filtering engine state through AND or OR algorithms at Footprint Lookup modules. A Real-Time Dynamic Engine and Filter updated protected VoIP service from DoS attacks, and strengthened Ubiquitous Security anger, and were turned out to be.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.2
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• pp.104-113
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• 2009

Recently, high-speed Internet users and broadband network infrastructure are increasing. However, the TCP protocol widely used in the Internet is an appropriate transport only for the legacy Internet, it is inefficient for traffic transport for network environments with large bandwidth-delay product. In order to remedy this problem, there have been many researches about improved congestion control algorithms for broadband networks. In these studies, most of performance evaluation has been done by simulations. In this paper, after we reviewed the proposed high bandwidth transport layer protocols, we implemented a real testbed, measured the performance and analyzed the problems of high-speed transport protocols through the international research network TEIN2 between three nations, Korea, China, and Germany.

• Journal of KIISE:Information Networking
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• v.30 no.5
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• pp.595-603
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• 2003

To provide mobile nodes with continuous communication services, it is important to reduce the packet losses during handoffs. The handoffs of mobile nodes cause packet losses and decrease of TCP throughput on account of a variety of factors. One of those is the congestion in the new cell. Due to the congestion, not only the node moving into the cell but also the already existing nodes that were successfully communicating in the cell suffer the performance degradation. In this paper we propose a new handoff mechanism called‘packet freeze control’, which avoids the congestion caused by handoffs by regulating the influx of traffic burst into the new cell. Packet freeze control is applicable to a wireless network domain in which FAs(Foreign Agents) are connected hierarchically and constitute a logical tree. It gradually increases the number of packets transferred to the new cell by buffering packets in the FAs on the packet delivery path over the wireless network domain. The simulation results show that the proposed mechanism not only reduces the packet losses but also enhances the TCP throughput of other mobile nodes in the cell.

• The KIPS Transactions:PartC
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• v.9C no.4
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• pp.605-614
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• 2002

Amount of the web traffic web server handles are explosively increasing, which requires that the performance of the web server should be improved for the various web services. Although the analysis for the HTTP traffic with the proper tuning for the web server is essential, the research relevant to the subject are insignificant. In particular, although most of applications are implemented over HTTP 1.1 protocol, the researches mostly deal with the performance evaluation of the HTTP 1.0 protocol. Consequently, the modeling approach and the performance evaluation over HTTP 1.1 protocol have not been well formed. Therefore, basing on the HTTP 1.1 protocol supporting persistent connection, we present an analytical end-to-end tandem queueing model for web server to consider the specific hardware configuration inside web server beginning at accepting the user request until completing the service. we compare various performances between HTTP 1.0 and HTTP 1.1 under the overloading condition, and then analyze the characteristics of the HTTP traffic that include file size requested to web server, the OFF time between file transfers, the frequency of requests, and the temporal locality of requests. Presented model is verified through the comparing the server throughput according to varying requests rate with the real web server. Thereafter, we analyze the performance evaluation of the web server, according to the interrelation between TCP Listen queue size, the number of HTTP threads and the size of the network buffers.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.9B
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• pp.823-833
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• 2003

To satisfy the requirements of the real-time Internet services, queue management and scheduling schemes should be enhanced to accommodate the delay and jitter characteristic of them. Although the existing queue management schemes can address the congestion problems of TCP flows, they have some problems in supporting real-time services. That is, they show performance degradation when burst traffics are continuously going into the system after the queue is occupied at a predefined threshold level. In addition, under the congestion state, they show large jitter, which is not a desirable phenomenon for real-time transmissions. To resolve these problems, we propose a SDRED (State Dependent Random Early Detection) and dynamic scheduling scheme that can improve delay and jitter performances by adjusting RED parameters such as ma $x_{th}$ and $w_{q}$ according to the queue status. The SDRED is designed to adapt to the current traffic situation by adjusting the max,$_{th}$ and $w_{q}$ to four different levels. From the simulation results, we show that the SDRED decreases packet delays in a queue and has more stable jitter characteristics than the existing RED, BLUE, ARED and DSRED schemes.mes.mes.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.6
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• pp.75-84
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• 2014

In order to solve the problems such as overload, single point of failure, non-optimized data path, and network scalability in conventional central mobility management protocols, distributed mobility management schemes have been continually studied in and around the IETF. In this paper, a network-based partially-distributed mobility management mechanism, pDMMv6, is suggested and the performance comparison with traditional protocols such as PMIPv6 and MIPv6 is made through simulation under the various user traffic environment. The simulation results include UDP packet delivery ratio, end-to-end packet delay, binding delay for registration signaling, CPU utilization in each node, and response delays in several server-client TCP applications such as web browsing, e-mail, telnet remote login, FTP file up/down-load, and database access.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.17-24
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• 2005

Traditional Internet applications such as FIP and E-mail are increasingly sharing bandwidth with newer, more demanding applications such as Web browsing, IP telephony, video conference and online games. These new applications require Quality of Service (QoS), in terms of delay, loss and throughput that are different from QoS requirements of traditional applications. Unfortunately, current Active Queue Management (AQM) approaches offer monolithic best-effort service to all Internet applications regardless of the current QoS requirements. This paper proposes and evaluates a new AQM technique, called MCDT that provides dynamic and separated buffer threshold for each Applications, those are FTP and e-mail on TCP traffic, streaming services on tagged UDP traffic, and the other services on untagged UDP traffic. Using a new QoS metric, our simulations demonstrate that MCDT yields higher QoS in terms of the delay variation and a packet loss than RED when there are heavy UDP traffics that include streaming applications and data applications. MCDT fits the current best-effort Internet environment without high complexity.