• Proceedings of the Korean Information Science Society Conference
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• 2003.04d
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• pp.232-234
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• 2003

TCP(Transmission Control Protocol)는 신뢰성 있는 전송계층 프로토콜이다. 이것은 데이터 스트림을 TCP 로 전달하는 응용프로그램이 전체 스트림을 순서에 맞고 오류 없이 전달하는 것을 의미한다. TCP 는 오류 제어를 이용하여 신뢰성을 제공하는데, 오류제어는 손상 세그먼트, 손실 세그먼트, 순서가 어긋난 세그먼트, 그리고 중복 세그먼트를 감지하는 메커니즘이 포함되며 특히 타이머(timer)를 이용한 오류제어를 본 내용에서 설명하고 있다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.11A
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• pp.867-875
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• 2003

Current end-to-end congestion control depends only on the information of end points (using three duplicate ACK packets) and generally responds slowly to the network congestion. This mechanism can't avoid TCP global synchronization in which TCP congestion window size is fluctuated during congestion period. Furthermore, if RTT(Round Trip Time) is increased, three duplicate ACK packets are not correct congestion signals because congestion might already disappear and the host may send more packets until it receives three duplicate ACK packets. Recently there are increasing interests in solving end-to-end congestion control using AQM(Active Queue Management) to improve the performance of TCP protocols. AQM is a variation of RED-based congestion control. In this paper, we first evaluate the effectiveness of the current AQM schemes such as RED, CHOKe, ARED, FRED and SRED, over traffic with different rates and over traffic with mixed responsive and non-responsive flows, respectively. In particular, CHOKe mechanism shows greater unfairness, especially when more unresponsive flows exist in a shared link. We then propose a new AQM scheme using CHOKe mechanism, called DQC(Double Queue CHOKe), which uses two FIFO queues before applying CHOKe mechanism to adaptive congestion control. Simulation shows that it works well in protecting congestion-sensitive flows from congestion-causing flows and exhibits better performances than other AQM schemes. Also we use partial state information, proposed in LRURED, to improve our mechanism.

• Journal of KIISE:Information Networking
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• v.32 no.1
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• pp.12-19
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• 2005

Reliable transport protocols such as TCP are tuned to Perform well in traditional networks where packet losses occur mainly because of congestion. In a wireless network, however, packet losses will occur more often due to reasons such as the high bit error rate and the handoff rather than due to congestion. When using TCP over wireless network, TCP responds to losses due to the high bit error rate and the handoff by invoking congestion control and avoidance algorithms, resulting in the degraded end-to-end performance in the wireless network. There have been several schemes for improving TCP performance over wireless links. Among them, SNOOP Is a very promising scheme because of the localized retransmission. In this thesis, an efficient scheme is proposed by modifying SNOOP scheme. The invocation of congestion control mechanism is now minimized by knowing the cause of packet loss.

• Proceedings of the Korea Institutes of Information Security and Cryptology Conference
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• 2002.11a
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• pp.273-275
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• 2002

고도의 기술을 이용한 최근의 사이버 공격을 방어하기 위해서는 자신의 도메인만을 보호하는 현재의 수동적인 네트워크 보안 서비스보다는 액티브 네트워크 기반 하에 침입자의 위치를 역추적하고 침입자의 근원지에서 네트워크로의 접근을 차단하는 능동적인 대응이 필요하다. 본 논문에서는 TCP 기반의 우회 공격인 TCP 확장 연결 공격을 역추적하고 침입자를 네트워크로부터 고립시키는 메커니즘에 대해 기술한다.

• Proceedings of the Korean Information Science Society Conference
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• 2005.07a
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• pp.334-336
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• 2005

최근 오디오나 비디오 스트리밍과 같은 멀티미디어 트래픽이 증가하고 있다. 이러한 트래픽들은 패킷을 전달하는데 대부분 UDP(User Datagram Protocol)기반의 RTP(Realtime Transport Protocol)를 사용한다. 하지만 UDP기반의 RTP는 기본적으로 혼잡제어 메커니즘이 없으며 현재 인터넷의 주요 트래픽인 TCP(Transmission Control Protocol)와의 형평성을 보장하지 않는다는 문제점을 갖는다. 본 논문에서는 스트리밍 트래픽의 TCP 친화적인 전송률 조절 기법으로 TF-RTP(TCP-Friendly RTP)를 제안하였다. TF-RTP는 네트워크 상태가 혼잡하여 패킷 손실이 발생할 경우, 개선된 파라미터들을 사용하여 경쟁하는 TCP의 전송률을 보다 정확하게 계산하여 스트리밍 트래픽의 전송률을 조절함으로써 경쟁하는 TCP 트래픽과 친화적으로 동작하며 네트워크 대역폭을 보다 공평하게 사용하게 된다. 실험을 통해 제안한 TF-RTP가 TCP의 전송률을 보다 정확하게 계산하며 TCP 친화성, 공평성 측면에서도 성능 개선을 보임을 확인할 수 있었다.

• The KIPS Transactions:PartC
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• v.9C no.4
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• pp.513-522
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• 2002

Current end-to-end congestion control depends only on the information of end points (using three duplicate ACK packets) and generally responds slowly to the network congestion. This mechanism can't avoid TCP global synchronization which TCP congestion window size is fluctuated during congestion occurred and if RTT (Round Trip Time) is increased, three duplicate ACK packets is not a correct congestion signal because congestion maybe already disappeared and the host may send more packets until receive the three duplicate ACK packets. Recently there is increasing interest in solving end-to-end congestion control using active network frameworks to improve the performance of TCP protocols. ACC (Active congestion control) is a variation of TCP-based congestion control with queue management In addition traffic modifications nay begin at the congested router (active router) so that ACC will respond more quickly to congestion than TCP variants. The advantage of this method is that the host uses the information provided by the active routers as well as the end points in order to relieve congestion and improve throughput. In this paper, we model enhanced ACC, provide its algorithm which control the congestion by using information in core networks and communications between active routers, and finally demonstrate enhanced performance by simulation.

• Proceedings of the Korean Information Science Society Conference
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• 2004.04a
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• pp.781-783
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• 2004

최근 종단 호스트에서 나타나는 데이터 전송의 병목현상이 두드러지면서 종단 호스트에 관한 많은 연구가 진행되고 있다. 기존의 연구를 통해 종단 호스트의 병목현상의 원인으로 TCP의 비효율적인 버퍼관리가 제기되었다. 특히 임베디드 환경에서는 시스템 자원이 제한적일 수밖에 없기 때문에 TCP의 버퍼관리의 효율성이 매우 중요하다. 본 논문에서는 새로운 TCP 버퍼할당 기법인 RTBA(Rate-Based TCP Buffer Allocation) 기법을 제안하였다. RTBA는 플로우의 RTT(Round Trip Time), RTO(Retransmit Time Out), 패킷 손실율 둥을 고려한 TCP의 기대전송율을 기반으로 버퍼를 동적으로 할당함으로써 상대적으로 적은 버퍼를 사용하는 임베디드 환경에서도 다수의 TCP 플로우들이 높은 전송성능을 얻게 한다. 실험을 통해 제안한 RTBA 기법이 기존의 고정적인 버퍼할당 기법에 비해 우수한 성능을 보임을 확인할 수 있었다.

• The Journal of the Korea Contents Association
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• v.7 no.1
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• pp.131-138
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• 2007

In wired networks, packet losses mostly occur due to congestion. TCP reacts to the congestion by decreasing its congestion window, thus to reduce network utilization. In wireless networks, however, losses may occur due to the high bit-error rate of the transmission medium or due to fading and mobility. Nevertheless, TCP still reacts to packet losses according to its congestion control scheme, thus to reduce the network utilization unnecessarily. This reduction of network utilization causes the performance of TCP to decrease. In this paper, we predict packet loss by using RSS(Received Signal Strengths) on the wireless and suggest adding RSS flag bit in ACK packet of MH. By using RSS flag bit in ACK, the FH(Fixed Host) decides whether it adopt congestion control scheme or not for the maximum throughput. The result of the simulation by NS-2 shows that the proposed mechanism significantly increases sending amount and receiving amount by 40% at maximum.

• Journal of IKEEE
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• v.12 no.1
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• pp.8-17
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• 2008

Recently, GMAHN that provides interface between MANET and Wired Network has been focused in mobile communication. It is necessary that the technology provide reliable data transmission technology between mobile node and wired network in MANET environment that is varied by the node movement. In this paper, using the TCP protocol(Tahoe, Reno, Vegas, SACK)that increases reliability between source and destination, we applied the TCP protocol mechanism to various environment, and proposed the most efficient TCP mechanism by comparing each mechanism.

• The KIPS Transactions:PartC
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• v.14C no.1 s.111
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• pp.55-64
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• 2007

Behind the popularity of VoIP in these days, it may present significant security challenges in privacy and accounting. Authentication and message encryption are considered to be essential mechanisms in VoIP to be comparable to PSTN. SIP is responsible for setting up a secure call in VoIP. SIP employs TLS, DTLS or IPSec combined with TCP, UDP or SCTP as a security protocol in VoIP. These security mechanisms may introduce additional overheads into the SIP performance. However, this overhead has not been understood in detail by the community. In this paper we present the effect of the security protocol on the performance of SIP by comparing the call setup delays among security protocols. We implement a simulation of the various combinations of three security protocols and three transport layer protocols suggested for SIP. UDP with any combination of security protocols performs a lot better than the combination of TCP. TLS over SCTP may impose higher impact on the performance in average because TLS might have to open secure channels as the same number of streams in SCTP. The reasons for differences in the SIP performances are given.