• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.9
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• pp.1654-1662
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• 2006

An ad hoc network is multi-hop network composed of radio links. and the transmission quality of a radio link is more unstable than that of a wired circuit. Packet loss thus occurs frequently in an ad hoc network, and the consequent connection failure results in a severe deterioration of TCP performance. TCP performance also deteriorates because of the collision of data packets and ACK packets in the radio channel. In this paper we study to improve the performance of the Mobile Ad-hoc network by using Delayed ACK algorithm with our proposed ODA(Ordering-Delayed ACK) method. The proposed ODA algorithm increases the number of the data packets orderly at the receiver side which is going to be applied for the Mobile Ad-hoc network. We accomplished a computer simulation using NS-2. From the simulation results, we find the proposed ODA algorithm obviously enlarge the channel capacity and improve the network performance at the situation of multi-hop of ad-hoc network than the existing Delayed ACK algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.8B
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• pp.667-680
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• 2008

In this work, we present the results of our empirical study on 802.11 wireless LAN network traffic. We collect the packet trace from existing campus wireless LAN infra-structure. We analyzed four different data sets: aggregate traffic, upstream traffic, downstream traffic, tcp only packet trace from aggregate traffic. We analyze the time series aspect of underlying traffic (byte count process and packet count process), marginal distribution of time series, and packet size distribution. We found that in all four data sets there exist long-range dependent property in byte count and packet count process. Inter-arrival distribution is well fitted with Pareto distribution. Upstream traffic, i.e. from the user to Internet, exhibits significant difference in its packet size distribution from the rests. Average packet size of upstream traffic is 151.7 byte while average packet size of the rest of the data sets are all greater than 260 bytes. Packets with full data payloads constitutes 3% and 10% in upstream traffic and the downstream traffic, respectively. Despite the significant difference in packet size distribution, all four data sets have similar Hurst values. The Hurst alone does not properly explain the stochastic characteristics of the underlying traffic. We model the underlying traffic using fractional-ARIMA (FARIMA) and fractional Gaussian Noise (FGN). While the fractional Gaussian Noise based method is computationally more efficient, FARIMA exhibits superior performance in accurately modeling the underlying traffic.

• The KIPS Transactions:PartC
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• v.13C no.7 s.110
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• pp.889-896
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• 2006

ATM Forum has defined a guaranteed frame rate (GFR) service to serve Internet traffic efficiently. The GFR service provides virtual connections (VCs) for minimum cell rate (MCR) guarantees and allows them to fairly share the residual bandwidth. And ATM Forum has recommended a frame-based generic cell rate algorithm (F-GCRA) as a frame classifier, which determines whether an Am cell is eligible to use the guaranteed bandwidth in a frame level. An ATM switch accommodates cells in its buffer or drops them in a frame level according to current buffer occupancy. A FIFO shared buffer has so simple structure as to be feasibly implemented in switches, but has not been able to provide an MCR guarantee for each VC without buffer management based on per-VC accounting. In this paper, we enhance the F-GCRA frame classifier to guarantee an MCR of each VC without buffer management based on per-VC accounting. The enhanced frame classifier considers burstness of TCP traffic caused by congestion control algorithm so as to enable each VC to use its reserved bandwidth sufficiently. In addition, it is able to alleviate the unfairness problem in usage of the residual bandwidth. Simulation results show that the enhanced frame classifier satisfies quality of services (QoSs) of the GFR service for the TCP traffic.

• 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.

• Proceedings of the Korean Information Science Society Conference
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• 2002.04a
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• pp.508-510
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• 2002

인터넷 수요의 증대에 따라 오디오나 비디오 같은 실시간 전송을 요구하는 멀티미디어 데이터 응용들도 증가하게 되었다. 이런, 멀티미디어 데이터들의 전송은 패킷 손실과 지연을 최소화 하고, 대역폭 상태에 대해 TCP보다 순화된 전송률 조절이 가능한 혼잡제어 기법이 필요하다. Receiver-based 혼잡 제어방식은 network traffic의 변화에 신속히 대응할 수 있으며, 이 방식은 Sender-based 방식에 비해 멀티캐스팅 환경에 적당하다는 장점도 가진다. 본 논문에서는 인터넷 상에서 audio/video 데이터들의 종 단간 Layered Quality Adaptation을 제공하는 receiver-based 혼잡 제어 기법을 소개한다. 이를 통해 사용자들은 보다 좋은 QoS를 제공받을 수 있다.

• Annual Conference of KIPS
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• 2003.05b
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• pp.1229-1232
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• 2003

인터넷의 발달로 멀티미디어 데이터 처리가 필요하다. 이를 위해 TCP/UDP를 혼합 환경에서 사용하게 되었고 발생하는 혼잡이나 공정성의 문제를 해결하기 위해 많은 논의가 이루어지고 있다. 본 논문에서는 RED방식에서 사용하는 4개의 파라미터가 라우터 큐의 동작에 미치는 영향을 분석한다. RED 파라미터 분석을 통해서 고정된 파라미터 값을 사용할 경우 다양한 트래픽 상황에 적절히 반응하지 못하는 것을 보여준다. 또한 NS2를 이용한 시뮬레이션에서 DT방식과 RED방식을 비교하면서 공정성이 향상됨을 보인다.

• Proceedings of the Korea Multimedia Society Conference
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• 2002.05c
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• pp.385-389
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• 2002

Ad-hoc 무선망은 고정된 유선망을 가지지 않고 이동 노드들로만 구성된 망이다. 여러 가지 이동 노드에 대한 이동 제약이 없고, 유선망이나 기지국 같은 기반 구조를 필요를 하지 않기 때문에 여러 환경에 적용이 가능하다. 그러나 라우터들 사이의 이동성과 다른 연결 요소들의 가변적인 요소는 잠재적으로 속도감 있고 예측 불가능하게 변하는 망을 가져올 수 있다. 이에 따른 패킷의 손실이 무선 네트워크에서 자주 발생하게 된다. 최근 몇 년 동안 Ad-hoc망에서 사용되는 라우팅 프로토콜의 각각에 대한 성능을 분석하는 연구가 있었지만, Ad-hoc 무선망에서 특정한 프로토콜의 TCP 버전별 비교 분석에 대한 연구가 미비한 실정이다. 따라서 본 논문에서는 무선 네트워크가 가지는 단점을 보완하고 트래픽 성능을 향상시키기 위한 한 방편으로, Ad-hoc 라우팅 알고리즘인 DSR 프로토콜을 이용하여 TCP Tahoe, Sack, Reno 버전별로 토폴로지의 크기의 변화와 이동 노드의 이동 속도의 변화에 따라 트래픽의 성능을 모의 실험하여 비교 분석하였다.

• International Journal of Computer Science & Network Security
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• v.21 no.9
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• pp.132-140
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• 2021

Under the mobile ad-hoc network system, the main reason for causing congestion is because of the limited availability of resources. On the other hand, the standardised TCP based congestion controlling mechanism is unable to control and handle the major properties associated with the shared system of wireless channels. It creates an effect on the design associated with suitable protocols along with protocol stacks through the process of determining the mechanisms of congestion on a complete basis. Moreover, when bringing a comparison with standard TCP systems the major environment associated with mobile ad hoc network is regraded to be more problematic on a complete basis. On the other hand, an agent-based mobile technique for congestion is designed and developed for the part of avoiding any mode of congestion under the ad-hoc network systems.

• 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.

• Journal of Convergence for Information Technology
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• v.10 no.12
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• pp.22-30
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• 2020

With the development of information and communication technology, DDoS and DRDoS continue to become security issues, and gradually develop into advanced techniques. Recently, IT companies have been threatened with DRDoS technology, which uses protocols from normal servers to exploit as reflective servers. Reflective traffic is traffic from normal servers, making it difficult to distinguish from security equipment and amplified to a maximum of Tbps in real-life cases. In this paper, after comparing and analyzing the DNS amplification and Memcached amplification used in DRDoS attacks, a countermeasure that can reduce the effectiveness of the attack is proposed. Protocols used as reflective traffic include TCP and UDP, and NTP, DNS, and Memcached. Comparing and analyzing DNS protocols and Memcached protocols with higher response sizes of reflective traffic among the protocols used as reflective traffic, Memcached protocols amplify ±21% more than DNS protocols. The countermeasure can reduce the effectiveness of an attack by using the Memcached Protocol's memory initialization command. In future studies, various security-prone servers can be shared over security networks to predict the fundamental blocking effect.