• Proceedings of the Korean Information Science Society Conference
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• 1998.10a
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• pp.480-482
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• 1998

최근, 멀티캐스트 기법을 사용하는 멀티미디어 응용 프로그램들이 인터넷에 등장하고 있다. 이들 응용 프로그램들의 성공 여부는 수신자들에게 전송되는 음성/영상의 품질에 의해 좌우된다. 인터넷은 응용프로그램의 QoS(Quality of Service) 에 대한 요구를 보장할 수 없기 때문에 멀티케스트 트래픽(multicast traffic)을 위하여 인터넷의 성능을 최대한 효율적으로 이용할 수 있도록 흐름제어에 대한 많은 연구가 진행되고 있다. 그 중 IVS(INRIA Video conferencing System)에서 제안한 멀티캐스트 트래픽 흐름제어 알고리즘은 수신자가 주기적으로 전달하는 RTCP 의 패킷손실 정보에 의해 송신자가 전송율을 조절하는 것이다. 그러나 이 알고리즘은 네트워크 상태가 무부하(unload)임에도 불구하고 느린 피드백으로 인하여 가용 네트워크 대역폭을 빠르게 파악하지 못하기 때문에, TCP트래픽과 경쟁 상태에서 네트워크 대역폭을 불공정(unfairness)하게 사용하게 되고 네트워크 상태에 알맞는 전송율을 결정하지 못한다. 본 논문에서는 더욱 공정하게 대역폭을 공유할 수 있고 전체 링크 이용율을 높이는 두 가지 기법을 제안한다. 첫째, 측정된 네트워크 혼잡상태에 따라 RTCP 피드백의 전송 빈도를 동적으로 조절하는 것이다. 둘째, TCP와 같이 전송율을 증가/감소시킴으로써 공정하게 네트워크를 공유하도록 하는 것이다. 본 논문에서는 이 두 가지 기법들이 TCP 트래픽에 영향을 주지 않고 또한 RTCP피드백의 양을 증가시키지 않으면서도 공정하게 네트워크 대역폭을 공유함으로써 링크의 이용율을 높일 수 있다는 것을 시뮬레이션을 통하여 보여준다.안 모니터링 기 능 등으로 조사되었다.도 멜-켑스트럼을 사용한 경우 67.5%, K-L계수를 사용한 경우 75.3%로 7.8%의 향상된 인식률을 보였으며 K-L계수와 회귀계수를 결합한 경우에서도 비교적 높은 인식률을 보여 숫자음에 대해서도 K-L계수의 유효성을 확인할 수 있었다..rc$ 구입할 때 중점적으로 살펴보는 사항은 신선도와 순수재래종 여부, 위생상태였다. 한편 소비자가 언제나 구입할 수 없다는 의견이 85.2%나 되어 원활한 공급과 시장조성이 아직 정착되지 않고 있었다. $\bigcirc$ 현재 유통되고 있는 재래종닭은 소비자 대부분이 잡종으로 인식하고 있었으며, 재래종과 일반육계와의 구별은 깃털색, 피부색, 정강이색등 외관상으로 구별하고 있었다. 체중에 대한 반응은 너무 작다는 의견이었고, 식품으로의 인식도는 비교적 고급식품으로 인식하고 있다. $\bigcirc$ 재래종닭고기의 브랜드화에 대한 견해는 젊고 소득이 높은 계층에서 브랜드화의 필요성을 강조하고 있다. $\bigcirc$ 재래종달걀의 소비형태는 대부분의 소비자가 좋아하였으나 아직 먹어보지 못한 응답자가 많았다. 재래종달걀의 맛에 대해서는 고소하고 독특하여 차별성을 느끼고 있었다. $\bigcirc$ 재래종달걀의 구입장소는 계란판매점(축협.농협), 슈퍼, 백화점, 재래닭 사육 농장등 다양하였으며 포장단위는 10개를 가장 선호하였고, 포장재료는 종이, 플라스틱, 짚의 순으로 좋아하였다. $\bigcirc$ 달걀의 가격은 200원정도를 적정하다고 하였으며, 크기는 (평균 52g)는 가장 적당하다고

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.10
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• pp.1-9
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• 2004

User-Perceived application-level performance is a key to the adoption and success of CDMA 2000. To predict this performance in advance, a detailed end-to-end simulation model of a CDMA network was built to include application traffic characteristics, network architecture, network element details, and protocol features. We assess the user application performance when a Radio Access Network (RAN) and a Core Network (CN) adopt different transport architectures such as ATM and If. For voice Performance, we found that the vocoder bypass scenario shows 8% performance improvement over the others. For data packet performance, we found that HTTP v.1.1 shows better performance than that of HTTP v.1.0 due to the pipelining and TCP persistent connection. We also found that If transport technology is better solution for higher FER environment since the IP packet overhead is smaller than that of ATM for web browsing data traffic, while it shows opposite effect to small size voice packet in RAN architecture. Though simulation results we showed that the 3G-lX EV system gives much better packet delay performance than 3G-lX RTT, the main conclusion is that end-to-end application-level performance is affected by various elements and layers of the network and thus it must be considered in all phases of the technology evolution process.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.2
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• pp.35-46
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• 2013

As a rapid increase of mobile network usage, many studies on solution for network traffic's demand problem have been done. Especially network environment measurement area provides basis for solving network traffic's demand problem by finding causes of problems through accurate network analysis. However, as increase of demand for smartphone, we should consider effects of mobile platform's property measuring mobile network. In this paper, we design a network traffic measurement system considering mobile platform. Through the information from packets, this system calculates packet transmission delay and throughput. We minimize computation cost required for a mobile device that is a client in this system. When fully using network resources, we found that Wi-Fi has shorter transmission delay, higher maximum throughput and lower loss rate than 3G, Android has shorter transmission delay and higher maximum throughput than iOS, and UDP has longer transmission delay and higher maximum throughput through this system.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.3
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• pp.523-541
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• 2011

In current IEEE 802.11 wireless LAN, every unicast transmission requires an ACK from the receiver for reliability, though it consumes energy and bandwidth. There have been studies to remove or reduce ACK overhead, especially for energy efficiency. However none of them are practically used now. This paper introduces a noble method of selective unacknowledged transmission, where skipping an ACK is dynamically decided frame by frame. Utilizing the fact that a multicast frame is transmitted without accompanying an ACK in 802.11, the basic unacknowledged transmission is achieved simply by transforming the destination address of a frame to a multicast address. Since removing ACK is inherently more efficient but less strict, its practical profit is dependent on traffic characteristics of a frame as well as network error conditions. To figure out the selective conditions, energy and performance implications of unacknowledged transmission have been explored. Extensive experiments show that energy consumption is almost always reduced, but performance may be dropped especially when TCP exchanges long data with a long distance node through a poor wireless link. An experiment with a well-known traffic model shows that selective unacknowledged transmission gives energy saving with comparable performance.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.13 no.12
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• pp.5948-5971
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• 2019

Estimating loss rates with the packet traces captured from some point in the middle of the network has received much attention within the research community. Meanwhile, existing intermediate-point methods like [1] require the capturing system to capture all the TCP traffic that crosses the border of an access network (typically Gigabit network) destined to or coming from the Internet. However, limited to the performance of current hardware and software, capturing network traffic in a Gigabit environment is still a challenging task. The uncaptured packets will affect the total number of captured packets and the estimated number of packet losses, which eventually affects the accuracy of the estimated loss rate. Therefore, to obtain more accurate loss rate, a method of strengthening packet loss measurement from the network intermediate point is proposed in this paper. Through constructing a series of heuristic rules and leveraging the binomial distribution principle, the proposed method realizes the compensation for the estimated loss rate. Also, experiment results show that although there is no increase in the proportion of accurate estimates, the compensation makes the majority of estimates closer to the accurate ones.

• 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.

• Journal of KIISE:Information Networking
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• v.31 no.4
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• pp.414-428
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• 2004

Many portable devices are coming to be commercially successful and provide useful services to mobile users. Mobile devices may request a variety of data types, including text and multimedia data, thanks to the rich content of the Internet. Different types of data and/or different classes of users may need to be treated with different qualities of service. The implementation of service differentiation in wireless networks is very difficult because of device mobility and wireless channel contention when the backoff algorithm is used to resolve contention. Modification of the t)mary exponential backoff algorithm is one possibility to allow the design of several classes of data traffic flows. We present a study of modifications to the backoff algorithm to support three classes of flows: sold, silver, and bronze. For example, the gold c]ass flows are the highest priority and should satisfy their required target bandwidth, whereas the silver class flows should receive reasonably high bandwidth compared to the bronze class flows. The mixture of the two different transport protocols, UDP and TCP, in ad hoc networks raises significant challenges when defining backoff algorithm modifications. Due to the different characteristics of UDP and TCP, different backoff algorithm modifications are applied to each class of packets from the two transport protocols. Nevertheless, we show by means of simulation that our approach of backoff algorithm modification clearly differentiates service between different flows of classes regardless of the type of transport protocol.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.8
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• pp.975-983
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• 1999

최근 네트워크의 속도가 빨라지고 멀티미디어 데이터를 다루기 위한 기술들이 개발됨에 따라 많은 멀티미디어 응용 프로그램들이 인터넷에 등장하고 있다. 그러나 이들 응용프로그램들은 수신자에게 전송되는 영상.음성의 품질이 낮기 때문에 기대만큼 빠르게 확산되지 못하고 있다. 영상.음성의 품질이 낮은 이유는 현재 인터넷이 실시간 응용프로그램이 요구하는 만큼 빠르고 신뢰성 있게 데이터를 전송할 수 없기 때문이다. 현재 인터넷의 내부구조를 바꾸지 않고 품질을 높이기 위해 많은 연구들이 진행되고 있는데 그 중 하나는 동적으로 변화하는 인터넷의 상태에 맞게 멀티캐스트 트래픽의 전송율을 조절하는 종단간의 흐름제어이다. 본 논문은 기존의 흐름제어 기법인 IVS와 RLM의 성능을 개선시키기 위한 두 가지 흐름제어 기법을 소개한다. IVS는 송신자가 주기적으로 측정된 네트워크 상태에 따라 전송율을 일정하게 조절한다. 송신자가 하나의 데이타 스트림을 생성하는 IVS와는 달리 RLM에서는 송신자가 계층적 코딩에 의하여 생성된 여러개의 데이타 스트림을 전송하고 각 수신자는 자신의 네트워크 상태에 맞게 데이타 스트림을 선택하는 기법이다. 그러나 IVS는 송신자가 전송율을 일정하게 증가시키고, RLM은 각자의 네트워크 상태를 고려하지 않고 임의의 시간에 하나 이상의 데이타 스트림을 받기 때문에 성능을 저하시킬 수 있다. 본 논문에서는 TCP-like IVS와 Adaptive RLM이라는 두 가지 새로운 기법을 소개한다. TCP-like IVS는 송신자가 전송율을 동적으로 결정하고, Adaptive RLM은 하나 이상의 데이타 스트림을 받기 위해 적당한 시간을 선택할 수 있다. 본 논문에서는 시뮬레이션을 통해 여러 가지 네트워크 구조에서 두 가지 방식이 기존의 방식에 비하여 더욱 높은 대역폭 이용율과 10~20% 정도 적은 패킷손실율을 이룬다는 것을 보여준다.Abstract Nowadays, many multimedia applications for the Internet are introduced as the network gets faster and many techniques manipulating multimedia data are developed. These multimedia applications, however, do not spread widely and are not fast as expected at their introduction time due to the poor quality of image and voice delivered at receivers. The poor quality is mainly attributed to that the current Internet can not carry data as fast and reliably as the real-time applications require. To improve the quality without modifying the internal structure of the current Internet, many researches are conducted. One of them is an end-to-end flow control of multicast traffic adapting the sending rate to the dynamically varying Internet state. This paper proposes two flow-control techniques which can improve the performance of the two conventional techniques; IVS and RLM. IVS statically adjusts the sending rate based on the network state periodically estimated. Differently from IVS in which a sender produces one single data stream, in RLM a sender transmits several data streams generated by the layered coding scheme and each receiver selects some data streams based on its own network state. The more data streams a receiver receives, the better quality of image or voice the receiver can produce. The two techniques, however, can degrade the performance since IVS increases its sending rate statically and RLM accepts one more data stream at arbitrary time regardless of the network state respectively. We introduce two new techniques called TCP-like IVS and Adaptive RLM; TCP-like IVS can determine the sending rate dynamically and Adaptive RLM can select the right time to add one more data stream. Our simulation experiments show that two techniques can achieve better utilization and less packet loss by 10-20% over various network topologies.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.12
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• pp.34-43
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• 2007

Recently, in addition to the sharp increase of mobile nodes, various kinds of wireless technologies are available for mobile nodes. If IPv6 technology is applied to the network, multi-homing terminals which have several public IP addresses on one interface will be common. Accordingly, there are many research activities on mobility management for multi-interface, multi-homming nodes. In this paper we propose an extended fast handover mechanism for multi-interface MIPv6 environments that uses multi-interface FBU (MFBU) message instead of the existing FBU message. The MFBU message has the "tunnel destination" mobility option that points a specific tunnel destination other than NAR, and "T" flag that indicates the existence of tunnel destination option. The proposed mechanism can improve the TCP performance by mitigating packet reordering during FMIPv6 handover that can cause unnecessary congestion control due to 3 duplicate ACKs. In this paper, we implemented a multi-Interface MIPv6 simulator by extending a single-interface MIPv6 simulator in NS-2, and showed that the performance of TCP traffic is improved by using the proposed multi-interface fast MIPv6.

• Journal of Information Technology and Architecture
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• v.10 no.1
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• pp.101-111
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• 2013

Internet traffic volume has been increasing rapidly due to popularization of various smart devices and Internet development. In particular, HTTP-based traffic volume of smart devices is increasing rapidly in addition to desktop traffic volume. The increased mobile traffic can cause serious problems such as network overload, web security, and QoS. In order to solve these problems of the Internet overload and security, it is necessary to accurately detect applications. Traditionally, well-known port based method is utilized in traffic classification. However, this method shows low accuracy since P2P applications exploit a TCP/80 port, which is used for the HTTP protocol; to avoid firewall or IDS. Signature-based method is proposed to solve the lower accuracy problem. This method shows higher analysis rate but it has overhead of signature generation. Also, previous signature-based study only analyzes applications in HTTP protocol-level not application-level. That is, it is difficult to identify application name. Therefore, previous study only performs protocol-level analysis. In this paper, we propose a signature generation method to classify HTTP-based traffics in application-level using the characteristics of typical semi HTTP header. By applying our proposed method to campus network traffic, we validate feasibility of our method.