• The KIPS Transactions:PartC
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• v.9C no.5
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• pp.647-654
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• 2002

Ad-hoc networks consist of a set of mobile hosts that communicate using wireless links, without the use of other supporting communication facilities (such as base stations, etc.). Therefore, the topology of an Ad-hoc network frequently changes due to the movement of mobile host, which nay lead to sudden packet loss. Recently, the large amount of research has focused on the routing protocols needed in such an environment. In this paper, TCP Reno, Sack, and Tahoe versions are analysed using DSR protocol which is the representative On-Demand routing protocol in Ad-hoc wireless network. As the result of this simulation, we know that TCP Reno relatively has higher throughput than that of Sack and Tahoe, and TCP Reno has more stable performance than that of TCP Tahoe and Sack, regardless of the speed of mobile node and the size of topology.

• The KIPS Transactions:PartC
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• v.11C no.6 s.95
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• pp.787-798
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• 2004

High performance net재rking is one of key factors to provide support for data intensive applications in the Internet. Extended frame size has a major impact on end to-end performance with increasing effective TCP throughput and decreasing system overhead. Most of the research about extended frames has focused on local area network performance and the impact that extended frame size has on the system elements including memory, network interface card and so forth. In the paper, we analyse the effects of the extended frames to the other traffic flows sharing Internet paths for the wide area performance of TCP by conducting various network simulations. Results show that securing available bandwidth in no loss and low delay networks is indispensable to exploit the efficiency of extended frames.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1496-1502
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• 1998

In this paper, a real-time audio broadcasting service system which is robust to loaded traffic on the Internet is developed. For implementing reliable real-time data transfer, the transfer characteristics of TCP/IP and UDP/IP was compared and analyzed. For lost packet recovery, redundant audio data algorithm was used and interleaving technique was applied for scattering consecutive packet loss. Test results showed, when using TCP/IP, pause occurred during playback, and when using UDP/IP, a stable receive rate was noticeable but the quality of the sound was lower than that of uisng TCP/IP. The recovery rate using redundant audio data and interleaving technique is shown in Fig. 9 and the delay is shown in Fig 4.

• International Journal of Computer Science & Network Security
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• v.22 no.2
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• pp.327-333
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• 2022

Internet Protocol Television (IPTV) has emerged as a personal entertainment source for home users. Streaming IPTV content over a wireless medium with good Quality of Service (QoS) can be a challenging task as IPTV content requires more bandwidth and Wireless Local Area Networks (WLANs) are susceptible to packet loss, delay and jitter. This research presents the capacity of IPTV using User Datagram Protocol (UDP) and TCP Friendly Rate Control (TFRC) over IEEE 802.11ac WLANs in good and bad network conditions. Experimental results show that in good network conditions, UDP and TFRC could accommodate a maximum of 78 and 75 Standard Definition Television (SDTV) users, respectively. In contrast, 15 and 11 High-Definition Television (HDTV) users were supported by UDP and TFRC, respectively. Performance of UDP and TFRC was identical in bad network conditions and same number of SDTV and HDTV users were supported by TFRC and UDP. With background Transmission Control Protocol (TCP) traffic, both UDP and TFRC can support nearly the same number of SDTV users. It was found that TFRC can co-exist fairly with TCP by giving more throughput to TCP unlike UDP.

• The KIPS Transactions:PartC
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• v.8C no.1
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• pp.51-59
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• 2001

In order to support various QoS, IETF has proposed the Differentiated Services Model which provides discrimination service according to t the user’s requirements and payment intention intention for each traffic characteristic. This model is an excellent mechanism, which is not too c complicated in terms of the management for service and network model. Also, it has scalability that satisfies the requirement of Differentiated Services. In this paper, We define the Differentiated Services Model using feedback control, propose its control procedure, and analyze its p performance. In conventional model, non-adaptive traffic, such as UDP traffic, is more occupied the network resource than adaptive traffic, such a as TCP traffic. On the other hand, the Differentiated Services Model using feedback control fairly utlizes the network resources and even p prevents congestion occurrence due to its ability of congestion expectation.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.21 no.3
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• pp.549-557
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• 2017

ICMP(Internet Control Message Protocol) is a main protocol in TCP/IP protocol stack. ICMP compensates the disadvantages of the IP that does not support error reporting. If any transmission problem occurred, a router or receiving host sends ICMP message containing the error cause to sending host. However, in this process, an attacker sends a fake ICMP messages to the host so that the communication can be terminated abnormally. An attacker host can paralyzes system of victim host by sending a large number of messages to the victim host at a high rate of speed. To solve this problem, we have designed and implemented outbound traffic controller that prevents various ICMP attacks. By preventing the transmission of attack messages in different ways according to each case, various network attacks can be prevented. In addition, unnecessary network traffic can be filtered before transmitted.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.3
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• pp.762-777
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• 2014

With the increasing usage of cloud applications such as MapReduce and social networking, the amount of data traffic in data center networks continues to grow. Moreover, these appli-cations follow the incast traffic pattern, where a large burst of traffic sent by a number of senders, accumulates simultaneously at the shallow-buffered data center switches. This causes severe packet losses. The currently deployed TCP is custom-tailored for the wide-area Internet. This causes cloud applications to suffer long completion times towing to the packet losses, and hence, results in a poor quality of service. An Explicit Congestion Notification (ECN)-based approach is an attractive solution that conservatively adjusts to the network congestion in advance. This legacy approach, however, lacks scalability in terms of the number of flows. In this paper, we reveal the primary cause of the scalability issue through analysis, and propose a new congestion-control algorithm called FaST. FaST employs a novel, virtual congestion window to conduct fine-grained congestion control that results in improved scalability. Fur-thermore, FaST is easy to deploy since it requires only a few software modifications at the server-side. Through ns-3 simulations, we show that FaST improves the scalability of data center networks compared with the existing approaches.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.782-787
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• 2004

Internet Engineering Task Force (IETF) has been considering the deployment of the Random Early Detection (RED) in order to avoid the increasing of packet loss rates which caused by an exponential increase in network traffic and buffer overflow. Although RED mechanism can prevent buffer overflow and hence reduce an average values of packet loss rates, but this technique is ineffective in preventing the consecutive drop in the high traffic condition. Moreover, it increases a probability and average number of consecutive dropped packet in the low traffic condition (named as "uncritical condition"). RED mechanism effects to TCP congestion control that build up the consecutive of the unnecessary transmission rate reducing; lead to low utilization on the link and consequently degrade the network performance. To overcome these problems, we have proposed a new mechanism, named as Extended Drop slope RED (ExRED) mechanism, by modifying the traditional RED. The numerical and simulation results show that our proposed mechanism reduces a drop probability in the uncritical condition.

• 대한공업교육학회지
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• v.33 no.2
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• pp.287-299
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• 2008

HTTP is one of the most widely used protocols of the WWW. Currently it uses TCP as the transport layer protocol to provide reliability. The HTTP uses separate TCP connection for each file request and adds unnecessary head-of-line blocking overhead for the file retrieval. The web application is short sized and affected by the increased handover latency of TCP in wireless environment. SCTP has attractive features such as multi-streaming and multi-homing. SCTP's multi-streaming and multi-homing avoid head-of-line blocking problem of TCP and reduce handover latency of TCP in wired and wireless environment. Mean response time is the important measure in most web application. In this paper, we present the comparison of mean response time between HTTP over SCTP with that of HTTP over TCP in wired and wireless environments using NS-2 simulator. We measured mean response time for varying packet loss rate, bandwidth, RTT, and the number of web objects in wired environment and mean response time and packet loss rate for varying moving speed and region size in wireless environment. Our experimental result shows that SCTP reduces the mean response time of TCP based web traffic.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.711-721
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• 2004

To provide the end-to-end service differentiation for assured services, the random early demotion and promotion (REDP) marker in the edge router at each domain boundary monitors the aggregate flow of the incoming in-profile packets and demotes in-profile packets or promotes the previously demoted in-profile packets at the aggregate flow level according to the negotiated interdomain service level agreement (SLA). The REDP marker achieves UDP fairness in demoting and promoting packets through random and early marking decisions on packets. But, TCP fairness of the REDP marker is not obvious as fur UDP sources. In this paper, to improve TCP fairness of the REDP marker, we combine a dropper with the REDP marker. To make packet transmission rates of TCP flows more fair, at the aggregate flow level the combined dropper drops incoming excessive in-profile packets randomly with a constant probability when the token level in the leaky bucket stays In demotion region without incoming demoted in-profile packets. It performs a dropping in the demotion at a domain boundary only if there is no prior demotion. The concatenate dropping at multiple domains is avoided to manifest the effect of a dropping at a domain boundary on TCP fairness. We experiment with the REDP marker with the combined dropper using ns2 simulator for TCP sources. The simulation results show that the REDP marker with the combined dropper improves TCP fairness in demoting and promoting packets by generating fair demoted in-profile traffic compared to the REDP marker. The effectiveness of the selected drop probability is also investigated with showing its impact on the performance of the REDP marker with the combined dropper.