• Journal of Korea Game Society
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• v.2 no.1
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• pp.23-29
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• 2002

Multicast protocols are efficient methods of group communication such as video conference, Internet broadcasting and On-Line Game, but they do not support the various transmission protocol services like a reliability guarantee, FTP, or Telnet that TCPs do. The Purpose or this Paper is to find a method to utilize multicast routers can simultaneously transport multicast packets and TCP packets. For multicast network scalability and error recovery the existing SRM(Scalable Reliable Multicast)method has been used. Three packets per TCP transmission control window site are used for transport and an ACK is used for flow control. A CBR(Constant Bit Rate) and a SRM is used for UDP traffic control. Divided on whether a UDP multicast packet and TCP unicast packet is used simultaneously or only a UDP multicast packet transport is used, the multicast receiver with the longest delay is measured on the number of packets and its data receiving rate. It can be seen that the UDP packet and the TCP's IP packet can be simultaneously used in a server router.

• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.3
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• pp.303-308
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• 2016

Recently, as the rapid development of network technology and the increase of services required high bandwidth such as multimedia service, the network traffic dramatically increases. This massive increase of network traffic causes some problems such as the degradation of QoS and the lack of network resources and, to solve these problems, various research to guarantee QoS have been performing. Currently, The most representative method to guarantee the QoS is the DiffServ(: Differentiated Service). The DiffServ defines the AF(: Assured Forwarding) PHB(: Per Hop Behavior) and statistically ensures the throughput over the certain level of data rate. However, the TCP congestion control method that make up the majority of the Internet traffic is not fundamentally suitable to the DiffServ that guarantees the throughput without managing the individual flow. Therefore, in this paper, we present this mismatch through the simulation as an example and propose the solution by controlling the TCP of the terminal in the network. The proposed scheme utilizes the information of the reception window size included in the ACK frame and does not require any modification of the TCP algorithms currently in use.

• Journal of KIISE:Information Networking
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• v.32 no.4
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• pp.443-451
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• 2005

Stateful inspection devices must maintain flow information. These devices create the flow information also for network attack packets, and it can fatally inflate the dynamic memory allocation on stateful inspection devices under network attacks. The memory inflation leads to memory overflow and subsequent performance degradation. In this paper, we present a guideline to set the flow entry timeout for a stateful inspection device to remove harmful embryonic entries created by network attacks. Considering Transmission Control Protocol (TCP) if utilized by most of these attacks as well as legitimate traffic, we propose a parsimonious memory management guideline based on the design of the TCP and the analysis of real-life Internet traces. In particular, we demonstrate that for all practical purposes one should not reserve memory for an embryonic TCP connection with more than (R+T) seconds of inactivity where R=0, 3, 9 and $1\leqq{T}\leqq{2}$ depending on the load level.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.1B
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• pp.29-40
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• 2004

In this paper, we propose a new early congestion detection and notification technique called QR-AQM. Unlike RED and it's variation, QR-AQM measures the total traffic rate from TCP sessions, predicts future network congestion, and determine the packet marking probability based on the measured traffic rate. By incorporating the traffic rate in the decision process of the packet marking probability, QR-AQM is capable of foreseeing future network congestion as well as terminating congestion resolution procedure in much more timely fashion than RED. As a result, simulation results show that QR-AQM maintains the buffer level within a fairly narrow range around a target buffer level that may be selected arbitrarily as a control parameter. Consequently, compared to RED and its variations, QR-AQM is expected to significantly reduce the jitter and delay variance of packets traveling through the buffer while achieving nearly identical link utilization.

• Journal of Communications and Networks
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• v.11 no.5
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• pp.518-528
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• 2009

IEEE 802.11 wireless local area network (WLAN) has become the prevailing solution for wireless Internet access while transport control protocol (TCP) is the dominant transport-layer protocol in the Internet. It is known that, in an infrastructure-based WLAN with multiple stations carrying long-lived TCP flows, the number of TCP stations that are actively contending to access the wireless channel remains very small. Hence, the aggregate TCP throughput is basically independent of the total number of TCP stations. This phenomenon is due to the closed-loop nature of TCP flow control and the bottleneck downlink (i.e., access point-to-station) transmissions in infrastructure-based WLANs. In this paper, we develop a comprehensive analytical model to study TCP dynamics in infrastructure-based 802.11 WLANs. We calculate the average number of active TCP stations and the aggregate TCP throughput using our model for given total number of TCP stations and the maximum TCP receive window size. We find out that the default minimum contention window sizes specified in the standards (i.e., 31 and 15 for 802.11b and 802.11a, respectively) are not optimal in terms of TCP throughput maximization. Via ns-2 simulation, we verify the correctness of our analytical model and study the effects of some of the simplifying assumptions employed in the model. Simulation results show that our model is reasonably accurate, particularly when the wireline delay is small and/or the packet loss rate is low.

• Journal of IKEEE
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• v.10 no.1 s.18
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• pp.30-39
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• 2006

The TCP protocol can provide some reliability using sliding window mechanism for data transmission, flow control, and congestion control. However, TCP has some limitations in that it has basically been designed solely for wired communication environments. If traditional TCP protocol is used also in wireless networks, the end-to-end data transmission performance degrades dramatically due to frequent packet losses caused by transmission errors and hand-offs. While there have been some research efforts on TCP enhancements considering the mobility of wireless communication devices, in this paper we propose a new method to improve the TCP performance by combining the Snoop and the Freeze-TCP methods. In the proposed scheme, the TCP end-to-end semantics is maintained and no changes of existing protocols in sending systems or in routers are required. It has the advantage of simple implementation because TCP code changes are limited to mobile devices for applying the Freeze-TCP and it requires only to add Snoop modules in base stations. Accordingly, the proposed scheme can operate well in the existing networks. Finally, in this study, we compared the performance of the proposed scheme with traditional TCP, other approaches through simulations using ns-2.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2015.05a
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• pp.649-652
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• 2015

Previous marking policy for the AF service of TCP traffic in the Diffserv network have no sufficient consideration on the effect of RTT and target rate. In this paper, in order to improve fairness Index by the effect RTT difference of TCP traffic, we propose the modified TSW3CDM(Time Sliding Window Three Color Dynamic Marker) based on average transfer rate estimation and the flow state. The proposed algorithm is dynamic marking policy that do allocate band width in proportion to transmission rate. To evaluate the performance of the proposed algorithm, We accomplished a computer simulation using NS-2. From simulation results, the proposed TSW3CDM algorithm improves fairness index by comparison with TSW3CM.

• The KIPS Transactions:PartC
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• v.11C no.2
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• pp.245-252
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• 2004

Stream Control Transmission Protocol(SCTP) is a new connection-oriented, reliable delivery transport protocol operating on top of an unreliable connectionless packet service such as IP. It inherits many of the functions developed for TCP, including flow control and packet loss recovery functions. In addition, it also supports transport layer multihoming and multistreaming In this paper, we study the impact of multi-homing on the performance of SCTP. We first compare performance of single-homed SCTP. multi-homed SCTP, TCP Reno and TCP SACK. We, then describe potential flaw in the current SCTP retransmission policy, when SCTP host is multihomed. Our Results show that SCTP performs better than TCP Reno and TCP SACK due to several changes from TCP in its congestion control mechanism. In particular. multi-homed SCTP shows the best result among the compared schemes. Through experimentation for multi-homed SCTP, we found that the current SCTP retransmission policy nay deteriorate the perfomance when the retransmission path it worse than the original path. Therefore, the condition of retransmission path is a very important factor In SCTP performance and a proper mechanism would be required to measure the condition of the retransmission path.

• Journal of IKEEE
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• v.11 no.4
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• pp.331-339
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• 2007

The problem of architecting a reliable transport system across an overlay network using split TCP connections as the transport primitive is mainly considered. The considered overlay network uses the application-level switch in each intermediate host. We first argue that natural designs based on store-and-forward principles that are maintained by split TCP connections of hop-by-hop approaches. These approaches in overlay networks do not concern end-to-end TCP semantics. Then, a new transport protocol-Overlay Transport Protocol (OTP)-that manages the end-to-end connection and is responsible for the congestion/flow control between source host and destination host is proposed. The proposed network model for the congestion and flow control mechanisms uses a new window size-Ownd-and a new timer in the source host and destination host. We validate our analytical findings and evaluate the performance of our OTP using a prototype implementation via simulation.

• The Journal of the Korea Contents Association
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• v.8 no.8
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• pp.57-64
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• 2008

In this paper, to resolve the problem caused by a network state information inaccuracy the slow delay time that conclusion of network state of one way delay time which accuracy delay time information, according to network state changes on the based TFRC flow control, and suggest that flow control mechanism to adjust transfer rate fit of real time multimedia data. In simulation, to measure of netowork state information that on the average about 12% difference of compared RTT and $OWD{\times}2$. When used RTT, used fair bandwidth TFRC much better than TCP about 32%, when used OWD, difference about 3% used fair bandwidth. Thus, conclusion of accuracy network state that used fair bandwidth according to network state changes on the based TFRC, users can support service of high quality that flow control mechanism to adjust transfer rate fit of real time data.