• Journal of Advanced Navigation Technology
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• v.11 no.1
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• pp.31-36
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• 2007

TCP (Transmission Control Protocol) is one of the protocols which are widely used from the Internet environments. Through the flow control of TCP, it could be increased efficiency for the loss and a re-transmission of data and the flow control become accomplished through window technique which puts the limit of size. By the flow control, TCP divided in various versions. In this paper, it is analyzed the simulation result which applies the error model in the Newreno which is an improved model of the representative Tahoe, Reno.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.1061-1064
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• 1999

UDP has likely been used for real-time applications, such as video and audio. UDP supplies minimized transmission delay by omitting the connection setup process, flow control, and retransmission In general, more than 80 percent of the WAN resources are occupied by Transmission Control Protocol(TCP) traffic as opposed to UDP's simplicity, TCP adopts a unique flow control in this paper, I report new methods to minimize a udp packet loss considering TCP flow control on the real-time application the better performance of real time application can be obtained when they reduce a packet size and FIFO buffer scheduling method competing with TCP bandwidth for the bandwidth and buffering.

• The Transactions of the Korea Information Processing Society
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• v.7 no.5
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• pp.1520-1524
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• 2000

TCP congestion control algorithm prevents network congestion through the control of outgoing traffic size. The network, therefore, should monitor the incoming traffic size of a TCP to determine whether or not a TCP follows standard congestion control algorithms. Some TCP friendly test algorithms are proposed, But, these algorithms cannot be used in real environments because a router in a network does not know the RTT of a TCP flow. In this study, we propose a new RTT determination algorithm that can be used in a router. Our proposed algorithms is validated through the simulation studies.

• Journal of Institute of Control, Robotics and Systems
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• v.18 no.8
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• pp.774-779
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• 2012

We design and implement a RTT (Round Trip Time) based TCP (Transmission Control Protocol) for USN (Ubiquitous Sensor Network). We adopt a basic update algorithm for window size from FAST TCP that uses the queuing delay at link as the congestion measure. The designed TCP estimates the queuing delay at link from the measured RTT in the network layer, and updates the window size based on the estimated queuing delay. The designed TCP allows to utilize the full capacity of USN links and avoids the waste of the given link capacity that is common without the flow control in the transport layer. The experiment results show that the window size of the sender converges within a small range of variations without any packet loss, and verify the stability and performance of the designed TCP.

• Journal of KIISE:Information Networking
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• v.31 no.1
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• pp.20-26
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• 2004

A congestion control algorithm to prevent buffer overflow in MSR(Mobility Support Router) for I-TCP is proposed. Due to high bit error rate and frequent hand-offs over wireless environment, the current congestion control scheme in TCP Reno over mixed(wired and wireless) network exhibits lower throughput than the throughput achieved over wired only network. I-TCP has been proposed to address this by splitting a TCP connection into two TCP connections over wired section and wireless section, respectively. However, buffer overflow in MSR may occur whenever there are excessive bit errors or frequent hand-offs. This may lead to the loss of packets acked by MSR(resident in buffer) to the sender, but not received by the receiver, breaking TCP end-to-end semantics. In this Paper, a new scheme is proposed to prevent the MSR buffer from overflow by introducing “flow control” between the sender and the MSR. Advertised window for the TCP connection between the sender and the MSR is tied to the remaining MSR buffer space, controlling the flow of packets to the MSR buffer before overflow occurs.

• Journal of KIISE:Information Networking
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• v.30 no.5
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• pp.649-655
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• 2003

Network-based computing such as cluster computing requires a reliable high-speed transport protocol. TCP is a representative reliable transport protocol on the Internet, which implements many mechanisms, such as flow control, congestion control, and retransmission, for reliable packet delivery. This paper, however, finds out that Myrinet does not incur any packet losses caused by network congestion. In addition, we ascertain that Myrinet supports reliable and ordered packet delivery. Consequently, most of reliable routines implemented in TCP produce unnecessarily additional overheads on Myrinet. In this paper, we show that we can attain the reliability only by flow control on Myrinet and propose a new reliable protocol based on UDP named RUM (Reliable UDP on Myrinet) that performs a flow control. As a result, RUM achieves a higher throughput by 45% than TCP and shows a similar one-way latency to UDP.

• Journal of KIISE:Information Networking
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• v.31 no.1
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• pp.91-100
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• 2004

This paper presents a receiver-driven TCP flow control mechanism, which is adaptive to the wireless condition, for memory constrained mobile receiver. A receiver-driven TCP flow control is, in general, achieved by adjusting the size of advertised window at the receiver. The proposed method constantly measures at the receiver both the available wireless bandwidth and the packet round-trip time. Depending on the measured values, the receiver adjusts appropriately the size of advertised window. Constrained by the adjusted window which reflects the current state of the wireless network, the sender achieves an improved TCP throughput as well as the reduced round-trip packet delay. Its implementation only affects the protocol stack at the receiver and hence neither the sender nor the router are required to be modified. The mechanism has been implemented in real environments. The experimental results show that in CDMA2000 1x networks the TCP throughput of the proposed method has improved about 5 times over the conventional method when the receiver's buffer size is limited to 2896 bytes. Also, with 64Kbytes of buffer site, the packet round-trip time of the proposed method has been reduced in half, compared the case with the conventional method.

• Journal of Advanced Navigation Technology
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• v.9 no.1
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• pp.61-69
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• 2005

TCP and UDP is a transport layer protocol of Internet. TCP is a connection oriented protocol which supports a reliable data transfer by offering error and flow control, but it bring a transmission delay. On the other hand, the UDP is a connectionless protocol which does not carry out error and flow control, but it guarantees a realtime transmission. There are hardly any protocols which supports not only realtime functions but also data reliability. In this paper, we have designed and implemented a new TCP mode option which supports reliable realtime transmission. Our designed TCP performs an error recovery process during a fixed amount of time. This time is negotiated during the connection establishment phase. Our designed TCP is tested in real environments, and we find that it is relatively faster than the standard TCP and more reliable than the UDP. It can be used for the reliable transfer of realtime multimedia data.

• Journal of Communications and Networks
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• v.10 no.3
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• pp.338-350
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• 2008

The differentiated services (DiffServ) is proposed to provide packet level service differentiations in a scalable manner. To provide an end-to-end service differentiation to users having a connection over multiple domains, as well as a flow marker, an intermediate marker is necessary at the edge routers, and it should not be operated at a flow level due to a scalability problem. Due to this operation requirement, the intermediate marker has a fairness problem among the transmission control protocol (TCP) flows since TCP flows have intrinsically unfair throughputs due to the TCP's congestion control algorithm. Moreover, it is very difficult to resolve this problem without individual flow state information such as round trip time (RTT) and sending rate of each flow. In this paper, to resolve this TCP fairness problem of an intermediate marker, we propose a fair scalable marker (FSM) as an intermediate marker which works with a source flow three color marker (sf-TCM) operating as a host source marker. The proposed fair scalable marker improves the fairness among the TCP flows with different RTTs without per-flow management. Through the simulations, we show that the FSM can improve TCP fairness as well as link utilization in multiple domain DiffServ networks.

• Proceedings of the IEEK Conference
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• 2000.11a
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• pp.21-24
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• 2000

End-to-end congestion control mechanism have been critical to the robustness and stability of the Internet. Most of today's Internet traffic is TCP, and we expect this to remain so in the future. TCP/IP is the intermediate transport layer candidate for today's applications. TCP uses an adaptive window-based flow control. The congestion avoidance and control algorithms deployed by TCP aims at using the available network bandwidth. This paper compares different congestion control policies, and proposes the new design mechanism for future public networks