• The KIPS Transactions:PartC
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• v.12C no.1 s.97
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• pp.121-128
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• 2005

Tho existing TCP(Transmission Control Protocol) is known to be unsuitable for a network with the characteristics of high RDP(Bandwidth-Delay Product) because of the fixed small or large buffer size at the TCP sender and receiver. Thus, some trial cases of adjusting the buffer sizes automatically with respect to network condition have been proposed to improve the end-to-end TCP throughput. ATBT(Automatic TCP fluffer Tuning) attempts to assure the buffer size of TCP sender according to its current congestion window size but the ATBT assumes that the buffer size of TCP receiver is maximum value that operating system defines. In DRS(Dynamic Right Sizing), by estimating the TCP arrival data of two times the amount TCP data received previously, the TCP receiver simply reserves the buffer size for the next arrival, accordingly. However, we do not need to reserve exactly two times of buffer size because of the possibility of TCP segment loss. We propose an efficient TCP buffer tuning technique(called TBT-PLR: TCP buffer tuning algorithm based on packet loss ratio) since we adopt the ATBT mechanism and the TBT-PLR mechanism for the TCP sender and the TCP receiver, respectively. For the purpose of testing the actual TCP performance, we implemented our TBT-PLR by modifying the linux kernel version 2.4.18 and evaluated the TCP performance by comparing TBT-PLR with the TCP schemes of the fixed buffer size. As a result, more balanced usage among TCP connections was obtained.

• The KIPS Transactions:PartC
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• v.10C no.3
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• pp.311-316
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• 2003

To deploy multicast protocols, fairness to current Internet traffic, particularly TCP, is an important requirement. PGMCC is one of the most promising multicast congestion control proposals but it suffers from degradation of fairness by fixed timeout and uncertain acker selection. In this paper, we suggest addition of an adaptive timeout mechanism and NAK suppression in router using throughput comparison to improve fairness. Our simulation show improved fairness.

• 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 Institute of Control, Robotics and Systems
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• v.10 no.11
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• pp.1077-1081
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• 2004

In this Paper we discuss an algorithm for a real-time transmission of moving color image on the TCP/IP network using wavelet transform and neural network. The Image frames received from the camera are two-level wavelet-transformed in the server, and are transmitted to the client on the network. Then, the client performs the inverse wavelet-transform using only the received pieces of each image frame within the prescribed time limit to display the moving images. When the TCP/IP network is busy, only a fraction of each image frame will be delivered. When the line is free, the whole frame of each image will be transferred to the client. The receiver warns the sender of the condition of traffic congestion in the network by sending a special short frame for this specific purpose. The sender can respond to this information of warning by simply reducing the data rate which is adjusted with a neural network or fuzzy logic. In this way we can send a stream of moving images adaptively adjusting to the network traffic condition.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.12A
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• pp.975-983
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• 2011

In this paper, we investigate how the IEEE 802.16 based wireless system can adopt the network coding. To avoid the problem of overhearing, we focus on the bidirectional traffic, where each end node exchanges network coded data over a relay node. The bidirectional traffic is usually observed in Internet, where TCP makes congestion control and error recovery based on the acknowledgement from the opposite direction. Thus, enhancing the spectral efficiency of wireless Internet through the network coding is expected. Our simulation with realistic radio characteristics and TCP-like traffic shows that the network coding improves the throughput by an average of 36 percent compared to the simple relay case.

• The Journal of the Korea institute of electronic communication sciences
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• v.15 no.6
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• pp.1045-1054
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• 2020

In this paper, we designed a TCP/IP based ring buffer system that can stably transfer bulk data streams in the unstable network environments. In the scheme we proposed, The observation data stream generated and output by each radio observatory's backend system as a UDP frame is stored as a UDP packet in a large capacity ring buffer via a socket buffer in the client system. Thereafter, for stable transmission to the remote destination, the packets are processed in TCP and transmitted to the socket buffer of server system in the correlation center, which packets are stored in a large capacity ring buffer if there is no problem with the packets. In case of errors such as loss, duplication, and out of order delivery, the packets are retransmitted through TCP flow control, and we guaranteed that the reliability of data arriving at the correlation center. When congestion avoidance occurs due to network performance instability, we also suggest that performance degradation can be minimized by applying parallel streams.

• Proceedings of the Korea Contents Association Conference
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• 2007.11a
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• pp.126-129
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• 2007

Recently the use of wireless network increases according to it solves the hand-off and with path loss, pading, noise etc of wireless network the research for transmission error improvement is developed. TCP and SCTP of standard where it guarantees the reliability of wire network apply in wireless network the congestion control, flow control mechanism used it decreases the efficiency of data transfer throughputs. In this paper, It mixes SCTP and SNOOP for SCTP apply on wireless network, to improve BS(Basic Station) operation processes when the transmission error occurs in wireless network. BS send ZWP(Zero Window Probe) to MN(Mobile Node) when the transmission error occurs so, check path and status and update RWND and error status checked. It selects the new path, send ZWA(Zero Window Advertisement) to FH(Fixed Host) and the prevents call to congestion control or flow control and it does to make wait status standing. Continuously of data transfer after the connection of wireless network is stabilized, it make increase about 10% the transmission throughput of data.

• Journal of KIISE:Information Networking
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• v.32 no.2
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• pp.180-189
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• 2005

This paper describes a transmission power control method for power saving in 802.11b wireless LANs. We have first explored how much effects reducing transmission power has on communication performance. Then we propose a power control algorithm, whose approach is similar to that of TCP congestion control, determining an appropriate transmission power level by monitoring the retransmission rate. We have implemented an utility software on a Linux-based system and made several experiments to validate the proposed method. The results show that it is possible to save energy consumption by controlling transmission power without sacrificing communication performance.

• Journal of Internet of Things and Convergence
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• v.1 no.1
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• pp.1-4
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• 2015

This paper aims to present the mathematical model to find the mean web object transfer latency in the slow-start phase of TCP congestion control mechanism, which is one of the main control techniques of Internet. Mean latency is an important service quality measure of end-user in the network. The application area of the proposed latency model is the narrowband environment including multi-hop wireless network and Internet of Things(IoT), where packet loss occurs in the slow-start phase only due to small window. The model finds the latency considering initial window size and the packet loss rate. Our model shows that for a given packet loss rate, round trip time and initial window size mainly affect the mean web object transfer latency. The proposed model can be applied to estimate the mean response time that end user requires in the IoT service applications.

• Journal of Advanced Navigation Technology
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• v.18 no.5
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• pp.461-467
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• 2014

Active queue management (AQM) algorithms run on routers and detect incipient congestion by typically monitoring the instantaneous or average queue size. When the average queue size exceeds a certain threshold, AQM algorithms infer congestion on the link and notify the end systems to back off by proactively dropping some of the packets arriving at a router or marking the packets to reduce transmission rate at the sender. Among the existing AQM algorithms, random early detection (RED) is well known as the representative queue-based management scheme by randomizing packet dropping. To reduce the number of timeouts in TCP and queuing delay, maintain high link utilization, and remove bursty traffic biases, the RED considers an average queue size as a degree of congestions. However, RED do not well in the specified networks conditions due to the fixed parameters($P_{max}$ and $TH_{min}$) of RED. This paper addresses a extended RED to be adapted in various networks conditions. By sensing network state, $P_{max}$ and $TH_{min}$ can be automatically changed to proper value and then RED do well in various networks conditions.