• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.149-158
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• 2005

This paper proposes a congestion avoidance algorithm which provides stable throughput and transmission rate regardless of buffer size by limiting the TCP congestion window in fixed bandwidth networks. Additive Increase, Multiplicative Decrease (AIMD) is the most commonly used congestion control algorithm. But, the AIMD-based TCP congestion control method causes unnecessary packet losses and retransmissions from the congestion window increment for available bandwidth verification when used in fixed bandwidth networks. In addition, the saw tooth variation of TCP throughput is inappropriate to be adopted for the applications that require low bandwidth variation. We present an algorithm in which congestion window can be limited under appropriate circumstances to avoid congestion losses while still addressing fairness issues. The maximum congestion window is determined from delay information to avoid queueing at the bottleneck node, hence stabilizes the throughput and the transmission rate of the connection without buffer and window control process. Simulations have performed to verify compatibility, steady state throughput, steady state packet loss count, and the variance of congestion window. The proposed algorithm can be easily adopted to the sender and is easy to deploy avoiding changes in network routers and user programs. The proposed algorithm can be applied to enhance the performance of the high-speed access network which is one of the fixed bandwidth networks.

• Journal of Internet Computing and Services
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• v.10 no.4
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• pp.199-211
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• 2009

Recent advancements in wireless networks have enabled support for mobile applications to transfer data over heterogeneous wireless paths in parallel using data striping technique [2]. Traditionally, high performance data transfer requires tuning of multiple TCP sockets, at sender's end, based on bandwidth delay product (BDP). Moreover, traditional techniques like Automatic TCP Buffer Tuning (ATBT), which balance memory and fulfill network demand, is designed for wired infrastructure assuming single flow on a single socket. Hence, in this paper we propose a buffer tuning technique at senders end designed to ensure high performance data transfer by striping data at transport layer across heterogeneous wireless paths. Our mechanism has the capability to become a resource management system for transport layer connections running on multi-homed mobile host supporting features for wireless link i.e. mobility, bandwidth fluctuations, link level losses. We show that our proposed mechanism performs better than ATBT, in efficiently utilizing memory and achieving aggregate throughput.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.11 s.353
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• pp.149-158
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• 2006

We proposed a method for improving a performance of TCP downstream between a desktop PC as a fixed host and a PDA as a mobile host in a wired and wireless network based on IEEE 802.11x wireless LAN. With data transmission between these heterogeneous terminals a receiving time during downstream is slower than that during upstream by 20% at maximum. The reason is that their congestion window size will be oscillated due to a significantly lower packet processing rate at receiver compared to a packet sending rate at sender. Thus it will cause to increase the number of control packets to negotiate their window size. To mitigate these allergies, we proposed two distinct methods. First, by increasing a buffer size of a PDA at application layer an internal processing speed of a socket receive buffer of TCP becomes faster and then the window size is more stable. However, a file access time in a PDA is kept nearly constant as the buffer size increases. With the buffer size of 32,768bytes the receiving time is faster by 32% than with that of 512bytes. Second, a delay between packets to be transmitted at sender should be given. With an inter-packet delay of 5ms at sender a resulting receiving time is faster by 7% than without such a delay.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2004.05b
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• pp.395-398
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• 2004

This paper describes the implementation of an internet-based servo-control system with novel direct internet control architecture which is insensitive to the inherent internet time delay. The servo control system can be controlled by using a PC provided at a local site. However, a large internet time delay may make some control inputs distorted. Moreover, since it is acted by the number of the internet nodes and loads, this delay is variable and unpredictable. This system consists of servo mechanism and Ethernet control adapter. Intention of this paper is to reduce the expenses, to manage effectively for plant and to increase the productivity through linking each plant of several factories by TCP/IP and Ethernet, and then many control plants and manager minimize the needed work. Experiments are partially fulfilled using the Labview and Matlab.

• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.25 no.1
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• pp.50-55
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• 2021

Multi-path system is a system in which utilizes various networks simultaneously. It is expected that multi-path system can enhance communication speed, reliability, security of network. In this paper, we focus on path selection in multi-path system. To select optimal path, we propose deep reinforcement learning algorithm which is rewarded by the round-trip-time (RTT) of each networks. Unlike multi-armed bandit model, deep Q learning is applied to consider rapidly changing situations. Due to the delay of RTT data, we also suggest compensation algorithm of the delayed reward. Moreover, we implement testbed learning server to evaluate the performance of proposed algorithm. The learning server contains distributed database and tensorflow module to efficiently operate deep learning algorithm. By means of simulation, we showed that the proposed algorithm has better performance than lowest RTT about 20%.

• Journal of Communications and Networks
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• v.8 no.1
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• pp.93-105
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• 2006

The domination of the Internet by TCP-based services has spawned many efforts to provide high network utilization with low loss and delay in a simple and scalable manner. Active queue management (AQM) algorithms attempt to achieve these goals by regulating queues at bottleneck links to provide useful feedback to TCP sources. While many AQM algorithms have been proposed, most suffer from instability, require careful configuration of nonintuitive control parameters, or are not practical because of slow response to dynamic traffic changes. In this paper, we propose a new AQM algorithm, hybrid random early detection (HRED), that combines the more effective elements of recent algorithms with a random early detection (RED) core. HRED maps instantaneous queue length to a drop probability, automatically adjusting the slope and intercept of the mapping function to account for changes in traffic load and to keep queue length within the desired operating range. We demonstrate that straightforward selection of HRED parameters results in stable operation under steady load and rapid adaptation to changes in load. Simulation and implementation tests confirm this stability, and indicate that overall performances of HRED are substantially better than those of earlier AQM algorithms. Finally, HRED control parameters provide several intuitive approaches to trading between required memory, queue stability, and response time.

• The KIPS Transactions:PartC
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• v.10C no.4
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• pp.479-484
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• 2003

In the current Internet, it is difficult for an Internet Phone to guarantee the QoS due to variable network conditions such as packet loss rate, delay and bandwidth. In addition, the QoS of an Internet Video Phone is more hard to guarantee because of video data. In this paper, we investigate application-level QoS control schemes that can adapt to variable network conditions, and describe an error control scheme and a congestion control scheme. Based on these QoS control schemes, we have designed and implemented an Internet Video Phone System that supports adaptive audio and video delivery. Through experiments, we found that the Internet Video Phone can reduce the packet loss rate considerably as well as adjust the transmission rate considering other TCP flows.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.12
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• pp.210-216
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• 2014

In this paper, it was proposed to develop front and rear image monitoring system for vehicle that help a driver to cope with urgent situation about a dangerous element. When parking a vehicle, the risk factors to be formed by the dead zone can be resolved by using anterior and posterior cameras of the vehicle. In embedded system environment, a SoC(System on Chip) and two high-resolution CMOS (Complementary metal-oxide-semiconductor) image sensors were used to transfer two high-resolution image data through he TCP/ IP-based network. To transfer image data through he TCP/ IP-based network, the images received by two cameras were compressed by using H.264 and they were transmitted with wireless method(Wi-Fi) by using real-time transport protocol (Real-time Transport Protocol). Transmission loss, transmission delay and transmission limit were solved in wireless (Wi-Fi) environment and the bit-rate of two image data compressed by H.264 was adjusted. And the system for the optimal transmission in wireless (Wi-Fi) environment was materialized and experimented.

• Journal of Korea Multimedia Society
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• v.8 no.1
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• pp.71-78
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• 2005

When a PDA is used as a mobile terminal in an infrastructure network based on 802.11b standard wireless LAN, we measured and evaluated the performance of multimedia data transmission. In general, PDAs are mainly used tot receiving bulk data like multimedia data. However, while a PDA as a mobile host transfers multimedia data to a desktop PC as a fixed host in such a network, the time taken to receive data from the PC to the PDA is always longer than the time taken to send data from the PDA to the PC by at least 53%. Thus, we consider some critical factors that affect the time taken to receive Our experimental results show that with the PDA the time taken to access a file for writing is longer than the time taken to access a file for reading of to 4.11 times. In addition, a dramatic change of the site of the receiver window, from 686 bytes to 32,383 bytes, prolongs the time taken to receive. To transfer data without any inter-packet interval for the PDA and to increase the size of the TCP receive buffer will be effective in improving the delay to receive data.