• Journal of the Institute of Electronics Engineers of Korea TE
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• v.37 no.3
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• pp.86-92
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• 2000

In this paper, we develop protocol analyzer that can analyze and monitor LAN(Local Area Network)-based TCP/IP protocol using the OOP(object-oriented programming) in Windows98/NT environment. TCP/IP(Transmission Control Protocol/Internet Protocol) protocol analyzer is consist of interface hardware, protocol analysis software and GUI(Graphic User Interface). It is designed for the real-time analysis using the real-time object. In results of Performance test, TCP/IP Protocol analyzer is showed that it can analyze and monitor without frame error in LAN-based. Also, developed protocol analyser operates better than conventional protocol analyzer in performance. It can be used in maintenance fields of communication and network.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.4
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• pp.741-746
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• 2012

Transmission control protocol(TCP) widely used as a transport protocol in the Internet includes a loss recovery function that detects and recovers packet losses by retransmissions. The loss recovery function consists of the two algorithms; fast retransmit and fast recovery. There have been researches to avoid nonnecessary retransmission timeouts (RTOs), which leads to selective acknowledgement (SACK) option and limited transmit scheme that are standardized by IETF (Internet Engineering Task Force). Recently, a method that covers the case in which a retransmitted packet is lost again has been propsed. The method, however, is not proved in terms of the additive increase multiplicative decrease (AIMD) principle of TCP congestion control. In this paper, therefore, we analyzed the method in terms of the principle by ns-simulations.

• Journal of Communications and Networks
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• v.12 no.3
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• pp.275-284
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• 2010

The current multihome-aware protocols (like stream control transmission protocol (SCTP) or parallel TCP for concurrent multipath data transfer (CMT) are not designed for high-capacity and large-latency networks; they often have performance problems transferring large data files over shared long-distance wide area networks. It has been shown that SCTP-CMT is more sensitive to receive buffer (rbuf) constraints, and this rbuf-blocking problem causes considerable throughput loss when multiple paths are used simultaneously. In this research paper, we demonstrate the weakness of SCTP-CMT rbuf constraints, and we then identify that rbuf-blocking problem in SCTP multihoming is mostly due to its loss-based nature for detecting network congestion. We present a simulation-based performance comparison of FAST TCP versus SCTP in high-speed networks for solving a number of throughput issues. This work proposes an end-to-end transport layer protocol (i.e., FAST TCP multihoming as a reliable, delaybased, multihome-aware, and selective ACK-based transport protocol), which can transfer data between a multihomed source and destination hosts through multiple paths simultaneously. Through extensive ns-2 simulations, we show that FAST TCP multihoming achieves the desired goals under a variety of network conditions. The experimental results and survey presented in this research also provide an insight on design decisions for the future high-speed multihomed transport layer protocols.

• Proceedings of the IEEK Conference
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• 2002.07b
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• pp.1117-1120
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• 2002

In general, standard TCP/IP (transmission control protocol-internet protocol), which is called TCP/IP, is using as the communication basis protocol between any collections of networks that is connected. In this paper, using this TCP/IP implementation of remote system and suitable program for long distance communication is proposed. This system can make system, which basic Ethernet and TCP/IP used system, to mini modeling, so all module that is using here can be used. Therefore, intention of this paper is to reduce expenses, to effective manage for plant and to increase of productivity as linking each plant of several factories to TCP/IP and Ethernet, and then many control plant and manager minimize the needed course.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.14 no.4
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• pp.1976-1981
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• 2013

In this paper, we propose a frequency-domain interference cancellation algorithm for direct-sequence spread spectrum systems. In the previously proposed frequency-domain interference cancellation algorithms that protocol defines the rules concerning the collection of means of Transmission Control Protocol (TCP: Transmission Control Protocol) is the most widely used in the transport layer. Two-way traffic through the network path to the same end-to-end transfer of data in the opposite direction between pairs of nodes are infused with two or more TCP connection using the network traffic patterns from the exchanger and routers share results of approval. Per-flow input/output structure of matter using the LTS online reaction when evaluated as this is the most important factor. TCP-MT when the connection duration is one of the largest performance gains.

• Journal of information and communication convergence engineering
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• v.4 no.2
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• pp.71-74
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• 2006

A basic access method about IEEE 802.11 MAC layer protocol using IEEE 802.11 wireless LANs is the DCF thatis based on the CSMA/CA. But, cause of IEEE 802.11 MAC layer uses original backoff algorithm (exponential backoff method), when collision occurred, the size of contention windows increases the double size Also, a time of packet transmission delay increases and efficienty is decreased by original backoff scheme. In this paper, we have analyzed TCP packet transmission time of IEEE 802.11 MAC DCF protocol for wireless LANs a proposed enhanced backoff algorithm. It is considered the transmission time of transmission control protocol (TCP) packet on the orthogonal frequency division multiplexing (OFDM) in additive white gaussian noise (A WGN) and Rician fading channel. From the results, a proposed enhanced backoff algorithm produces a better performance improvement than an original backoff in wireless LAN environment. Also, in OFDM/quadrature phase shift keying channel (QPSK), we can achieve that the transmission time in wireless channel decreases as the TCP packet size increases and based on the data collected, we can infer the correlation between packet size and the transmission time, allowing for an inference of the optimal packet size in the TCP layer.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2006.05a
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• pp.530-533
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• 2006

Recently, a basic access method about IEEE 802.11 MAC layer protocol using IEEE 802.11 wireless LANs is the DCF thatis based on the CSMA/CA. But, cause of IEEE 802.11 MAC layer uses original backoff algorithm (exponential backoff method), when collision occurred, the size of contention windows increases the double size. Also, a time of packet transmission delay increases and efficiency is decreased by original backoff scheme. In this paper, we have analyzed TCP packet transmission time of IEEE 802.11 MAC DCF protocol for wireless LANs a proposed enhanced backoff algorithm. It is considered the transmission time of transmission control protocol (TCP) packet on the orthogonal frequency division multiplexing (OFDM) in additive white gaussian noise (AWGN) and Rician fading channel. From the results, a proposed enhanced backoff algorithm produces a better performance improvement than an original backoff in wireless LAN environment. Also, in OFDM/quadrature phase shift keying channel (QPSK), we can achieve that the transmission time in wireless channel decreases as the TCP packet size increases and based on the data collected, we can infer the correlation between packet size and the transmission time, allowing for an inference of the optimal packet size in the TCP layer.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.10
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• pp.4977-4996
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• 2016

The existence of excessively large and too filled network buffers, known as bufferbloat, has recently gained attention as a major performance problem for delay-sensitive applications. Researchers have made three types of suggestions to solve the bufferbloat problem. One is End to End (E2E) congestion control, second is deployment of Active Queue Management (AQM) techniques and third is the combination of above two. However, these solutions either seem impractical or could not obtain good bandwidth utilization. In this paper, we propose a Transmission Control Protocol（TCP）delayed window update mechanism which uses a congestion detection approach to predict the congestion level of networks. When detecting the network congestion is coming, a delayed window update control strategy is adopted to maintain good protocol performance. If the network is non-congested, the mechanism stops work and congestion window is updated based on the original protocol. The simulation experiments are conducted on both high bandwidth and long delay scenario and low bandwidth and short delay scenario. Experiment results show that TCP delayed window update mechanism can effectively improve the performance of the original protocol, decreasing packet losses and queuing delay while guaranteeing transmission efficiency of the whole network. In addition, it can perform good fairness and TCP friendliness.

• Journal of Satellite, Information and Communications
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• v.10 no.3
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• pp.53-56
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• 2015

In this paper, we want provide improved services with faster transmission, IEEE 802.11n was standardized. A-MPDU (Aggregation MAC Protocol Data UNIT) is a vital function of the IEEE 802.11n standard, which was proposed to improve transmission rate by reducing frame transmission overhead. In this paper, we show the problems of TCP retransmission with A-MPDU and propose a solution utilizing the property of TCP cumulative ACK. If the transmission of an MPDU subframe fails, A-MPDU mechanism allows selective re-transmission of failed MPDU subframe in the MAC layer. In TCP traffic transmission, however, a failed MPDU transmission causes TCP Duplicate ACK, which causes unnecessary TCP re-transmission. Furthermore, congestion control of TCP causes reduction in throughput. By supressing unnecessary duplicate ACKs the proposed mechanism reduces the overhead in transmitting redundant TCP ACKs, and transmitting only a HS-ACK with the highest sequence number. By using the RACK mechanism, through the simulation results, it was conrmed that the RACK mechanism increases up to 20% compared the conventional A-MPDU, at the same time, it tightly assures the fairness among TCP flows.

• Proceedings of the Korean Information Science Society Conference
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• 2005.07a
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• pp.334-336
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• 2005

최근 오디오나 비디오 스트리밍과 같은 멀티미디어 트래픽이 증가하고 있다. 이러한 트래픽들은 패킷을 전달하는데 대부분 UDP(User Datagram Protocol)기반의 RTP(Realtime Transport Protocol)를 사용한다. 하지만 UDP기반의 RTP는 기본적으로 혼잡제어 메커니즘이 없으며 현재 인터넷의 주요 트래픽인 TCP(Transmission Control Protocol)와의 형평성을 보장하지 않는다는 문제점을 갖는다. 본 논문에서는 스트리밍 트래픽의 TCP 친화적인 전송률 조절 기법으로 TF-RTP(TCP-Friendly RTP)를 제안하였다. TF-RTP는 네트워크 상태가 혼잡하여 패킷 손실이 발생할 경우, 개선된 파라미터들을 사용하여 경쟁하는 TCP의 전송률을 보다 정확하게 계산하여 스트리밍 트래픽의 전송률을 조절함으로써 경쟁하는 TCP 트래픽과 친화적으로 동작하며 네트워크 대역폭을 보다 공평하게 사용하게 된다. 실험을 통해 제안한 TF-RTP가 TCP의 전송률을 보다 정확하게 계산하며 TCP 친화성, 공평성 측면에서도 성능 개선을 보임을 확인할 수 있었다.