• Journal of KIISE:Information Networking
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• v.29 no.5
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• pp.495-500
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• 2002

We propose a new estimation method for rate adjustment in the face of a packet loss in the TFRC protocol, a TCP-Friendly congestion control protocol for UDP flows. Previous methods respond in a sensitive way to a single packet loss, resulting in oscillatory transmission behavior. This is an undesirable for multimedia services demanding constant bandwidth. The proposed TFRC provides more smooth and fair (against TCP flows) transmission through collective response based on multiple packets loss events. We show our "Exponential smoothing method" performs better than known "Weight smoothing method" in terms of smoothness and fairness.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.12B
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• pp.1106-1114
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• 2003

Several analytic models describe transmission control protocol (TCP) performance such as steady-state throughput as an averaged ratio of number of transmissions to latency. For more detailed analysis of TCP latency, the latency during packet losses are recovered should be considered. In this paper, we derive the expected duration of loss recovery latency considering the number of packet losses recovered by retransmissions. Based on the numerical results verified by simulations, TCP using selective acknowledgement (SACK) option is more effective than TCP NewReno from the aspect of loss recovery latency.

• Journal of IKEEE
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• v.2 no.1 s.2
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• pp.42-52
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• 1998

This paper proposes an extended TCP/IP protocol, LAN/TCP, that can be used for the distributed soft real-time systems connected through the Ethernet-based local area network. Since LAN/TCP shows soft real-time performance with keeping compatibility and interoperability with the standard TCP/IP, the existing application software can be used without any modification. LAN/TCP also provides the periodic transmission mode(PTM) with which the periodical data collection and updating the control signals can be efficiently implemented with relatively small traffic overhead. This paper includes the computer simulation and experimental results of the proposed protocol.

• KIPS Transactions on Computer and Communication Systems
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• v.10 no.5
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• pp.137-144
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• 2021

The IETF has recently been standardizing the QUIC protocol for HTTP/3 services. It is noted that HTTP/3 uses QUIC as the underlying protocol, whereas HTTP/1.1 and HTTP/2 are based on TCP. Differently from TCP, the QUIC uses 0-RTT or 1-RTT transmissions to reduce the connection establishment delays of TCP and SCTP. Moreover, to solve the head-of-line blocking problem, QUIC uses the multi-streaming feature. In addition, QUIC provides various features, including the connection migration, and it is available at the Chrome browser. In this paper, we analyze the performance of QUIC for HTTP-based web and streaming services by comparing with the existing TCP and Streaming Control Transmission Protocol (SCTP) in the network environments with different link delays and packet error rates. From the experimental results, we can see that QUIC provides better throughputs than TCP and SCTP, and the gaps of performances get larger, as the link delays and packet error rates increase.

• Electronics and Telecommunications Trends
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• v.18 no.3 s.81
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• pp.11-20
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• 2003

최근 SCTP(Stream Control Transmission Protocol)는 TCP/UDP 이후의 차세대 수송계층 프로토콜로서 주목 받고 있다. SCTP는 기존 TCP 및 UDP의 문제점을 극복하도록 설계되었으며 특히 multi-streaming 및 multi-homing 특성을 제공한다. 본 고에서는 SCTP 프로토콜의 기본 특징에 대하여 알아보고, 현재 논의중인 확장작업의 주요 골자를 살펴본다.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.1 no.2
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• pp.131-136
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• 2008

From the reciprocal connection by this Internet network researchs about the efficiency improvement of the whole system is accomplished with the method which reduces delays in message transmission. From here, we will do a comparative study between the user data program protocol (UDP) and the zero copy which does not use the buffer cache to fine out the valid method to improve the efficiency. In this thesis, I will change the message copy from execution process of the buffer cache of the TCP/IP on Unix OS with process on Linux OS. The object of conversion is to show you that the zero copy which doesn't use the buffer cache from transfer control class improves the communication efficiency.

• The KIPS Transactions:PartC
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• v.17C no.6
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• pp.451-458
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• 2010

To detect and recover packet losses over wireless links is very important in terms of reliability in packet transmission. Most wireless communication systems adopt an automatic repeat request (ARQ) protocol operating at link layer. However, it has been constantly addressed that the interaction not harmonized sufficiently between ARQ and TCP rather degrades TCP performance. In this paper, therefore, we propose an improved scheme from the aspect of the interaction with TCP loss recovery mechanism that can be applied to selective repeat ARQ (SR-ARQ) protocol and prove that the proposed scheme improves TCP performance significantly by OPNET simulations.

• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.297-312
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• 2005

The multimedia traffic of continuous video and audio data via streaming service accounts for a significant and expanding portion of the Internet traffic. This streaming data delivery is mostly based on RTP with UDP. However, UDP does not support congestion control. For this reason, UDP causes the starvation of congestion controlled TCP traffic which reduces its bandwidth share during overload situation. In this paper, we propose a new TCP-friendly rate control scheme called 'TF-RTP(TCP-Friendly RTP)'. In the congested network state, the TF-RTP exactly estimates the competing TCP's throughput by using the modified parameters. Then, it controls the sending rate of the video streams. Therefore, the TF-RTP adjusts its sending rate to TCP-friendly and fair share with competing TCP traffics. Through the simulation, we prove that the TF-RTP correctly estimates the TCP's throughput and improves the TCP-friendliness and fairness.

• The KIPS Transactions:PartC
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• v.9C no.2
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• pp.197-212
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• 2002

Transmission Control Protocol (TCP) [1] has been tuned as a reliable transfer protocol for traditional networks comprising wired links and stationary hosts with same link characteristics. TCP assumes that congestion in the network be a primary cause for packet losses and unusual delays. TCP performs welt over such networks adapting to end-to-end delays and congestion losses, by standard congestion control mechanisms, such as slow-start, congestion avoidance, fast retransmit and recovery. However, networks with wireless and other lossy links suffer from significant losses due to bit errors and handoffs. An asymmetry network such as ADSL has different bandwidth for both directions. As a result, TCP's standard mechanisms incur end-to-end performance degradation in various links. In this paper, we analyze the TCP problems in wireless, satellite, and asymmetry links, and measure the new TCP mechanisms that are recommended by IETF Performance Implications of Link Characteristics (PILC) WG[2], by using Network Simulator 2 (NS-2).

• Journal of Communications and Networks
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• v.7 no.3
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• pp.385-393
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• 2005

In this paper, we propose a simple algorithm to adaptively adjust the value of dupthresh, the duplicate acknowledgement threshold that triggers the transmission control protocol (TCP) fast retransmission algorithm, to improve the TCP performance in a network environment with persistent packet reordering. Our algorithm uses an exponentially weighted moving average (EWMA) and the mean deviation of the lengths of the reordering events reported by a TCP receiver with the duplicate selective acknowledgement (DSACK) extension to estimate the value of dupthresh. We also apply an adaptive upper bound on dupthresh to avoid the retransmission timeout events. In addition, our algorithm includes a mechanism to exponentially reduce dupthresh when the retransmission timer expires. With these mechanisms, our algorithm is capable of converging to and staying at a near-optimal interval of dupthresh. The simulation results show that our algorithm improves the protocol performance significantly with minimal overheads, achieving a greater throughput and fewer false fast retransmissions.