• Journal of Communications and Networks
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• v.4 no.2
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• pp.108-117
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• 2002

TCP, which was developed on the basis of wired links, supposes that packet losses are caused by network congestion. In a wireless network, however, packet losses due to data corruption occur frequently. Since TCP does not distinguish loss types, it applies its congestion control mechanism to non-congestion losses as well as congestion losses. As a result, the throughput of TCP is degraded. To solve this problem of TCP over wireless links, previous researches, such as split-connection and end-to-end schemes, tried to distinguish the loss types and applied the congestion control to only congestion losses; yet they do nothing for non-congestion losses. We propose a novel transport protocol for wireless networks. The protocol called VS-TCP (Variable Segment size Transmission Control Protocol) has a reaction mechanism for a non-congestion loss. VS-TCP varies a segment size according to a non-congestion loss rate, and therefore enhances the performance. If packet losses due to data corruption occur frequently, VS-TCP decreases a segment size in order to reduce both the retransmission overhead and packet corruption probability. If packets are rarely lost, it increases the size so as to lower the header overhead. Via simulations, we compared VS-TCP and other schemes. Our results show that the segment-size variation mechanism of VS-TCP achieves a substantial performance enhancement.

• Journal of IKEEE
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• v.10 no.1 s.18
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• pp.30-39
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• 2006

The TCP protocol can provide some reliability using sliding window mechanism for data transmission, flow control, and congestion control. However, TCP has some limitations in that it has basically been designed solely for wired communication environments. If traditional TCP protocol is used also in wireless networks, the end-to-end data transmission performance degrades dramatically due to frequent packet losses caused by transmission errors and hand-offs. While there have been some research efforts on TCP enhancements considering the mobility of wireless communication devices, in this paper we propose a new method to improve the TCP performance by combining the Snoop and the Freeze-TCP methods. In the proposed scheme, the TCP end-to-end semantics is maintained and no changes of existing protocols in sending systems or in routers are required. It has the advantage of simple implementation because TCP code changes are limited to mobile devices for applying the Freeze-TCP and it requires only to add Snoop modules in base stations. Accordingly, the proposed scheme can operate well in the existing networks. Finally, in this study, we compared the performance of the proposed scheme with traditional TCP, other approaches through simulations using ns-2.

• Proceedings of the Korea Institutes of Information Security and Cryptology Conference
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• 2003.07a
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• pp.121-125
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• 2003

In this paper, we propose a framework of Transmission Control Protocol with Software Rejuvenation methodology, which is applicable for network system survivability. This method is utilized to improve the survivability because it can limit the damage caused by successful attacks. The main objectives are to detect intrusions in real time, to characterize attacks, and to survive in face of attacks. To counter act the attacks' attempts or intrusions, we perform the Software Rejuvenation methods such as killing the intruders' processes in their tracks, halting abuse before it happens, shutting down unauthorized connection, and responding and restarting in real time. These slogans will really frustrate and deter the attacks, as the attacker can't make their progress. This is the way of survivability to maximize the deterrence against an attack in the target environment. We address a framework to model and analyze the critical intrusion tolerance problems ahead of intrusion detection on Transmission Control Protocol (TCP).

• Proceedings of the IEEK Conference
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• 2003.11a
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• pp.255-258
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• 2003

TCP is taking over 95％ among the Internet traffics. Recently the demands of multimedia services in the Internet has been increasing. These multimedia services mostly need real-time deliverly, and then RTP has been a de facto to transmission protocol for these real-time multimedia services. RTP uses UDP as its underlying transport protocol, and thus it does not support any rate and congestion control. Thus, for fair use of the Internet bandwidth with TCP traffics. RTP also needs a rate control. One constraint of RTP is that the feedback information(delivered by, RTP's twin protocol, RTCP) is recommended to be sent no less than 5 seconds. In this paper, we propose a TCP-friendly RTP rate control which use only RTCP feedback information at every 5 seconds. The experiment results show that our proposed algorithm works. But, it is found that we need more time to test the effects of parameters and policies of the algorithms, which will be reported later.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.1
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• pp.129-137
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• 2016

Transmission Control Protocol as a transport layer protocol provides steady data transfer service. There are some serious concerns about the performance of TCP over diverse networks. The vital concern in TCP network environment is congestion which may occur due to quick transmission rates or because of large number of new connections entering the network at the same time. Size of queues in routers grows thus resulting in packet drops. Retransmission of the dropped packets, and reduced throughput can prove costly. Explicit Congestion Notification (ECN) in conjunction with Active Queue Management mechanisms (AQM) such as Random early detection (RED) is used for packet marking rather than dropping. In IP packet header ECN bits can be added as a sign of congestion thus avoiding needless packet drops. The proposed ECN and AQM mechanism can be implemented with help of ns2 simulator and the performance can be tested on different TCP variants.

• Journal of Communications and Networks
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• v.12 no.5
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• pp.499-509
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• 2010

A plethora of transmission control protocol (TCP) congestion control algorithms have been devoted to achieving the ultimate goal of high link utilization and fair bandwidth sharing in high bandwidth-delay product (HBDP) networks. We present a new insight into the TCP congestion control problem; in particular an end-to-end delay-based approach for an HBDP network. Our main focus is to design an end-to-end mechanism that can achieve the goal without the assistance of any network feedback. Without a router's aid in notifying the network load factor of a bottleneck link, we utilize goodput and throughput values in order to estimate the load factor. The obtained load factor affects the congestion window adjustment. The new protocol, which is called TCP-goodput and throughput (GT), adopts the carefully designed inversely-proportional increase multiplicative decrease window control policy. Our protocol is stable and efficient regardless of the link capacity, the number of flows, and the round-trip delay. Simulation results show that TCP-GT achieves high utilization, good fairness, small standing queue size, and no packet loss in an HBDP environment.

• Information and Communications Magazine
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• v.32 no.9
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• pp.31-36
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• 2015

TCP (Transmission Control Protocol)는 IP (Internet Protocol)와 더불어 인터넷 핵심 프로토콜 중 하나로써, 현재 Web, e-mail, FTP 등 대다수의 애플리케이션에서 사용되고 있는 연결 지향적이고 신뢰성 전송을 보장하는 전송 계층 프로토콜이다. 유/무선 네트워크가 진화함에 따라 TCP 역시 보다 나은 서비스를 제공하기 위해 많은 연구가 이루어졌고, 이를 통해 최근에는 스트리밍과 같이 지연이 중요한 애플리케이션에서도 많이 사용되고 있다. 본 논문은 모바일 데이터가 폭증함에 따라 무선 네트워크에서의 TCP 성능이 큰 관심을 받고 있는 현 시점에서 최근에 이슈가 되고 있는 무선 네트워크에서 bufferbloat문제의 심각성과 이를 해결하기 위한 방안에 대해 다룬다.

• Proceedings of the Korea Information Processing Society Conference
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• 2004.05a
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• pp.1355-1358
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• 2004

This paper describes a modification to the SACK (Selective Acknowledgement) Transmission Control Protocol's (TCP), called SACK TCP with Probing Device, SACK works in conjunction with Probing Device, for improving SACK TCP performance when more than half a window of data lost that is typical in handoff as well as unreliable media. It shows that by slightly modifying the congestion control mechanism of the SACK TCP, it can be made to better performance to multiple packets lost from one window of data.

• Journal of Digital Contents Society
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• v.16 no.3
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• pp.381-388
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• 2015

TCP, which performs congestion control in congestion condition, is able to help a reliable transmission. However, packet loss can be increased because congestion window is increased by the time the packet is dropped in the process of congestion avoidance. In this paper, to solve the above problem, we propose a new congestion estimation based TCP congestion control scheme using the weighted average value of the RTT. After measuring a SRTT, which means the weighted average value of RTTs, at this point of time when a buffer overflow is occurred by an overloaded packet, the proposed scheme estimates the time, when the same SRTT is made in packet transmission, as a congestion time and then decreases the congestion window. The simulation results show that the proposed schem has a good performance in terms of packet loss rate and throughput when the packet loss due to buffer overflow is larger than that due to wireless channel.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.9
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• pp.2323-2340
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• 2012

Recent advances in cognitive radio technology have drawn immense attention to higher layer protocols above medium access control, such as transmission control protocol (TCP). Most proposals to improve the TCP performance in cognitive radio (CR) networks have assumed that either all nodes are in CR networks or the TCP sender side is in CR links. In those proposals, lower layer information such as the CR link status could be easily exploited to adjust the congestion window and improve throughput. In this paper, we consider a TCP network in which the TCP sender is located remotely over the Internet while the TCP receiver is connected by a CR link. This topology is more realistic than the earlier proposals, but the lower layer information cannot be exploited. Under this assumption, we propose an enhanced TCP protocol for CR networks called TCP for cognitive radio (TCP-CR) to improve the existing TCP by (1) detection of primary user (PU) interference by a remote sender without support from lower layers, (2) delayed congestion control (DCC) based on PU detection when the retransmission timeout (RTO) expires, and (3) exploitation of two separate scales of the congestion window adapted for PU activity. Performance evaluation demonstrated that the proposed TCP-CR achieves up to 255% improvement of the end-to-end throughput. Furthermore, we verified that the proposed TCP does not deteriorate the fairness of existing TCP flows and does not cause congestions.