• 대한공업교육학회지
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• v.33 no.1
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• pp.233-248
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• 2008

Stream Control Transmission Protocol(SCTP) is a new transport protocol that is known to provide improved performance than Transmission Control Protocol(TCP) in multi-homing environment that is having two and more IP addresses. But currently single-homed computer is used primarily that is having one IP address. To identify whether mean transfer time for SCTP is faster that for TCP in single-homed environment considering packet loss, we make up real testbed regulating the bandwidth, delay time and packet loss rate on router and observe the transfer time. We write server and client applications to measure SCTP and TCP mean transfer time by C language. Analysis of these experimental results from the testbed implementation shows that mean transfer time of SCTP is not better than performance of TCP in single homed environment exceptional case. Main reasons of performance are that SCTP compared to TCP stops transmitting data by timeout and data transmission is often delayed when SACK congestion happens. The result of study shows that elaborate performance tuning is required in developing a new SCTP module or using a implemented SCTP module.

• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.6
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• pp.1194-1201
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• 2003

Measurements of network traffic have shown that self-similarity is a ubiquitous phenomenon spanning across diverse network environments. We have advance the framework of multiple time scale congestion control and show its effectiveness at enhancing performance for fast TCP prototype control. In this paper, we extend the fast TCP prototype control framework to window-based congestion control, in particular, TCP. This is performed by interfacing TCP with a large time scale control module which adjusts the aggressiveness of bandwidth consumption behavior exhibited by TCP as a function of "large time scale" network state. i.e., conformation that exceeds the horizon of the feedback loop as determined by RTT. Performance evaluation of fast TCP prototype is facilitated by a simulation bench-mark environment which is based on physical modeling of self-similar traffic. We explicate out methodology for discerning and evaluating the impact of changes in transport protocols in the protocol stack under self-similar traffic conditions. We discuss issues arising in comparative performance evaluation under heavy-tailed workload. workload.

• International Journal of Computer Science & Network Security
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• v.22 no.1
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• pp.276-282
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• 2022

One of the most essential foundations upon which big institutions rely in delivering cloud computing and hosting services, as well as other kinds of multiple digital services, is the security of infrastructures for digital and information services throughout the world. Distributed denial-of-service (DDoS) assaults are one of the most common types of threats to networks and data centers. Denial of service attacks of all types operates on the premise of flooding the target with a massive volume of requests and data until it reaches a size bigger than the target's energy, at which point it collapses or goes out of service. where it takes advantage of a flaw in the Transport Control Protocol's transmitting and receiving (3-way Handshake) (TCP). The current study's major focus is on an architecture that stops DDoS attacks assaults by producing code for DDoS attacks using a cloud controller and calculating Round-Tripe Time (RTT).

• The KIPS Transactions:PartC
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• v.9C no.6
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• pp.865-874
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• 2002

HTTP (HyperText Transfer Protocol) is a transfer protocol used by the World Wide Web distributed hypermedia system to retrieve the objects. Because HTTP is a connection oriented protocol, it uses TCP (Transmission control Protocol) as a transport layer. But it is known that HTTP interacts with TCP badly. it is discussed about factors affecting the performance or HTTP over TCP, the transaction time obtained by the per-transaction TCP connections for HTTP access and the TCP slow-start overheads, and the transaction time for T-TCP (Transaction TCP) which is one or methods improving the performance or HTTP over TCP. Average waiting time and average turnaround time are important parameters to satisfy QoS (Quality of Service) of end users. Formulas for calculating two parameters are derived. Such formulas can be used for the environment in which each TCP or T-TCP transaction time is same or different. Some experiments and computational experiences indicate that the proposed formulas are well acted, can be applied to the environment which the extension of bandwidth is necessary, and time characteristics of T-TCP are superior to that of TCP. Also, the load distribution method of web server based on the combination of bandwidths is discussed to reduce average waiting time and average turnaround time.

• Journal of Korea Game Society
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• v.2 no.1
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• pp.23-29
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• 2002

Multicast protocols are efficient methods of group communication such as video conference, Internet broadcasting and On-Line Game, but they do not support the various transmission protocol services like a reliability guarantee, FTP, or Telnet that TCPs do. The Purpose or this Paper is to find a method to utilize multicast routers can simultaneously transport multicast packets and TCP packets. For multicast network scalability and error recovery the existing SRM(Scalable Reliable Multicast)method has been used. Three packets per TCP transmission control window site are used for transport and an ACK is used for flow control. A CBR(Constant Bit Rate) and a SRM is used for UDP traffic control. Divided on whether a UDP multicast packet and TCP unicast packet is used simultaneously or only a UDP multicast packet transport is used, the multicast receiver with the longest delay is measured on the number of packets and its data receiving rate. It can be seen that the UDP packet and the TCP's IP packet can be simultaneously used in a server router.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• 2004.04a
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• pp.43-48
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• 2004

RTP that is proposed supplement of real-time services on internet environment, as Real-time Transport Protocol, is the protocol that for the purpose of sending data of stream type. RTP and RTCP(Real-time Transport Control Protocol) basically work at the same time, RTCP serves with state information of network at present. RTP has important properties of a transport protocol that runs on end-to-end systems and provides demultiplexing. It also offer reliability and protocol-defined flow／congestion control that transport protocol like TCP can not provides. In this paper, we look around concept and construction of Differentiated sen1ice tint run on RTP and by setting parameters of packet transfer method be used CBQ(Class-Based Queuing) for packet transfer on Differentiated service, each service queue controls properly through packet scheduling method, such as WRR(Weighted Round Robin) and PRR(Packet-by-packet Round Robin) all service classes do not experience the starvation and confirm the performance through computer simulation to achieve fairly scheduling.

• Journal of KIISE:Information Networking
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• v.30 no.5
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• pp.649-655
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• 2003

Network-based computing such as cluster computing requires a reliable high-speed transport protocol. TCP is a representative reliable transport protocol on the Internet, which implements many mechanisms, such as flow control, congestion control, and retransmission, for reliable packet delivery. This paper, however, finds out that Myrinet does not incur any packet losses caused by network congestion. In addition, we ascertain that Myrinet supports reliable and ordered packet delivery. Consequently, most of reliable routines implemented in TCP produce unnecessarily additional overheads on Myrinet. In this paper, we show that we can attain the reliability only by flow control on Myrinet and propose a new reliable protocol based on UDP named RUM (Reliable UDP on Myrinet) that performs a flow control. As a result, RUM achieves a higher throughput by 45% than TCP and shows a similar one-way latency to UDP.

• 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.

• 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.