• 대한공업교육학회지
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• v.32 no.2
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• pp.199-216
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• 2007

Stream Control Transmission Protocol(SCTP) is a transport layer protocol to support the data transmission. SCTP is similar to Transmission Control Protocol(TCP) in a variety of aspects. However, several features of SCTP including multi-homing and multi-streaming incur the performance difference from TCP. This paper highlights the data transfer during the initial slow start phase in SCTP congestion control composed of slow start phase and congestion avoidance phase. In order to compare the mean transfer time between SCTP and TCP, we experiment with different performance parameters including bandwidth, round trip time, and data length. By varying data length, we also measure the corresponding initial window size, which is one of factors affecting the mean transfer time. For the experiment, we have written server and client applications by C language using SCTP socket API and have measured the transfer time by ethereal program. We transferred data between client and server using round-robin method. Analysis of these experimental results from the testbed implementation shows that larger initial window size of SCTP than that of TCP brings the reduction in the mean transfer time of SCTP compared with TCP by 15 % on average during the initial slow start phase.

• Proceedings of the Korean Information Science Society Conference
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• 2005.07a
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• pp.361-363
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• 2005

최근 컴퓨터 기술의 발전과 더불어 급속하게 발전하는 네트워크 기술이 실생활에 보급되고 네트워크를 통한 멀티미디어 데이터의 교환이나 전송과 같은 서비스들이 활성화 되면서 멀티미디어 데이터가 점점 대용량화 다양화되고 있다. 기존의 멀티미디어 서비스는 정해진 서버로부터 미리 데이터를 받아서 보는 다운로드 서비스가 대부분이었고 스트리밍 서비스라 할지라도 사용자의 기호나 원하는 요구사항에는 미치지 못 하였다. 지금까지 스트리밍 데이터를 전송 하는 프로토콜로 주로 UDP(User Datagram Protocol)를 사용하였다. 하지만 UDP는 혼잡제어를 하지 않으며 현재 인터넷의 주요 트래픽인 TCP(Transmission Control Protocol) 트래픽은 혼잡제어를 한다. 그래서 UDP에 의한 스트리밍 서비스는 TCP 트래픽의 전송률을 저하시키며 더 나아가 네트워크의 전체의 성능을 저하시키는 요인이 될 수 있다. 본 논문에서는 IETF(Internet Engineering Task Force)에서 제정한 실시간 스트리밍 데이터 서비스를 위한 표준인 RTP(Real-Time Transport Protocol)/RTCP(Real-Time Transport Control Protocol)를 적용하여 RTCP의 정보를 가지고 현재 네트워크 상태를 판단하고 스트리밍 서비스를 할 때 데이터의 전송률은 TCP 친화적인 전송률로 조절하는 스트리밍 서비스를 구현하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.5B
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• pp.465-475
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• 2004

Transport Control Protocol (TCP) has been implemented in software running on CPU in end systems, and the protocol processing has appeared as a new bottleneck due to advanced link technology. TCP processing is a critical issue in Storage Area Network (SAN) such as iSCSL, and the overall performance of the Storage Area Network heavily depends on speed of TCP processing. TCP Engine implemented in hardware reduces the load of CPU in end systems as well as accelerates the protocol processing, and hence high speed data processing is achieved. In this paper, we have proposed a hardware engine for TCP processing. TCP engine consists of three major block, TCP Connection block Rx TCP block and Tx TCP block TCP Connection block is responsible for managing TCP connection states. Rx TCP block is responsible for receive flow which receives packets from network and sends to CPU. Rx TCP performs header and data processing and sends header information to TCP connection block and Tx TCP block It also assembles out-of-ordered data to in-ordered before it transfers data to CPU. Tx TCP block is responsible for transmit flow which transfers data from CPU to network. Tx TCP performs retransmission for reliable data transfer and management of transmit window and sequence number. Various test-cases are used to verify the TCP functions. The TCP Engine is synthesized using 0.18 micron technology and results in 51K gates not including buffers for temporal data storage.

• The KIPS Transactions:PartA
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• v.13A no.7 s.104
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• pp.573-580
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• 2006

The conventional transport layer protocol TCP is designed to work under condition of packet loss is due to the network congestion, so that it's suitable in the traditional wired network with fixed hosts but it's inefficient on the wireless network where the environment of fading, noise, and transmission error comes from interference. This result from the needless transmission control of the bit error is due to treats the packet loss as a packet congestion control in the wireless network. In this paper, we propose the advanced SNOOP protocol with the consecutive packet loss and TCP window control to avoid the needless congestion management algorithm in wireless network for the wireless TCP packet transmission enhancement. We verify the performance of the advanced module from the simulation experiment result.

• The Journal of the Korea Contents Association
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• v.9 no.8
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• pp.73-83
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• 2009

TCP(Transmission Control Protocol) is currently used connection-oriented protocol as a typical transport layer protocol in the Internet. However, it has deficiency not be able to communicate with other TCP entities when any link included in the path is down because of single-homing on single path. SCTP(Stream Control Transmission Protocol) suggested as the new transport layer protocol supports multi-homing feature, which provides several paths between source and destination. It can communicate with other SCTP entities using alternate path even when any link on the primary path is down. This paper aims to measure and analyze the multi-homing effect of SCTP over TCP in case of link-down using NS-2 simulator. We classify SCTP into $SCTP_{single-homing}$ and $SCTP_{multi-homing}$ because SCTP with single-homing can also be used like TCP. We measured throughput and bandwidth utilization varying link-down duration, bandwidth, and RTT(round trip time), Simulation results show that throughput of $SCTP_{multi-homing}$ is more than that of TCP by 18 % on average. It is also shown that $SCTP_{multi-homing}$ on varying RTT and bandwidth increases the throughput of TCP by 'l7% and 9% on average, respectively in the link-down environment. In above cases, more or less difference between $SCTP_{single-homing}$ and TCP on throughput and bandwidth utilization was found To summarize, multi-homing effect of SCTP over TCP on throughput is about 18 % on average in the link-down environment This experimental result can be used as the benchmark in order to estimate the multi-homing effect of SCTP over TCP when the link-down happens in the real Internet.

• The KIPS Transactions:PartC
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• v.11C no.2
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• pp.245-252
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• 2004

Stream Control Transmission Protocol(SCTP) is a new connection-oriented, reliable delivery transport protocol operating on top of an unreliable connectionless packet service such as IP. It inherits many of the functions developed for TCP, including flow control and packet loss recovery functions. In addition, it also supports transport layer multihoming and multistreaming In this paper, we study the impact of multi-homing on the performance of SCTP. We first compare performance of single-homed SCTP. multi-homed SCTP, TCP Reno and TCP SACK. We, then describe potential flaw in the current SCTP retransmission policy, when SCTP host is multihomed. Our Results show that SCTP performs better than TCP Reno and TCP SACK due to several changes from TCP in its congestion control mechanism. In particular. multi-homed SCTP shows the best result among the compared schemes. Through experimentation for multi-homed SCTP, we found that the current SCTP retransmission policy nay deteriorate the perfomance when the retransmission path it worse than the original path. Therefore, the condition of retransmission path is a very important factor In SCTP performance and a proper mechanism would be required to measure the condition of the retransmission path.

• Journal of Advanced Navigation Technology
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• v.9 no.1
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• pp.61-69
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• 2005

TCP and UDP is a transport layer protocol of Internet. TCP is a connection oriented protocol which supports a reliable data transfer by offering error and flow control, but it bring a transmission delay. On the other hand, the UDP is a connectionless protocol which does not carry out error and flow control, but it guarantees a realtime transmission. There are hardly any protocols which supports not only realtime functions but also data reliability. In this paper, we have designed and implemented a new TCP mode option which supports reliable realtime transmission. Our designed TCP performs an error recovery process during a fixed amount of time. This time is negotiated during the connection establishment phase. Our designed TCP is tested in real environments, and we find that it is relatively faster than the standard TCP and more reliable than the UDP. It can be used for the reliable transfer of realtime multimedia data.

• Proceedings of the KIEE Conference
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• 2006.04a
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• pp.105-107
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• 2006

As internet streaming data increase, transport protocol such as TCP, TGP-Friendly is important to study control transmission rate and share of Internet bandwidth. In this paper, we propose a TCP-Friendly protocol using Neural Network for media delivery over wired Internet which has various traffic size(PTFRC). PTFRC can effectively send streaming data when occur congestion and predict one-step ahead round trip time and packet loss rate. A multi-layer perceptron structure is used as the prediction model, and the Levenberg-Marquardt algorithm is used as a traning algorithm. The performance of the PTFRC was evaluated by the share of Bandwidth and packet loss rate with various protocols.

• Journal of Institute of Control, Robotics and Systems
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• v.18 no.8
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• pp.774-779
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• 2012

We design and implement a RTT (Round Trip Time) based TCP (Transmission Control Protocol) for USN (Ubiquitous Sensor Network). We adopt a basic update algorithm for window size from FAST TCP that uses the queuing delay at link as the congestion measure. The designed TCP estimates the queuing delay at link from the measured RTT in the network layer, and updates the window size based on the estimated queuing delay. The designed TCP allows to utilize the full capacity of USN links and avoids the waste of the given link capacity that is common without the flow control in the transport layer. The experiment results show that the window size of the sender converges within a small range of variations without any packet loss, and verify the stability and performance of the designed TCP.

• 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.