• KSII Transactions on Internet and Information Systems (TIIS)
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• v.2 no.6
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• pp.333-351
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• 2008

Transmission control protocols have to overcome common problems in wireless networks. TCP employing both packet loss discrimination mechanism and available bandwidth estimation algorithm, known as the good existing solution, shows significant performance enhancement in wireless networks. For instance, TCP New Jersey which exhibits high throughput in wireless networks intends to improve TCP performance by using available bandwidth estimation and congestion warning. Even though it achieves 17% and 85% improvements in terms of goodput over TCP Westwood and TCP Reno, respectively, we further improve it by exploring maximized available bandwidth estimation, handling bit-error-rate error recovery, and effective adjustment of sending rate for retransmission timeout. Hence, we propose TCP NJ+, showing that for up to 5% packet loss rate, it outperforms other TCP variants by 19% to 104% in terms of goodput when the network is in bi-directional background traffic.

• The Journal of the Korea institute of electronic communication sciences
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• v.16 no.6
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• pp.1161-1174
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• 2021

UDP is an unreliable data transmission protocol, in contrast to TCP, which implements reliable data delivery based on flow control and retransmission methods. However, the TCP operation algorithm for guaranteeing reliability is inefficient in improving transmission performance, and above all, there is no need to transmit data using the TCP method for fields that do not require perfect integrity. In this paper, we intend to discuss ways to improve the stability while maintaining the existing performance of UDP. To this end, a program applied with packet buffering and relaying techniques was developed, and the performance and stability of the experiment were verified.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.1
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• pp.139-145
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• 2016

Dumbbell is the most basic topology that can be used in almost all kind of network experiment within it or just by little expansion. While Transmission Control Protocol TCP is the basic protocol that is used for the connectivity among networks and stations. TCP major and basic goal is to provide path and services to different applications for communication. For that reason TCP has to transfer a lot of data through a communication medium that cause serious congestion problem. To calculate the congestion problem, different kind of pre-cure solutions are developer which are Loss Based Variant and Delay Based Variant. While LBV keep track of the data that is going to be passed through TCP protocol, if the data packets start dropping that means congestion occurrence which notify as a symptom, TCP CUBIC use LBV for notifying the loss. Similarly the DBV work with the acknowledgment procedure that is used in when data ACK get late with respect to its set data rate time, TCP COMPOUND/VAGAS are examples of DBV. Many algorithms have been purposed to control the congestion in different TCP variants but the loss of data packets did not completely controlled. In this paper, the congestion control algorithms are implemented and corresponding results are analyzed in Dumbbell topology, it is typically used to analyze the TCP traffic flows. Fairness of throughput is evaluated for different TCP variants using network simulator (NS-2).

• Proceedings of the Korea Information Processing Society Conference
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• 2006.05a
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• pp.1097-1100
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• 2006

최근 네트워크 기술이 발전하면서 사용자들의 요구가 다양화되고 대용량의 멀티미디어 데이터에 대한 필요성이 증가하고 있다. 실시간성과 동시성이 중요시 되는 멀티미디어 데이터의 전송에는 UDP(User Datagram Protocol)가 사용되고 있는데, UDP는 현재의 인터넷 기반인 TCP (Transmission Control Protocol )와 경쟁관계에 있어 네트워크의 혼잡을 초래하는 경우가 많았다. 또한 TCP 트랙픽은 네트워크의 혼잡제어를 수행하지만 수신자의 관점을 고려하지 않은 점이 있어서 스트리밍 전송에는 적합하지 않은 단점을 가지고 있다. 본 논문에서는 UDP(User Datagram Protocol) 트래픽의 네트워크 형평성 문제를 해결하고 수신자의 관점을 고려하기 위해 네트워크에 우선순위를 적용하여 RTP/RTCP를 기반으로 수신자의 버퍼를 고려한 혼잡제어 알고리즘을 제안하였다. 시뮬레이션 결과를 통해 우선 순위가 있는 네트워크가 우선 순위가 없는 네트워크에 비해 패킷 손실률과 버퍼 점유도 면에서 스트리밍 전송에 적합한 환경을 제공하고 있음을 확인할 수 있었다.

• Journal of Institute of Control, Robotics and Systems
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• v.13 no.7
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• pp.694-702
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• 2007

The SSWM(Software Switching Mechanism) of I-processor concept using non-real time in-house software simulation program is an effective method in order to develop the flight control law in desktop or HQS environment. And, this system has some advantages compare to HSWM(Hardware Switching Mechanism) such as remove the time delay effectiveness and reduce the costs of development. But, if this system loading to the OFP(Operational Flight Program), the OFP guarantee the enough throughput in order to calculate the two control law at once. Therefore, the HSWM(Hardware Switching Mechanism) of 2-processor concept is necessary. This paper addresses the concept of HSWM of the HQS-PC interface using TCP/IP(Transmission Control Protocol/Internet Protocol) communication based on flight control law of advanced supersonic trainer. And, the fader logic of TFS(Transient Free Switch) and stand-by mode of reset '0' type are designed in order to reduce the abrupt transient response and minimize the integrator effect in pitch axis. The result of the analysis based on HQS pilot simulation using HSWM reveals that the flight control systems are switching between two computers without any problem.

• Smart Media Journal
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• v.2 no.4
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• pp.51-59
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• 2013

With increasing transmission speed of backbone networks, it is getting to provide enough available bandwidth. However, the bandwidth is not effectively utilized in volumetric data transfer. This mainly comes from the transmission protocol, TCP, which is used for most applications. TCP is inherently difficult to adapt the available bandwidth because of it's own characteristic of transfer mechanism. UDT is a prominent application level data transfer protocol which is targeting high speed network. In this paper, we propose UDT parallel transfer technologies which is adaptive to network status and then evaluate their performance in two points of view. Firstly, we measure data transfer rate of UDT with rate congestion control methods, and compare them with basic UDT. Secondly, we apply parallel transfer technologies adapted to network status, and measure their performance. Experimental results showed that UDT rate congestion control method outperforms UDT with 106% improvement in RTT 100ms section set with jitter 30ms. In addition, performance of parallel transfer with rate congestion control method showed 107% improvement than that of parallel transfer in RTT 400ms section set with jitter 20ms.

• Proceedings of the IEEK Conference
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• 2003.07d
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• pp.1629-1632
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• 2003

Generally an increasing number of recent applications have found TCP too limiting. There are some characteristics in the transmission of document and binary data which some transmission delay are tolerant but the content must completely be transferred. However voice signals are more sensitive with not some packet loss but some transmission delay. Therefore, Stream Control Transmission Protocol(SCTP) is proposed to minimize the delay and packet loss in the field of delivery of voice signal. SCTP is designed to transport PSTN signalling messages over IP networks, but is capable of broader applications. In this paper, the architecture of SCTP implementation is designed and some interface of SCTP software library which are implemented are specified.

• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.45-50
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• 2008

Many applications of sensor network require connection to the Internet. The transmission protocol of traditional sensor network was designed within the sensor network itself. However, based on 6LoWPAN which can be accessed using IPv6, direct connection is possible between the sensor network and the TCP/IP network outside. Transmission of data in applications of sensor network falls into two main categories. One is a small packet that is periodically produced such as packet related to temperature and humidity. The other is a relatively large packet that brings about network overheads such as images. We investigated the conformance test and pros and cons of application data over the transmission protocol of Zigbee and 6LoWPAN. As a result, both Zigbee and 6LoWPAN have shown low rate of loss for periodic data and have in creased reliability of data transfer. When transmitting streaming image data, both ACK, non ACK mode of Zigbee and UDP of 6LoWPAN minimized transmission time but suffered the consequences of high packet loss. Even though TCP of 6LoWPAN required a long transmission time, we were able to confirm that no loss has occurred.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.04b
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• pp.1483-1486
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• 2002

Multicast protocols are efficient methods of group communication, but they do not support the various transmission protocol services like a reliability guarantee, FTP, or Telnet that TCPs do. The purpose of this dissertation is to find a method to utilize sewer routers to form multicasts that can simultaneously transport multicast packets and TCP packets. For multicast network scalability and error recovery the existing SRM method has been used. Three packets per TCP transmission control window size are used for transport and an ACK is used for flow control. A CBR 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.

• Journal of the Institute of Convergence Signal Processing
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• v.3 no.3
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• pp.14-20
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• 2002

In this paper, we proposed the compression technique of the video data using SPIHT(set partitioning in hierarchical trees) based on wavelet and the multi-stream delivery method for best-effort service as fully utilizing the clients bandwidth over the current Internet. The experiment shows that the proposed method provides about 1.5dB better picture quality without block effects than DCT(discrete consine transform) based coding schemes at the same bit rates because of using the wavelet video coder. In addition, this technique implements the multi-stream transmission based on TCP(transmission control protocol). Thus, it is provided with the best-efforts service which is robust to the network jitter problem, and maximally utilizes the bandwidth of the client's.