• Journal of Korea Society of Industrial Information Systems
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• v.15 no.5
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• pp.37-51
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• 2010

In this paper, we discuss the program implementation and system optimization for the effective transfer of huge amount of data. In VLBI which is observing the celestial bodies by using radio telescope hundreds thousands km apart, it is necessary for each VLBI observatory to transfer up to terabytes of observed data. For this reason, e-VLBI research based on advanced network is being actively carried out for the transfer of data efficiently. Following this trend, in this paper, we discuss design & implementation of system for the high speed Gbps data transfer rates. As a data transfer protocol, we use UDP for designing data transmission program with much higher speeds than currently available via VTP(VLBI Transport Protocol). Tsunami-UDP algorithms is applied to implementing data transfer program so that transmission performance could be maximize, also we make it possible to transfer observed data more fast and reliable through optimization of computer systems in each VLBI statopm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1496-1502
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• 1998

In this paper, a real-time audio broadcasting service system which is robust to loaded traffic on the Internet is developed. For implementing reliable real-time data transfer, the transfer characteristics of TCP/IP and UDP/IP was compared and analyzed. For lost packet recovery, redundant audio data algorithm was used and interleaving technique was applied for scattering consecutive packet loss. Test results showed, when using TCP/IP, pause occurred during playback, and when using UDP/IP, a stable receive rate was noticeable but the quality of the sound was lower than that of uisng TCP/IP. The recovery rate using redundant audio data and interleaving technique is shown in Fig. 9 and the delay is shown in Fig 4.

• Journal of the Korea Society for Simulation
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• v.19 no.4
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• pp.43-50
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• 2010

Recently, the Internet telephony has been used rather than the traditional telephony by many Internet users, with low cost. Session Initiation Protocol(SIP) is the standard of application layer protocol for establishment and disconnection of the session for Internet telephony. SIP mainly runs over the UDP for transport. So in case of the loss of the INVITE request message, the message is retransmitted by an appropriate timer for reliable transmission of the UDP message. Though the retransmission is useful for ensuring the reliability of SIP messages sent by the users, it may cause the overload traffic in the SIP proxy server. The overload in SIP proxy servers results in the loss of many input messages. This paper presents a non-linear overload control algorithm to resolve the overload condition of the server. we simulate our proposed algorithm using the network simulator ns-2. The simulation results show that the throughput of the server with the proposed algorithm have been improved about 12% compared to the existing linear control algorithm.

• The KIPS Transactions:PartC
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• v.9C no.2
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• pp.235-246
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• 2002

While the Internet is suffering from the massive data such as video stream, IP multicast can ease the load of the Internet by enabling one copy of digital information to be received by multiple computers simultaneously. But If multicast is based on UDP, packets are delivered using a best-effort Policy without any reliability, congestion control or flow control. Multicast group members can join or leave a multicast group at will, and multicast uses broadcast mechanism, it's very hard to keep security from unauthorized members. In this paper, we introduce a new reliable multicast protocol TRDMF proper for one-to-many multicast model with reliability, flow control, congestion control and key management.

• Journal of the Korea Society of Computer and Information
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• v.21 no.1
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• pp.79-84
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• 2016

In this paper, we propose reliable message transmission scheme based on CoAP, considering the constrained feature of IoT device, such as low power, the limited memory size and low computing capacity. Recently, the various kinds of application protocol has been studied to support IoT environments. In particular, CoAP protocol was developed as application protocol for IoT at the IETF core WG. However, because CoAP protocol is deigned to be used in constrained node, this protocol uses UDP at transport layer. Thus, data loss may occur frequently in network congestion environments. The proposed scheme, in this paper, is to overcome the problem of frequent data loss with low overhead. Also it includes the function which is to minimize the data loss in sleep mode of IoT device.

• Journal of Advanced Navigation Technology
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• v.14 no.3
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• pp.343-350
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• 2010

Wireless Sensor Network is an event-driven system that a large number of micro sensor nodes are collected, giving and Wing information. Congestion can take place easily since a great number of sensor nodes are sensing data and transferring them to sink nodes when an event occurs, comparing to the existing wired network. Current internet transport protocol is TCP/UDP, however, this is not proper for wireless sensor network event-driven ESRT, STCP and CODA are being studied for reliable data transport in the wireless sensor network. Congestion are being detected local buffer and channel loading with these techniques. Broadcasting is mainly used and can avoid congestion when one happens. The way that combining local buffer and channel loading information and detecting congestion is being studied in this paper. When congestion occurs, buffering state is divided into three in order to lessen the broadcasting sending the message on congestion control to the node having frequent channel loading. Thus they have advantage of decreasing network's loading.

• The Journal of the Korea Contents Association
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• v.7 no.2
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• pp.100-108
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• 2007

Currently, multimedia stream services are provided in various network environments such as wireless networks. It is crucial to address QoS problem for multimedia service in heterogeneous networks. This paper suggests the scheme to increase the efficiency for the transmission of multimedia data. We also propose the adaptive algorithm that switch between UDP and TCP based on the size of the transmitted multimedia data. In addition, it presents the method that duplicates the multimedia data for the efficiency of transmission. We designed the MPEG stream service and conducted the simulation to verify our adaptive MPEG stream service.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.8
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• pp.1391-1400
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• 2008

In recent yeats, SCTP is known as the next transport layer protocol for connection-oriented and reliable data transfer after TCP/UDP. The SCTP was developed on the basis of the existing TCP and was designed to eliminate defects of TCP. SCTP has the different characteristic of multi-streaming and multi-homing from TCP. In this paper, we studied the defects of the performance of communication using the traits of multi-homing. And we evaluated the efficiency for each application services on the basis of SCTP of IPv6. Also, we analysed the delay time of a packet using SCTP and TCP. We noticed that the efficiency of SCTP is better than that of TCP.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1998.05a
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• pp.241-245
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• 1998

본 논문에서는 이동호스트에게 멀티미디어 데이터 전송 시 고려되어야 할 주요 특징과 문제점을 검토하고 이때 발생하는 문제점을 해결하기 위해 Mobile-IP환경에서 네트워크 링크의 공유에 의해 자원을 효율적으로 사용할 수 있는 멀티캐스트 방법을 기반으로 하고 이러한 멀티캐스트를 위한 인터넷 라우팅 프로토콜로는 링크-상태(link-state)알고리즘을 기반으로 하는 MOSPF(Multicast Open Shortest Path First)를 이용한다. 또한 인터넷상에서 라우터들이 멀티캐스트 그룹에 속하는 호스트들을 추적하기 위한 방법으로 IGMP(Internet Group Management Protocol)을 제안 사용하였다. 기존에 제안되어 사용되어지고 있는 UDP/IP based multicast를 사용할 경우에 패킷손실이나 전송지연 등을 위한 어떠한 성능도 보장받지 못하고 많은 대역폭이 요구되는 실시간 패킷들은 중계기의 트래픽 폭주(congestion) 현상에 의해 손실될 가능성이 높다. 이를 해결하기 위한 방안으로 실시간 데이터 전송 및 QOS를 제공하기 위해 RTP/RTCP 프로토콜을 접목시키는 방식을 기술하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37B no.9
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• pp.814-821
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• 2012

We propose a network processor called an OmniFlow processor capable of wire-speed flow management by a hardware-based flow admission control(FAC) in this paper. Because the OmniFlow processor can set up and release a wire-speed connection for flows, the update period of flows can be set to a short time, and only active flows can be effectively managed by terminating a flow that does not have a packet transmitted within this period. Therefore, the FAC can be used to provide a reliable transmission of UDP as well as TCP applications. This processor is fabricated in 65nm CMOS technology, and total gate count is 25 million. It has 40 Gb/s throughput performance in using the 32 RISC cores when maximum operating frequency is 555MHz.