• Proceedings of the Korean Society of Marine Engineers Conference
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• 2000.11a
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• pp.152-156
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• 2000

Many industrial equipments use RS-232C interface. But Ethernet using CSMA/CD was widly spread recently. This interface is not compatible with RS-232C and requires PC based system. So In this Paper, We decribe a Internet control and monitoring system that control and watch plants without PC. We developed IP controller consisted of microprocessor and NIC and monitoring system using VC++ and SQL database. The protocol used this system is UDP/IP because it is applicable for real time system.

• 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 Society of Broadcast Engineers Conference
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• 1997.11a
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• pp.169-172
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• 1997

본 논문은 ITU-T에서 제시한 H.323, H.225 표준설계를 토대로 MPEG-4 DMIF의 트랜스먹스 레이어 설계, 구현에 관한 것이다. 윈도우 95 및 윈도우 NT환경에서 Winsock을 이용하여 UDP로 MPEG-4 비트스트림을 전달하고 전송제어를 TCP로 구현하였다. 신뢰성이 없는 UDP의 문제를 보완하기 위해 신뢰성 있는 TCP를 한 프로세스에 동시에 할당하여 요구자의 MPEG-4 비트스트림 선택, UDP의 패킷발생률, 패킷사이즈 등 QoS를 제어함으로써 UDP를 보완하도록 구현하였다. MPEG 표준위원회가 제공하는 소프트웨어를 이용하였고, 본 연구에서는 서버에 있는 MPEG-4 비트스트림을 클라이언트가 다운로딩 할 수 있도록 MPEG-4 DMIF 다양한 트랜스먹스 레이어 네트워크 프로토콜 중 MPEG-4 FlexMux/UDP/IP 구현하는 것인데 이는 앞으로 해야 할 실시간 전송 프로토콜의 RTP에 대해 기본틀을 마련하였다고 할 수 있다.

• The KIPS Transactions:PartC
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• v.18C no.6
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• pp.423-432
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• 2011

Recently, OpenFlow has been paid attention to as a fundamental technology which provides a function of virtualization and programmability in network. In Korea, deployment of OpenFlow networks in campuses and the interconnection between them through tunneling in layer 3 has been performed. However, the performance of the interconnected networks is decreased due to delay in IP layer. In this paper, we design and deploy nation-wide, not local, OpenFlow networks in a pure layer 2 environment over KREONET. After that, we do end-to-end Round-trip Time measurements and TCP/UDP performance tests in OpenFlow and normal networks, and do comparison and analysis on the test results. The results show that the nation-wide OpenFlow networks provide equal performance to normal networks except for the initial packet loss for UDP streaming. In regards to the performance decrease due to early UDP packet loss, we can mitigate it by implementing exceptional procedures in a controller which deal with the same continuous "Packet_in" events.

• Journal of KIISE:Information Networking
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• v.29 no.3
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• pp.242-252
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• 2002

When a Mobile IP handoff occurs, the packets in flight can be lost because they are tunneled based on out-of-date location information. In this parer, we propose an enhanced handoff protocol that achieves no packet loss during Mobile IP handoff over a wireless LAN. Our handoff protocol predicts the next foreign agent that a mobile host is to visit by using: the information from the data link layer of wireless LAN. After that, when a Mobile IP handoff occult, the current foreign agent forwards the packets destined to a mobile host to the predicted foreign agent which buffers them. This eliminates packet loss and reduces the packet forwarding delay. Our handoff protocol is simulated using the Network Simulator-2 (ns-2) and shows the substantial performance enhancement of TCP with much less overhead up to 6.2 times compared to standard Mobile IP.

• The Journal of Korean Association of Computer Education
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• v.7 no.1
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• pp.37-44
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• 2004

Differentiated services network (DiffServ) aims to provide the same service to a group of connections that have similar Quality of Service requirements. One of the essential function to realize DiffServ is the traffic conditioning mechanism. The paper proposes the enhanced traffic conditioning mechanism which can assure the reserved rates of TCP and UDP flows and support fair distribution of excess bandwidth. The simulation results show that the new mechanism is rather insensitive of the effect of UDP against TCP throughput, and performs better both in terms of throughput assurance and fair distribution of excess bandwidth in case of well-provisioned and over-provisioned network environment.

• Proceedings of the IEEK Conference
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• 2002.06a
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• pp.311-314
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• 2002

This paper deals with an improved method of utilizing end-to-end bandwidth in the All-IP environment. The proposed method includes compression of UDP/RTP headers, and multiplexing of the RTP stream packets over the end-to-and media transfer. Although the conventional method of using TCRTP(Tunneling Multiplexed Compressed RTP) is an efficient mettled of maximizing tile network throughput, it is inadequate for the All-IP based end-to-end communication. The method is a link-layer independent solution that can be easily implemented in the NGN(Next Generation Network).

• 제어로봇시스템학회:학술대회논문집
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• 2002.10a
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• pp.121.2-121
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• 2002

$\textbullet$ Introduction $\textbullet$ Motivation $\textbullet$ EUDP/IP protocol $\textbullet$ Experiments $\textbullet$ Conclusion $\textbullet$ Reference

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2015.10a
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• pp.791-793
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• 2015

Recently, UCC (User Created Contents) and VoD (Video on Demand), and multimedia content are growing, IP-TV, Smart TV, OHTV (Open Hybrid TV) various services such as multi platform (Multi-platform) environment, services and QoS issues. To solve this problem, the network efficiently, and improve the quality of content is necessary for the system. In this paper, the network of channels State and transmission of multimedia data based on dynamic resource usage, TCP and UDP, Adaptive dual-streaming system used for design and analysis. In addition, the existing TCP and UDP streaming system using a single protocol for analysis and verification of the effectiveness of the difference between and. This is a disaster, and medical/first aid system will be utilized in the field of feed, are ubiquitous.

• Proceedings of the Korea Multimedia Society Conference
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• 2002.05d
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• pp.752-757
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• 2002

IP 멀티캐스트는 다자간 통신을 위한 효과적인 방법이지만 신뢰성 보장과 FTP나 Telnet등 TCP제공하는 다양한 전송프로토콜을 지원하지 않는다. 본 논문에서는 멀티캐스트의 UDP와 TCP계층의 전송 프로토콜을 동시에 사용할 수 있는 신뢰성 있는 멀티캐스트 서버라우터를 제안한다. 신뢰성 있는 멀티캐스트가 반드시 고려해야 할 확장성과 오류회복, 흐름제어를 위해, 그룹의 확장성과 오류회복에 적합한 기존의 SRM방식을 이용하였다. SRM방식은 신뢰성 있는 다-대-다 멀티캐스트 패킷을 전달한다. 각 참여자는 동시에 수신자와 송신자 역할을 할 수 있다. 모든 그룹 멤버는 낮은 주파수의 세션 메시지를 교환하고 그룹내의 주변 가입자에 대한 정보를 얻게 되며 패킷 손실 판단에 필요한 지연시간을 측정하고 복구하는 방식이다. UDP의 트래픽 기반은 멀티캐스트에 적합한 CBR과 SRM을 사용하였다. 본 연구에서는 멀티캐스트의 UDP 패킷과 TCP 패킷을 동시에 보낸 수신율과 멀티캐스트의 UDP만 보낸 수신율을 지연이 가장 큰 멀티캐스트 수신자측에서 패킷 번호와 수신율을 측정하고, 시뮬레이션을 통해서 검증을 통하여 기존의 멀티캐스트 전송방법과 비교하였다.