• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.532-534
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• 2002

본 논문에서는 UDP／IP기반 SAN의 구현을 소개하였고 TCP／IP기반 SAN 및 로컬디스크와의 성능 비교를 통해 UDP／IP기반 SAN이 클러스터 웹서버에서 사용될 수 있는 가능성을 보였다. 실험결과 UDP／IP기반 SAN은 TCP／IP기반 SAN의 경우보다 약 20%정도 성능이 우수함을 볼 수 있다. 이는 UDP의 경우 TCP의 프로토콜 오버헤드가 없고 slow start의 영향을 받지 않으며 ACK를 기다릴 필요가 없기 때문이다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.6
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• pp.1076-1085
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• 2004

Recently, the demands for real-time service and multimedia data are rapidly increasing. There are significant redundancies between header fields both within the same packet header and in consecutive packets belonging to the same packet stream. And there are many overheads in using the current UDP/IP protocol. Header compression is considered to enhance the transmission efficiency for the payload of small size. By sending the static field information only once initially and by utilizing dependencies and predictability for other fields, the header size can be significantly reduced for most packets. This work describes an implementation for header compression of the headers of IP/UDP protocols to reduce the overhead on Ethernet network. Typical UDP/IP Header packets can be compressed down to 7 bytes and the header compression system is designed and implemented in Linux environment. Using the Header compression system designed between a server and clients provides have the advantage of effective data throughput in network. Since the minimum packet size in Ethernet is 64 bytes, the amount of reduction by header compression in practical chatting environment was 6.6 bytes.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.8
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• pp.1696-1702
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• 2005

Real Time Transport Protocol (RTP) is the Internet standard protocol for transport of real time data audio/video IP Telephony, Multimedia Seivece. In case of 8kbps voice codec, the size of packet per data is 20bytes and become more large to minimal 40bytes with adding each layer's header in RTP/UDP/IP. To solve this problem, various header compression skill were suggested on point-to-point networks. But it compress even IP header and cannot be suitable to apply to end-to-end network Thus, We will renew header compression protocol to apply wired router-based network.

• The KIPS Transactions:PartA
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• v.14A no.5
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• pp.295-300
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• 2007

The Linux-based embedded systems such as mobile telephones. PDAs and MP3 players are widely in use. USB(Universal Serial Bus) is the interface for data communication between the computers and these peripheral devices. Some embedded systems like intelligent home networking and multimedia streaming require guaranteed QoS(Quality of Service), which is needed for real time transmission of UDP/IP messages through USB. Although USB Ethernet driver is supported by USB Gadget API in Linux, it is unable to provide the desirable QoS required by each type or small embedded systems due to the unpredictability or TCP/IP Stack in Linux. This paper proposes IP-Over-USB to improve QoS of UDP/IP message transmission in the embedded systems using USB in Linux system.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.23 no.3
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• pp.59-64
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• 2023

Fire and disaster broadcasting systems are divided into analog, digital, and network-based digital public address systems, and important specifications in network-based digital public address systems are low-latency audio, high sampling rate, and multi-channel input and output. In the past, it has been widely used to the AoE method for distinguishing based on the MAC address of the data link layer. However, this method has a problem of increasing complexity and cost. This proposal is an AoIP/UDP method, which allows communication to be easily distinguished by IP address without the need for a separate redundant network, so that the network can be freely used and configured, and cost can be reduced by reducing complexity. After implementing the AoIP/UDP method, the experimental results showed that the cost was improved with the equivalent performance with 2.66ms latency.

• Journal of Advanced Marine Engineering and Technology
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• v.27 no.1
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• pp.65-75
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• 2003

The internet telephony service is one of the successful internet application services. VoIP is the key technology for the service to come true. VoIP uses H.323 or SIP as the standard protocol for the distributed multimedia services over the internet environment, in which QoS is not guaranteed. VoIP carries the packetized voice by using the RTP/UDP/IP protocol stack. The UDP-based internet services cause the data transmission problem to the users behind the internet firewall. So does the internet telephony service. The users are not able to listen the voices of the counter-parts on the public internet or PSTN. It makes the problem more difficult that the internet telephony service addressed in this paper uses only one UDP port number to send the voice data of all sessions from gateway to terminal node. In this paper, two schemes including the usage of dummy UDP datagrams, and the protocol conversion are suggested. The implementation of one of the schemes, the protocol conversion, and the performance evaluation are described in detail.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.05a
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• pp.704-711
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• 2003

Recently, the demands for real-time service and multimedia data are rapidly increasing. There are significant redundancies between header fields both within the same packet header and in consecutive packets belonging to the same packet stream. But there are many overheads in using the current UDP/IP protocol. Header compression is considered to enhance the transmission efficiency for small size of payload. By sending the static field information only once initially and by utilizing dependencies and predictability for other fields, the header size can be significantly reduced for most packets. This work describes an implementation for header compression of the headers of U/UDP protocols to reduce overhead on Ethernet network. Typical UDP/IP Header packets can be compressed down to 7 bytes and the header compression system is designed and implemented on the Linux environment. Using the designed Header compression system between a server and a client have the advantage of effective data throughput in network.

• Journal of Advanced Navigation Technology
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• v.24 no.5
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• pp.438-443
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• 2020

To transfer files between nodes on network based on Ethernet, file transfer protocol (FTP) on TCP/IP and trivial file transfer protocol (TFTP) on UDP/IP are mostly used. Due to the lack of resources (processor, memory and so on) in the embedded system where we generally use for simple works with small firmware like ones; many of the systems implement only UDP/IP for their network stacks. Thus, TFTP is greatly to be preferred. For examples, environmental sensor devices for sensor networks, Boot Loader for general embedded device and preboot execution environment (PXE) boot for PC provide the TFTP. The logic of TFTP is simple for file transmission but, there is Stop-And-Wait problem during the process which occurs long blocking time. In this paper, we propose an algorithm which called QuTFTP(Quick UDP Trivial File Transfer Protocol) to reduce the length of the blocking time and to be compatible with the legacy TFTP.

• Journal of KIISE:Computing Practices and Letters
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• v.9 no.1
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• pp.47-59
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• 2003

The Internet telephony service is one of the commercially successful Internet application services. VoIP technology makes the service come true. VoIP deploys H.323 or SIP as the standard protocol for the distributed multimedia services over the Internet in which QoS is not guaranteed. VoIP carries the packetized voice over the RTP/UDP/IP protocol stack. The data transmission trouble is caused by UDP when the service is provided in private networks and some ISP-provided Internet access networks in the private address space. The Internet telephony users in such networks cannot listen the voices of the other parties in the public Internet or PSTN. Making the problem more difficult, the Internet telephony service considered in this paper gets the incoming voice packets of every session through only one UDP port number. In this paper, three schemes including the terminal proxy, the gateway proxy, and the protocol translation are suggested to solve the problems. The design and implementation of the NAT proxy server based on gateway proxy scheme are described in detail.

• 제어로봇시스템학회:학술대회논문집
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• 2000.10a
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• pp.438-438
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• 2000

본 논문에서는 네트웍의 traffic이 원활하지 못해 프레임에 대한 데이터 전부를 전송 받지 못하더라도 데이터가 들어오는 순서에 따라 영상을 복원한 수 있는 스트림에 대하여 연구를 하였다. 우선 CCD 카메라로부터 초당 24프레임의 영상을 획득한다. 그리고 획득한 영상을 2차원 웨이블릿 변환하여 얻은 중요한 계수를 다시 웨이블릿 변환을 한다. 위와 같은 방식으로 웨이블릿 변환을 네 번 반복하여 중요한 계수를 가지고 있는 대역에서 중요하지 않은 대역순으로 부호화하여 스트림을 구성한다. 그리고 영상을 서비스 받기를 친하는 수신자에게 UDP/IP를 이용하여 전송한다. 네트웍이 원활하지 않아 스트림 전부를 받지 못하더라도 먼저 전송받은 중요한 계수들로 복원하기 때문에 영상을 끊기지 않고 점진적으로 복원할 수 있었다. 이와 같은 땅법을 인터넷 생방송이나 VOD 서비스, 화상 채팅 등에 이용하면 훨씬 나은 영상을 서비스를 할 수 있을 것이다.