• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.4B
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• pp.227-233
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• 2007

According to the recent development of network technology, the demands of users are diversified and the needs of multimedia traffic are increasing. In general, UDP(User Datagram Protocol) traffic is used to transport multimedia data, which satisfied the real-time and isochronous characteristics. UDP traffic competes with TCP traffic and incur the network congestion. However, TCP traffic performs network congestion control but does not consider the receiver's status. Thus, it is not appropriate in case of streaming services. In this paper, we solve a fairness problems and proposed a network algorithm based on RTP/RTCP(Real-time Transport Protocol/Realtime Transport Control Protocol) in view of receiver status. The POBA(Priority Ordered Buffer Algorithm), which applies priorities in the receiver's buffer and networks, shows that it provides the appropriate environment for streaming services in view of packet loss ratio and buffer utilization of receiver's buffer compared with the previous method.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.2
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• pp.304-309
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• 2004

Recently, the needs for multimedia stream such as digital audio and video in the Internet has increased tremendously. Unlike TCP, the UDP protocol, which has been used to transmit streaming traffic thorough the Internet, does not apply any congestion control mechanism to regulate the data flow thorough the shared network And it leads to congestion collapse of the Internet and results in long-term transmission delay. To avoid any adverse effect on the current Internet functionality, a now protocol of modification or addition of some functionality to perform congestion control and to reduce huge transmission delay in transmitting of multimedia stream are in study. TCP-friendly congestion control mechanism is classified into two : one is window-based congestion control scheme using general window management functionalities, the other is rate-based congestion control scheme using TCP modeling equation. In this paper, we propose an algorithm for improving the transmitting rate on a hybrid TCP-friendly congestion control scheme combined with widow-based and late-based congestion control for multimedia stream. And we also simulate the performance of improved TEAR implementation using NS. With He simulation results, we show that the improved TEAR can provide better fairness and lower rate fluctuations than TCP.

• 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.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.11B
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• pp.736-743
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• 2005

NAT-PT is one of the IPv6 transition mechanisms, as defined in RFC2766, allowing IPv6-only devices to communicate with IPv4-only devices and vice versa. In NAT-PT, sender fail to verify TCP/UDP checksum and authentication data due to IP translation in the NAT-PT server The NAT-PT, therefore, has a limit to applying the IPsec that provides the end-to-end security such as confidentiality, authentication, and integrity. This paper proposes a scheme to apply the IPsec using IP HTI in NAT-PT environment.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.837-840
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• 2003

Recently the multimedia streaming traffic such as digital audio and video in the Internet has increased tremendously. Unlike TCP, the UDP protocol, which has been used to transmit streaming traffic through the Internet, does not apply any congestion control mechanism to regulate the data flow through the shared network. If this trend is let go unchecked, these traffic will effect the performance of TCP, which is used to transport data traffic, and may lead to congestion collapse of the Internet. To avoid any adverse effort on the current Internet functionality, A study on a new protocol of modification or addition of some functionality to existing transport protocol for transmitting streaming traffic in the Internet is needed. TCP-frienly congestion control mechanism is classified with window-based congestion control scheme and rate-based congestion control scheme. In this paper, we propose an algorithm for improving the transmitting rate on a hybrid TCP-friendly congestion control scheme combined with widow-based and rate-based congestion control for multimedia streaming in the internet.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.6
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• pp.1078-1082
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• 2001

This is an example of ABSTRACT format. We made a HTTP server using 8 bit microprocessor It was TMP84c015 which applied a z80 core and RTL8019AS was installed for an ethernet physical layer. Assembly language was used to optimized a performance of the MPU, to overcome an restriction of memory sire and to maximize the throughput of packet using TCP, UDP, IP, ICMP, ARP protocol. We used LabVIEW to verify the each protocol on the client side.

• Journal of KIISE:Information Networking
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• v.32 no.6
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• pp.732-744
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• 2005

Previous works for the AE(Assured Forwarding) service in the Diffserv network have no sufficient consideration on the fairness of bandwidth share based on the target rate and the effect or RTT and UDP. Also Previous works act like Best-effort service in the UPN(under-Provisioned Network) condition. In this paper, in order to solve these problems, we propose the PFDSA(Proportionally Fair Differentiated Service Architecture) composed of tmTRA3CM(tcp-microflow based Target rate and an Aware Three color Marking), um3CM(udp-microflow based Three color Marker), TRBD(Target Rate Based Dropper), and target rate adjusting function. In the results of comparing the performance among existing mechanisms and the PFDSA, the PFDSA was able to mitigate the RTT and UDP effect better than the former. The PFDSA was shown to provide good performance for transmission rates proportional to various target rates in the UPN condition.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.13 no.3
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• pp.111-118
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• 2013

The connection based TCP protocol provides reliable packet delivery, but it takes certain overhead compare to the connection-less UDP protocol. Thus, 3GPP devise the upper-layer RLC protocol which provides reliability based upon the UDP protocol. Consequently, each implement of a base station and/or mobile terminal require the development of the RLC protocol, and it must qualify various interoperable tests. In this paper, we implement a testbed which verifies the RLC protocol under various packet losses/inversion circumstances of networks. Finally, we propose our testbed as the RLC protocol tester for the developments.