• Proceedings of the Korean Information Science Society Conference
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• 한국정보과학회 2004년도 봄 학술발표논문집 Vol.31 No.1 (A)
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• pp.481-483
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• 2004

본 논문에서는 차등 서비스 네트워크(Diffserv network)의 확신 서비스(assured service)에서 발생할 수 있는 TCP와 UDP 플로우 사이의 공명성 문제를 해결하기 위한 TRBD(Target Rate-based Dropping) 기법을 제안한다. 차등 서비스 네트워크의 확신 서비스에서의 공평성 향상에 대한 기존 연구는 마킹 확률을 변경하여 공평성을 향상시키는 연구가 중심이 되어 왔다. 이러한 연구들은 코어노드에서의 큐 관리를 위하여 주로 MRED(Multiple RED)골 이용하기 때문에 TCP 및 UDP, 목표 전송률 등에 대한 고려가 부족하다. 본 논문에서 제안하는 TRBD 기법은 코어노드에서 목표 전송률에 따라 UDP의 폐기 확률을 조정하여 TCP-UDP 공평성 및 목표 전송률을 고려한 공평성을 향상시킬 수 있다. ns 시뮬레이터를 이용한 실험을 통하여 TRBD 기법의 TCP-UDP 공평성 향상을 확인하였다.

• The Journal of Korean Association of Computer Education
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• 제7권1호
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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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• 대한전자공학회 1999년도 추계종합학술대회 논문집
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• pp.1061-1064
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• 1999

UDP has likely been used for real-time applications, such as video and audio. UDP supplies minimized transmission delay by omitting the connection setup process, flow control, and retransmission In general, more than 80 percent of the WAN resources are occupied by Transmission Control Protocol(TCP) traffic as opposed to UDP's simplicity, TCP adopts a unique flow control in this paper, I report new methods to minimize a udp packet loss considering TCP flow control on the real-time application the better performance of real time application can be obtained when they reduce a packet size and FIFO buffer scheduling method competing with TCP bandwidth for the bandwidth and buffering.

• Journal of KIISE:Information Networking
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• 제30권5호
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• pp.649-655
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• 2003

Network-based computing such as cluster computing requires a reliable high-speed transport protocol. TCP is a representative reliable transport protocol on the Internet, which implements many mechanisms, such as flow control, congestion control, and retransmission, for reliable packet delivery. This paper, however, finds out that Myrinet does not incur any packet losses caused by network congestion. In addition, we ascertain that Myrinet supports reliable and ordered packet delivery. Consequently, most of reliable routines implemented in TCP produce unnecessarily additional overheads on Myrinet. In this paper, we show that we can attain the reliability only by flow control on Myrinet and propose a new reliable protocol based on UDP named RUM (Reliable UDP on Myrinet) that performs a flow control. As a result, RUM achieves a higher throughput by 45% than TCP and shows a similar one-way latency to UDP.

• Proceedings of the Korean Information Science Society Conference
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• 한국정보과학회 2002년도 가을 학술발표논문집 Vol.29 No.2 (3)
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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.

• The Journal of the Korea institute of electronic communication sciences
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• 제16권6호
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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.

• International Journal of Computer Science & Network Security
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• 제22권2호
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• pp.327-333
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• 2022

Internet Protocol Television (IPTV) has emerged as a personal entertainment source for home users. Streaming IPTV content over a wireless medium with good Quality of Service (QoS) can be a challenging task as IPTV content requires more bandwidth and Wireless Local Area Networks (WLANs) are susceptible to packet loss, delay and jitter. This research presents the capacity of IPTV using User Datagram Protocol (UDP) and TCP Friendly Rate Control (TFRC) over IEEE 802.11ac WLANs in good and bad network conditions. Experimental results show that in good network conditions, UDP and TFRC could accommodate a maximum of 78 and 75 Standard Definition Television (SDTV) users, respectively. In contrast, 15 and 11 High-Definition Television (HDTV) users were supported by UDP and TFRC, respectively. Performance of UDP and TFRC was identical in bad network conditions and same number of SDTV and HDTV users were supported by TFRC and UDP. With background Transmission Control Protocol (TCP) traffic, both UDP and TFRC can support nearly the same number of SDTV users. It was found that TFRC can co-exist fairly with TCP by giving more throughput to TCP unlike UDP.

• Journal of Korea Game Society
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• 제2권1호
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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.

• Journal of Broadcast Engineering
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• 제10권3호
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• pp.297-312
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• 2005

The multimedia traffic of continuous video and audio data via streaming service accounts for a significant and expanding portion of the Internet traffic. This streaming data delivery is mostly based on RTP with UDP. However, UDP does not support congestion control. For this reason, UDP causes the starvation of congestion controlled TCP traffic which reduces its bandwidth share during overload situation. In this paper, we propose a new TCP-friendly rate control scheme called 'TF-RTP(TCP-Friendly RTP)'. In the congested network state, the TF-RTP exactly estimates the competing TCP's throughput by using the modified parameters. Then, it controls the sending rate of the video streams. Therefore, the TF-RTP adjusts its sending rate to TCP-friendly and fair share with competing TCP traffics. Through the simulation, we prove that the TF-RTP correctly estimates the TCP's throughput and improves the TCP-friendliness and fairness.

• Journal of Advanced Navigation Technology
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• 제9권1호
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• pp.61-69
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• 2005

TCP and UDP is a transport layer protocol of Internet. TCP is a connection oriented protocol which supports a reliable data transfer by offering error and flow control, but it bring a transmission delay. On the other hand, the UDP is a connectionless protocol which does not carry out error and flow control, but it guarantees a realtime transmission. There are hardly any protocols which supports not only realtime functions but also data reliability. In this paper, we have designed and implemented a new TCP mode option which supports reliable realtime transmission. Our designed TCP performs an error recovery process during a fixed amount of time. This time is negotiated during the connection establishment phase. Our designed TCP is tested in real environments, and we find that it is relatively faster than the standard TCP and more reliable than the UDP. It can be used for the reliable transfer of realtime multimedia data.