• Annual Conference of KIPS
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• 2002.04b
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• pp.1483-1486
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• 2002

Multicast protocols are efficient methods of group communication, but they do not support the various transmission protocol services like a reliability guarantee, FTP, or Telnet that TCPs do. The purpose of this dissertation is to find a method to utilize sewer routers to form multicasts that can simultaneously transport multicast packets and TCP packets. For multicast network scalability and error recovery the existing SRM method has been used. Three packets per TCP transmission control window size are used for transport and an ACK is used for flow control. A CBR 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 Communications and Networks
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• v.13 no.3
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• pp.277-285
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• 2011

Real-time multimedia streaming over the Internet is rapidly increasing with the popularity of user-created contents, Web 2.0 trends, and P2P (peer-to-peer) delivery support. While many homes today are broadband-enabled, the quality of experience (QoE) of a user is still limited due to frequent interruption of media playout. The vulnerability of TCP (transmission control protocol), the popular transport-layer protocol for streaming in practice, to the packet losses, retransmissions, and timeouts makes it hard to deliver a timely and persistent flow of packets for online multimedia contents. This paper presents TCP-real-time online multimedia environment (ROME), a novel transport-layer framework that allows the establishment and coordination of multiple many-to-one TCP connections. Between one client with multiple home addresses and multiple co-located or distributed servers, TCP-ROME increases the total throughput by aggregating the resources of multiple TCP connections. It also overcomes the bandwidth fluctuations of network bottlenecks by dynamically coordinating the streams of contents from multiple servers and by adapting the streaming rate of all connections to match the bandwidth requirement of the target video.

• The Transactions of the Korea Information Processing Society
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• v.6 no.11S
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• pp.3260-3268
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• 1999

As the proliferation of multimedia traffic over High-speed Internet increases, ATM network will be vital to adopt as backbone network over various parts of Internet. In this paper, we investigate the performance of TCP/IP traffic flow over ABR and UBR of ATM service to study for the high throughput and good fairness by simulation technique. Although TCP is run in the transport layer, it is controlled by several methods, e.g, EPD, PPD, RED, EFCI, ER etc, in ATM layer when TCP uses the ABR/UBR service. Therefore, if one cell is discarded in ATM layer, a packet of TCp will be laost. And, also, along with the increasing of the number of VC among switches, the throughput and fairness will be degraded. In order to improve these degradations, we propose the effective parameter control operations of EFCI and ER on ABR service, and also suggest the buffer management methods on UBR service. Finally, through the simulation results, the improved throughput and fairness are shown.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.23 no.8
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• pp.93-98
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• 2009

Cogeneration means the highly energy-efficient generation system that improves energy ratio by generating electricity and heat, and it really affects the improvement of overall efficiency by using industrial process, district heating, and hot-water supply etc. after the energy produced through supplying power to system collects. This thesis indicates the screen of cogeneration flow in LabVIEW and the heat power control system that can be in long-distance control of a district energy system using TCP/IP. We simulated on four computers so that the heat power control system proves long-distance control possible.

• The Journal of the Korea institute of electronic communication sciences
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• v.16 no.6
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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.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.1
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• pp.414-435
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• 2017

As network adaptive streaming technology becomes increasingly common, transport protocol also becomes important in guaranteeing the quality of multimedia streaming. At the same time, because of the appearance of high-quality video such as Ultra High Definition (UHD), preventing buffering as well as preserving high quality while deploying a streaming service becomes important. The Internet Engineering Task Force recently published Multipath TCP (MPTCP). MPTCP improves the maximum transmission rate by simultaneously transmitting data over different paths with multiple TCP subflows. However, MPTCP cannot preserve high quality, because the MPTCP subflows slowly increase the transmission rate, and upon detecting a packet loss, drastically halve the transmission rate. In this paper, we propose a new multipath congestion control scheme for high-quality multimedia streaming. The proposed scheme preserves high quality of video by adaptively adjusting the increasing parameter of subflows according to the network status. The proposed scheme also increases network efficiency by providing load balancing and stability, and by supporting fairness with single-flow congestion control schemes.

• Proceedings of the KIEE Conference
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• 2006.07d
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• pp.1828-1829
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• 2006

In this paper, we propose sliding mode congestion controller for differentiated-services network. Two important issue in differentiated-services architecture are bandwidth guarantee and fair sharing of unsubscribed bandwidth among TCP flows with and without bandwidth reservation. We use tight upper and lower bounds for various settings of differentiated-services parameters using the loss-bounded model. The Sliding mode congestion controller scheme is designed using nonlinear control theory based on a nonlinear model of the network that is generated using fluid flow consideration. The methodology used is general and independent of technology, as for example TCP/IP or ATM. The sliding mode congestion controller methodology has been applied to an TCP network. We use NS-2 simulation to demonstrate that the proposed control methodology achieves the desired behavior of the network, and possesses important attributes. as e.g, stable and robust behavior, high utilization with bounded delay and loss, together with good steady-state and transient behavior.

• Proceedings of the Korean Information Science Society Conference
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• 2005.07a
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• pp.568-570
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• 2005

광대역 망에서 범용 TCP를 사용하는 경우 최대 전송 윈도우의 크기가 제한되어 가용한 망 자원이 낭비된다. 이를 해결하는 일반적인 방법은 윈도우 스케일 옵션을 사용하는 것인데 이 경우 송신측의 응용를 수정해야 하는 단점이 있다. 본 논문에서는 이를 해결하기 위하여 송신측을 수정하지 않고 전송 윈도우의 크기를 증가시키는 방법을 제안한다. 이 방법에서는 송수신측 사이에 있는 Snoop 게이트웨이가 망의 상태와 수신측의 처리 능력을 동적으로 파악하여 최적의 윈도우 확대 계수를 결정하고, 이 계수에 상응하는 수의 응답 패킷을 송신측에 전송하여 전송 윈도우 크기가 확대된 효과를 얻는다. 특히, 이 방법은 서로 다른 특성을 갖는 망이 snoop 게이트웨이를 사용하여 2-way 세그먼트 방식으로 연동된 환경에서 부가적인 비용없이 단대단 시맨틱을 유지하면서 성능을 향상시킬 수 있다.

• The Journal of the Korea institute of electronic communication sciences
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• v.15 no.6
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• pp.1045-1054
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• 2020

In this paper, we designed a TCP/IP based ring buffer system that can stably transfer bulk data streams in the unstable network environments. In the scheme we proposed, The observation data stream generated and output by each radio observatory's backend system as a UDP frame is stored as a UDP packet in a large capacity ring buffer via a socket buffer in the client system. Thereafter, for stable transmission to the remote destination, the packets are processed in TCP and transmitted to the socket buffer of server system in the correlation center, which packets are stored in a large capacity ring buffer if there is no problem with the packets. In case of errors such as loss, duplication, and out of order delivery, the packets are retransmitted through TCP flow control, and we guaranteed that the reliability of data arriving at the correlation center. When congestion avoidance occurs due to network performance instability, we also suggest that performance degradation can be minimized by applying parallel streams.

• Journal of KIISE:Information Networking
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• v.30 no.3
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• pp.437-447
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• 2003

In order to reduce the increasing packet loss rates caused by an exponential increase in network traffic, the IETF(Internet Engineering Task Force) is considering the deployment of active queue management techniques such as RED(Random Early Detection) algorithm. However, RED algorithm simple but does not protect traffic from high-bandwidth flows, which include not only flows that fail to use end-to-end congestion control such as UDP flow, but also short round-trip time TCP flows. In this paper, in order to solve this problem, we propose a simple fairness queue management scheme, called AFQM(Approximate Fair Queue Management) algorithm, that discriminate against the flows which submit more packets/sec than is allowed by their fair share. By doing this, the scheme aims to approximate the fair queueing policy Since it is a small overhead and easy to implement, AFQM algorithm controls unresponsive or misbehaving flows with a minimum overhead.