• 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.

• Proceedings of the Korean Information Science Society Conference
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• 2001.04a
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• pp.433-435
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• 2001

인터넷상에서 화상전화를 구현할 경우, 대용량의 멀티미디어 데이터의 전송은 네트웍 congestion의 원인이 될 수 있다. 이러한 congestion이 일어날 경우, congestion avoidance 알고리즘을 적용하는 TCP 데이터는 스스로 전송률을 줄이게 되므로, congestion 정책을 사용하지 않는 UDP 패킷과 같은 데이터와 함께 전송될 경우, TCP 데이터에 불리하게 된다. 이때, UDP 패킷 데이터에 TCP와 유사한 방법의 congestion avoidance 알고리즘을 적용하여 이를 해결할 수 있는데, 이것은 TCP-friendly Adaptation 알고리즘이다. 본 논문에서는 인터넷 화상전화의 구현에 대해 기술하고 인터넷 환경에서 화성전화를 사용할 때에 congestion을 control 하기 위해서, 그 출력 대역폭을 네트웍 상태에 따라 TCP와 유사한 방식으로 조절하는 TCP-friendly Adaptation 알고리즘을 적용한다.

• Proceedings of the Korea Information Processing Society Conference
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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 the Korean Institute of Intelligent Systems
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• v.14 no.3
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• pp.285-289
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• 2004

Recently, due to the high development of the internet, needs for multimedia streams such as digital audio and video is increasing much more. In case of transmitting multimedia streams using the User Datagram Protocol (UDP), it may cause starvation of TCP traffic on the same transmission path, thus resulting in congestion collapse and enormous delay because UDP does not perform TCP-like congestion control. Because of this problem, diverse researches are being conducted on new transmission schemes and protocols intended to efficiently reduce the transmission delay of real-time multimedia streams and perform congestion control. The TCP-friendly congestion control schemes can be classified into the window-based congestion control, which uses the general congestion window management function, and the rate-based congestion control, which dynamically adjusts transmission rate by using TCP modeling equations and the like. In this paper, we suggest the square-root congestion avoidance algorithm with the hybrid TCP-friendly congestion control scheme which the window-based and rate-based congestion controls are dealt with in a combined way. We apply the proposed algorithm to the existing TEAR. We simulate the performance of the proposed TEAR by using NS, and the result shows that it gives better improvement in the stability needed for providing congestion control than the existing TEAR.

• 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 Korean Institute of Information and Commucation Sciences Conference
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• 2014.05a
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• pp.571-573
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• 2014

In today's technology streaming service's QoS technologiey is an issue. On quality video streaming service there are some technical issues exist, such as buffering. This submission is "Adaptive dual-streaming system design" which is for the integrity of the data streaming that is sent to TCP and UDP for faster transmission of data to the stream. This system provides real-time incoming video encoding in bitrate of h.264-based H through a process based on the video footage of several server and client-to-TCP and UDP via Adaptive providing streaming services in a network environment. This is an unspecified number of buffers in a network environment and continued through the minimization of various streaming for playback of videos and multimedia will be utilized in the field.

• 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 Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.2B
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• pp.1-10
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• 2005

TCP/IP protocols have been implemented in software program running on CPU in end systems. As the increased demand of fast protocol processing, it is required to implement the protocols in hardware, and Host Interface is responsible for communication between external CPU and the hardware blocks of TCP/IP implementation. The Host Interface follows AMBA AHB specification for the communication with external world. For control flow, the Host Interface behaves as a slave of AMBA AHB. Using internal Command/status Registers, the Host Interface receives commands from CPU and transfers hardware status and header information to CPU. On the other hand, the Host Interface behaves as a master for data flow. Data flow has two directions, Receive Flow and Transmit Flow. In Receive Flow, using internal RxFIFO, the Host Interface reads data from UDP FIFO or TCP buffer and transfers data to external RAM for CPU to read. For Transmit Flow, the Host Interface reads data from external RAM and transfers data to UDP buffer or TCP buffer through internal TxFIFO. TCP/IP hardware blocks generate packets using the data and transmit. Buffer Descriptor is one of the Command/Status Registers, and the information stored in Buffer Descriptor is used for external RAM access. Several testcases are designed to verify TCP/IP functions. The Host Interface is synthesized using the 0.18 micron technology, and it results in 173 K gates including the Command/status Registers and internal FIFOs.

• Proceedings of the Korean Information Science Society Conference
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• 2001.10c
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• pp.424-426
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• 2001

현재의 인터넷 응용들을 살펴보면 종 단간 혼잡 제어 방법을 지원하는 TCP를 대표적인 전송 프로토콜로 사용하고 있음을 알 수 있다. 그러나 최근 폭발적으로 증가하는 멀티미디어 서비스들은 UDP 혹은 보다 적절한 RTP(Real-time Transport Protoco) 와 같은 미디어의 실시간 특성에 맞는 전송 프로토콜을 주로 사용하고 있다. 그런대 TCP-friendly 하지 않는 UDP나 RTP 같은 트래픽의 무분별한 증가는 같은 링크를 점유하며 공정하게 경쟁하는 TCP 연결들의 전송 효율을 억제 하는 특성을 나타낸다. 그러므로 이러한 현상은 네트워크 활용에 불균형 현상을 초래 한다. 본 논문에서는 이러한 문제를 해결 하기위해서 TCP-Friendly 멀티미디어 전송 프로토콜인 SRTP(Smart RTP)를 제안하였다. 또한 구현 및 성능평가를 통해서 이 프로토콜이 스트리밍 전송률을 동적으로 순화하면서 혼잡 상황에 적절히 적응 할 수 있음을 나타내었다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.6
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• pp.1496-1502
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• 1998

In this paper, a real-time audio broadcasting service system which is robust to loaded traffic on the Internet is developed. For implementing reliable real-time data transfer, the transfer characteristics of TCP/IP and UDP/IP was compared and analyzed. For lost packet recovery, redundant audio data algorithm was used and interleaving technique was applied for scattering consecutive packet loss. Test results showed, when using TCP/IP, pause occurred during playback, and when using UDP/IP, a stable receive rate was noticeable but the quality of the sound was lower than that of uisng TCP/IP. The recovery rate using redundant audio data and interleaving technique is shown in Fig. 9 and the delay is shown in Fig 4.