• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.11B
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• pp.969-978
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• 2006

Recently, there has been explosive growth in Internet service contents, and we have witnessed the emergence of many real-time applications on the Internet. These types of applications require special support from the underlying network such as reliability, timeless, and guaranteed delivery, as well as different levels of service quality. Estimation network status and transcoding data rate are researching for QoS of multimedia service by using the RTP/RTCP. We propose two schemes that the way to calculate the one-way delay by expanding the structure of RTP/RTCP in Internet that provides us 'best effort' service and to estimate the network condition by using this in this paper. And we propose the dynamic RTCP mechanism based on one-way delay to deal with rapidly changed network situation immediately. In simulation, proposed monitoring schemes using the one-way delay has similar in performance to monitoring schemes using the GPS, proposed dynamic RTCP scheme obtain the desired result of decreased packet loss rate by sending RTCP RR packet lower than other schemes.

• Journal of the Korea Society of Computer and Information
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• v.12 no.1 s.45
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• pp.137-144
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• 2007

Advances in networking technology digital media, and codecs have made it possible for the Internet evolves into a Broadband convergence Network (BcN) and provides various services including Voice over Internet Protocol (VoIP) and IPTV over their high-speed IP networks. In order for the Internet to make a profit as traditional Public Switched Telephone Network (PSTN), it must provide high qualify VoIP services. Therefore, real time qualify measurement framework is the most important requisite to provide VoIP service. For this, IETF (Internet Engineering Task Force) defined RTCP-Extended Reports (RTCP-XR) that extend RTCP (Real-Time Transport Protocol Control Protocol). However, procedure and method tot actually VoIP qualify measurement did not recommended nothing but defined item to measure voice quality. Our objective in this paper is to describes a practical measuring framework for end-to-end QoS of switched voice packet in an IP environment. It includes concepts as well as step-by-step procedures for measuring packetized voice streams. It also proposes new formats that extend RTCP-XR's concept.

• Journal of Korea Multimedia Society
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• v.2 no.3
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• pp.339-346
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• 1999

When a multicast video conference system utilizes RTP (Real Time Protocol) and RTCP (Real Time Control Protocol), the loss rate and the synchronization of transfer in RTCP affect the scalability of the system. The random delay technique introduced to resolve the problems is so simple that leads the network to meet some congestion in synchronizing feedback information when lots of people transfer the feedback information simultaneously, which reduces the scalability of system. In this paper, we propose a new feedback control algorithm that provides priority levels with the RTCP packet, which cuts down the feedback delay and increases the scalability. The criteria of providing priority Based on the decided priority level, Agent forced the session participants to provide much more RTCP packets, positively controlled, and the possible bandwidth can be measured. The simulation on this technique decreases the delay, and the feedback messages are equally distributed on a given time period.

• ETRI Journal
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• v.30 no.3
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• pp.441-450
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• 2008

The scalable extension of H.264, known as scalable video coding (SVC) has been the main focus of the Joint Video Team's work and was finalized at the end of 2007. Synchronization between media is an important aspect in the design of a scalable video streaming system. This paper proposes an efficient media synchronization mechanism for SVC video transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, a real-time transport protocol/RTP control protocol (RTP/RTCP) suite is usually employed. To provide an efficient mechanism for media synchronization between SVC video and audio, we suggest an efficient RTP packetization mode for inter-layer synchronization within SVC video and propose a computationally efficient RTCP packet processing method for inter-media synchronization. By adopting the computationally simple RTCP packet processing, we do not need to process every RTCP sender report packet for inter-media synchronization. We demonstrate the effectiveness of the proposed mechanism by comparing its performance with that of the conventional method.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.400-402
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• 2002

본 논문에서는 무선 모바일 환경하에서 동영상 데이터 전송시 네트워크 환경을 고려한 QoS (Quality of Service)를 지원하면서 PSNR(Peak Signal Noise Ratio)의 변화 폭을 최소화시키는 전송제어 알고리즘을 개발하기 위한 환경 구축에 대하여 기술하고 있다. 이를 위하여 ns(network simulator)에 실제 데이터를 전송할 수 있는 환경을 구축하였다. 그리고 이 환경에서 실제로 데이터를 전송시키면서 다양한 RTP／RTCP 제어를 적용시켜 최상의 알고리즘을 찾는 실험을 하였다. 실험에서 RTP／RTCP를 이용해서 데이터 전송측의 패킷전송 간격, 데이터그램 크기 등을 변화시킴으로서, 전송되는 동영상의 품질이 어떻게 변화되는지를 분석하였고, 이를 바탕으로 무선 모바일 환경하에서 QoS를 향상시킬 수 있는 RTP／RTCP 제어방법을 알아보았다.

• Proceedings of the Korean Information Science Society Conference
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• 1998.10a
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• pp.477-479
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• 1998

인터넷상에서의 다자간 회의는 수천명의 참가하는 대규모의 회의가 될 수 있으므로 다자간 회의 시스템에서는 확장성이 중요하다. 현재의 인터넷상에서의 다자간 회의 시스템은 대부분 RTP/RTCP를 이용하는데. RTCP를 이용한 피드백 정보 전송의 빈도 수와 전송 시간의 동기화 현상이 다자간 회의 시스템의 확장성에 큰 영향을 준다. 즉, 세션 참가자 수가 증가함에 따라 네트워크에 전송되는 RTCP 패킷의 숫자가 기하급수적으로 증가하게 된다. 피드백 정보의 전송 빈도 수 감소와 동기화 현상을 방지하기 위해 도입한 무작위 지연기법은 너무 단순하여 수많은 참가자들이 동시에 피드백 정보를 교환할 때에 피드백 정보 전송시간의 동기화 현상을 피하지 못해 네트워크에 혼잡 상태를 유발할 수 있다. 이러한 혼잡을 예방하기 위한 기존의 RTP/RTCP 확장 기법의 피드백 정보 전송지연은 송신자가 수신자의네트워크 상태에 따라 효율적으로 전송을 제어할 수 없게 한다. 본 논문에서는 RTP/RTCP 확장성을 증가시키는 기존의 기법들의 성능을 평가하고, 확장성 증가와 동시에 성능이 향상된 RTP/RTCP 확장 기법을 제안한다. 본 논문에서는 확장성 증가와 피드백 지연 정도를 줄이기 위해 빠른 제고 기법을 제안한다. 빠른 재고 기법은 두가지 세부 기법으로 나누어지는데, 첫째는 네트워크의 상태의 변화에 따라 RTCP피드백 정보의 전송지연 정도를 조절하는 것이고, 둘째는 무작위 지연을 선택적으로 조정하려 피드백 정보를 오랜 기간 동안에 보내지 못한 참가자에게 우선권을 주는 것이다. 본 논문에서는 시뮬레이션을 통해 제안된 확장성 기법을 이용할 때에 기존 방식에 비해 거의 비슷한 확장성을 보이면서도 초기 RTCP패킷 전송지연이 50%정도 감소함을 보여준다.구현되고 있다.팔일 전송 기법을 각각 제시하고 실험을 통해 이들의 특성을 비교분석하였다.미에서 uronic acid 함량이 두 배 이상으로 나타났다. 흑미의 uronic acid 함량이 가장 많이 용출된 분획은 sodium hydroxide 부분으로서 hemicellulose구조가 polyuronic acid의 형태인 것으로 사료된다. 추출획분의 구성단당은 여러 곡물연구의 보고와 유사하게 glucose, arabinose, xylose 함량이 대체로 높게 나타났다. 점미가 수가용성분에서 goucose대비 용출함량이 고르게 나타나는 경향을 보였고 흑미는 알칼리가용분에서 glucose가 상당량(0.68%) 포함되고 있음을 보여주었고 arabinose(0.68%), xylose(0.05%)도 다른 종류에 비해서 다량 함유한 것으로 나타났다. 흑미는 총식이섬유 함량이 높고 pectic substances, hemicellulose, uronic acid 함량이 높아서 콜레스테롤 저하 등의 효과가 기대되며 고섬유식품으로서 조리 특성 연구가 필요한 것으로 사료된다.리하였다. 얻어진 소견(所見)은 다음과 같았다. 1. 모년령(母年齡), 임신회수(姙娠回數), 임신기간(姙娠其間), 출산시체중등(出産時體重等)의 제요인(諸要因)은 주산기사망(周産基死亡)에 대(對)하여 통계적(統計的)으로 유의(有意)한 영향을 미치고 있어 $25{\sim}29$세(歲)의 연령군에서, 2번째 임신과 2번째의 출산에서 그리고 만삭의 임신 기간에, 출산시체중(出産時體重) $3.50{\sim}3.99kg$사이의 아이에서 그 주산기사망률(周産基死亡率)이 각각 가장 낮았다. 2. 사산(死産)과 초생아사망(初生兒死亡)을 구분(區分)하여 고려해

• Journal of KIISE:Computing Practices and Letters
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• v.8 no.3
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• pp.328-335
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• 2002

When we want to transfer video streaming data through computer networks, we will have to be allocated a larger bandwidth compared to a general application. In general, it causes a serious network overload inevitably due to the limited bandwidth. In this paper, in order to resolve the problem, we haute taken a method for transmitting video streaming data using RTP and RTCP. With RR(Receiver Report) packet in RTCP we will test it to check whether the traffic in a network has occurred or not. If it happened, we haute tried to reduce the overall network traffic by dynamically changing the quantization factor of the Motion JPEG that is one of the encoding styles in JMF. When the ratio of the average of transmission for each session to the average of overall transmission is greater than 7%, we should adjust the amount of data to be transmitted for each session to reach the session mean values. The experimental results show that the proposed method taken here reduces the overload effectively and therefore improves the efficiency for transmitting video streaming data.

• Journal of Korea Multimedia Society
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• v.12 no.5
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• pp.653-663
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• 2009

Synchronization between media is an important aspect in the design of multimedia communication-system. This paper proposes a precise media synchronization mechanism for video and audio transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a precise mechanism for media synchronization between video and audio, we suggest an efficient media synchronization algorithm based on NPT (Normal Play Time) which can be derivable from the timestamp information in the header part of RTP packet generated for the transport of video and audio. In the proposed method, we do not need to send and process any RTCP SR (sender report) packet which is required for conventional media synchronization scheme, and accordingly could reduce the number of required UDP ports and the amount of control traffic injected into the network.

• Proceedings of the Korean Information Science Society Conference
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• 2001.10c
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• pp.109-111
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• 2001

본 논문은 RTP/RTCP 표준 프로토콜을 이용하는 멀티미디어 스트리밍 환경에서 콘텐츠 특성에 영향받지 않고 프로토콜 수준의 캐슁을 지원하는 캐쉬 서버에 대하여 지술한다. 또한 RTP 패킷 단위로 캐슁된 콘텐츠에 대한 스트리밍 기법과 캐쉬 제어 기법을 기술한다.

• The Transactions of the Korea Information Processing Society
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• v.5 no.12
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• pp.3047-3062
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• 1998

This paper deseribes the desihn and the implementation of the Realtime Transport Protocol(RTP)/ Rdaltime Control Protocol(RTCP) (RFC 1889,1890) that is used to transmit the audio/video data to any destination and to feedback the Quality of Service (QoS) information of the received media data to the sender, in the teleconferencing systems proposed by ITU-T. These protocols are implemented with multi thead technique and run on top of UDP/IP-Multicast through the socket interface as the underlying protocol. The upper layer is impelmented such that in can be accessed by the H245 comference control protocol. The RTP packetizes the digitized audio/video data from the encoder info a fixed format, and multieast to the participants. The RTCP monitors RTP packets and extracts the QoS values from it such as round-trip delay, jiter and packet loss to form RTCP packets and non periokically sends them to the sender site. In this Paper, we also descritx the study of measurement and analysis for QoS factors that observed on performing teleconferencing system over Internet. The results from this experiment is indicate that RTT and Jitter value are acceptable even entwork load is high. However, it appears that packet loss rate is high in daytime and most losses periods have length one or two.