• 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.

• Proceedings of the Korean Information Science Society Conference
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• 2001.04a
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• pp.292-294
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• 2001

VoIP(Voice over IP) system은 현재 크게 2가지 형태로 진행되어가고 있다. 첫 번째는 H.323을 이용한 방법이고, 두 번째는 SIP(Session Initiation Protocol)를 이용한 방법이다. H.323은 실제 데이터를 전송하기전 호처리에 많은 signaling이 이루어지는 관계로 SIP보다 많은 RTT(Round Trip Time)를 소모하게 된다. 따라서 매우 복잡하고, LAN환경을 바탕으로 만들어서 확장성면에서도 여러 문제점을 가지고 있다. 그래서 본 논문은 호처리는 SIP를 이용하고, 실제 음성전송은 RTP(Real-Time Transport Protocol)와 RTCP(RTP Control Protocol)를 이용하는 시스템 구현을 제시한다. RTP는 실시간 특성을 가지는 데이터에 대해서 종단간 전송 서비스를 제공해주는 프로토콜로, 어떠한 인코딩에도 적합한 프레임워크를 제공한다. 그런데, RTP는 완전한 하나의 프로토콜이 되기 위해서는 RTP와 페이로드 포맷이 함께 제공되어야 하므로, 구현시스템은 음성신호를 PCM(Pulse Code Modulation), ADPCM(Adaptive Differential PCM)등의 여러 압축기술을 이용하여 파일을 생성하여 실시간으로 RTP와 RTCP를 이용하여 전송하는 방법을 제시한다.

• Proceedings of the IEEK Conference
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• 2003.11a
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• pp.255-258
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• 2003

TCP is taking over 95％ among the Internet traffics. Recently the demands of multimedia services in the Internet has been increasing. These multimedia services mostly need real-time deliverly, and then RTP has been a de facto to transmission protocol for these real-time multimedia services. RTP uses UDP as its underlying transport protocol, and thus it does not support any rate and congestion control. Thus, for fair use of the Internet bandwidth with TCP traffics. RTP also needs a rate control. One constraint of RTP is that the feedback information(delivered by, RTP's twin protocol, RTCP) is recommended to be sent no less than 5 seconds. In this paper, we propose a TCP-friendly RTP rate control which use only RTCP feedback information at every 5 seconds. The experiment results show that our proposed algorithm works. But, it is found that we need more time to test the effects of parameters and policies of the algorithms, which will be reported later.

• Proceedings of the Korean Information Science Society Conference
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• 2006.06d
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• pp.229-231
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• 2006

VoD(Video on Demand)나 IP 텔레폰, 화상채팅과 같은 멀티미디어 서비스는 일정한 대역폭의 자원을 요구하고 지터, 전송 지연에 매우 민감한 데이터이다. 멀티미디어 서비스의 QoS 보장과 TCP와 형평성을 위하여 TFRC, RAP, TLFC 와 같은 흐름제어 기법이 사용되고 있다. 하지만 흐름제어 기법에서 네트워크 판단을 위해서 사용하고 있는 RTT값은 서버와 클라이언트 사이의 왕복 전송 지연시간을 측정한 값으로써 클라이언트에서 서버로 전송되는 피드백 정보의 전송 지연시간에 따라서 오차가 발생한다. 본 논문에서는 서버에서 클라이언트로 단방향으로 데이터가 전송되는 특징을 가진 스트리밍 서비스를 위하여 RTP/RTCP를 이용한 단방향 전송 지연 측정을 제안한다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.49-54
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• 2009

Synchronization between media is an important aspect in the design of multimedia streaming system. This paper proposes a precise media synchronization mechanism for digital 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 streams. With 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 Korea Contents Association Conference
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• 2007.11a
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• pp.60-64
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• 2007

Increasing using multimedia services as VoIP, Video conference, DMB, IPTV, etc, it is necessary to increase network traffics and develop the mechanism about a flow control for real time traffic transmission. In order to transfer realtime multimedia data, the transfer rate can be control on network state data measuring packet losses of a receiver and delay time of packets through getting periodical feedback RTP/RTCP packet. This paper describes using efficiant flow control on multicast that can reduce errors according to getting feedback tranfer delay and proposes the mechanism that can adapt dynamic change of network. In simulation, the transfer rate can efficiently be control on dynamic change of network and it makes the maximum of the use of a bandwidth and the minimum of packet losses.

• The Journal of the Acoustical Society of Korea
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• v.21 no.4
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• pp.349-360
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• 2002

In ITU H.323 teleconference system, the RTP/RTCP protocol is offered to transfer real-time multimedia stream. Both sender and receiver hate experience in packet loss and jitter which result from network congestion over Internet. Audio quality over Internet depends on the number of lost packets and on jitter between successive packets. The goal of our study is to improve the speech quality over Internet by checking the packet loss characteristics of the network and adopting the but for control management mechanism at the receiver. We suggest a dynamic redundant audio transmission mechanism which examines the packet loss rate and uses the feedback information through RTCP.

• The Transactions of the Korea Information Processing Society
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• v.5 no.10
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• pp.2641-2653
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• 1998

• Proceedings of the Korea Multimedia Society Conference
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• 2000.11a
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• pp.323-326
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• 2000

시간에 의존적인 대용량의 멀티미디어 데이터를 다양한 네트웍 환경에 맞도록 전송하는 방법이 다양하게 연구되고 있다. 또한 멀티미디어 데이터를 실시간으로 전송하기 위한 프로토콜로 RTP(Real-time Transport Protocol)을 적용하여 하고 있으며, 다양한 상태의 네트웍을 통한 전송에 대한 피드백(feedback) 정보인 RTCP(Real-time Transport Control Protocol) 정보를 이용하여 QoS(Quality of Service)를 관리하는 메커니즘이 연구되고 있다. 본 논문에서는 RTP/RTCP 기반의 실시간 멀티미디어 데이터 전송을 위하여 멀티캐스팅(multicasting) 환경 하에서 각기 다른 대역 폭에 대한 동적인 QoS 관리 플랫폼을 제안한다.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2004.11a
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• pp.178-178
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• 2004