• Proceedings of the Korean Information Science Society Conference
• /
• 2001.10c
• /
• pp.772-774
• /
• 2001

ITU-T의 H.332 프로토콜은 ITU-T의 H.323 프로토콜을 기반으로 하는 멀티미디어 화상회의의 제한된 확장성을 극복하기 위한 방법을 제시한다. H.332 멀티미디어 화상회의 구조는 H.323 단말기인 몇 명의 패널과 나머지 수백 수천의 청중인 RTP/RTCP receiver들로 이루어진다. 따라서 패널과 청중 사이의 콘트롤 신호는 RTCP가 된다. 그리고, 참가자가 수백 수천 일 때의 H.323 프로토콜의 능력교환의 어려움을 극복하기 위해 화상회의의 주제나 미디어 형식 등은 회의가 열리기 전에 미리 알려준다. 이것을 위해 IETF 의 SDP를 이용한다. 따라서, 패널의 SDP 전달과 청중의 SDP 해석은 일방적인 능력교환이라고 할 수 있다. 위와 같이 함으로써 H.323 프로토콜의 단점인 확장성을 극복할 수 있다. 본 논문은 패널과 청중의 멀티미디어 데이터 전달을 위한 RTP/RTCP의 구현과 능력교환을 위한 SDP의 구현을 바탕으로 H.332 멀티미디어 화상회의의 설계 및 구현에 대해서 기술한다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2002.05a
• /
• pp.189-194
• /
• 2002

It is thesis that based on for Croup-Transferring a method of Transferring-Control during real-time. By the development of multimedia, spread of high-speed communication network and generalization of various multimedia services, various techniques an studied by requiring application services for multi-users (meeting-service, remote-service, broadcasting-service, education-service and so on). So we focused on the multimedia group transferring techniques except communication network techniques and multimedia processing techniques, and presented algorithms about a mettled of group-management, a method of delay-control-management that analyzed a effective method of Croup-Control and characters of each group network and used RTP(Real-Time Protocol) / RTCP(Real-Time Control Protocol).

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

• Proceedings of the Korea Contents Association Conference
• /
• 2007.11a
• /
• pp.60-64
• /
• 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.

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

• Proceedings of the Korea Multimedia Society Conference
• /
• 2001.06a
• /
• pp.351-357
• /
• 2001

분산 컴퓨팅 기술의 발전과 네트워크의 고성능화는 대용량의 멀티미디어 데이터를 송.수신하여 처리할 수 있게 되었다. 그러나 실시간 및 이기종간의 데이터 전송에는 네트워크 전송지연과 프로그램 구현의 어려움 등과 같이 극복해야 하는 문제점들이 많이 있다. 이 문제들을 해결하기 위하여 본 논문에서는 객체지향형 분산처리 환경 하에서 OMG가 제시한 멀티미디어 서비스 설계를 확장하는 방안을 제시한다. 이를 위해서 CORBA와 JMF lava Media API 기반으로 Smart Explorer라고 불리는 효율적인 실시간 오디오/비디오 멀티미디어 소프트웨어 기반 구조를 제시하고 있다. 제어 데이터와 미디어 데이터의 전송 경로를 분리하고 효율적인 실시간 오디오/비디오 전송을 위해서 RTP/RTCP 프로토콜을 사용한다. 또한 제안하는 소프트웨어 기반 구조를 이용하여 오디오/비디오 스트림 시스템을 분산 객체 환경에 맞게 구현한 결과도 제시한다. 향후 이 시스템은 방송, 원격 교육, 화상회의 등 실시간 통신 소프트웨어에 응용될 수 있을 것으로 기대된다.

• Journal of Korea Multimedia Society
• /
• v.4 no.4
• /
• pp.297-305
• /
• 2001

Recently advances in high-speed networks and multimedia computer technologies allow new types of multimedia applications to manipulate large volumes of multimedia data. However, in the real time and/or the heterogeneous data transmissions, there are many difficulties such as network transmission delay, the implementation difficulties, and so on. To solve these problems, in this paper, we extend the method of the multimedia service design which is proposed by OMG. To do this, we suggest an efficient real time audio/video stream framework, called Smart Explorer, based un CORBA and JMF Java Media API. And we separate the transmission path of control data from that of media data and use RTP/RTCP protocol for efficient real time audio/video transmission. Also we show the appropriate implementation of the audio/video stream system based on our suggested framework Smart Explorer. In the future, we expect our audio/video stream system to be applied to the real time communication software such as broadcasting, distance learning, and video conferencing.

• Journal of KIISE:Computing Practices and Letters
• /
• v.7 no.1
• /
• pp.24-37
• /
• 2001

This paper deals with our experience for the design and implementation of Transport and Session managers in the Integrated Streaming Service Architecture (IS SA) that was developed by the authors. The ISSA is a streaming framework that allows to develop integrated multimedia streaming applications such as VOD system in unicast/multicast and real-time broadcastOive-cast). It also facilities standard real-time transport protocols such as RTp(Real-Time Transport Protocol)/RTCP(Real-time Control Protocol) and RTSP(Real-Time Streaming ProtocoD that allows to user openness. The ISSA supports diverse media fonnats and is independent from underlying networks and operating systems, and compatible with the global real-time multimedia database system (BeeHive) so that streaming media are efficiently retrieved, stored, and serviced. The role of the Transport Manager is to do packetization and depacketization for the different types of multimedia data and delivers the packetized media data in real-time. The Transport Manager uses RTP protocol. The role of the Session Manager is to establish and control the media channel by using RTSP protocol and to deliver the database transactions for the multimedia database by using RTTP(Real-Time Transaction Protocol) protocol. Both the Transport and Session Manager are doing their functions through the network interface in the ISSA that allows developers to various network programming interfaces and provides flexibility to the system.

• Journal of KIISE:Computing Practices and Letters
• /
• v.5 no.2
• /
• pp.157-167
• /
• 1999

편리한 사용 방법과 멀티미디어 데이터 서비스를 지원하는 웹은 현 인터넷 환경에서 이종 컴퓨터간의 주요한 정보 전달 수단으로 자리잡았다. 그러나 웹이 사용하고 있는 통신 프로토콜 스택만으로는 부상하고 있는 정보 형태인 실시간 멀티미디어 데이터의 서비스를 적절히 지원할수 없었다. 이에 본 논문에서는 실시간 전송 프로토콜인 RTP/RTCP를 웹 통신 프로토콜 스택에 추가하여 실시간 멀티미디어 세션의 서비스를 지원할수 잇도록 그 기능이 확장된 웹 시스템의 구현을 소개한다. 이는 MBone을 통해 실시간으로 멀티캐스트되고 있는 멀티캐스트 되고 있는 멀티미디어 세션의 서비스를 웹에서 지원할수 있도록 한 것으로, 기존의 MBone응용들을 수용한 형태를 가진다. 이를 위하여, MBone상에서 개설된 세션 정보를 수집하여 이를 관리하고 그 정보들을 웹 문서로 제공하는 프로그램인 '웹 SDR'과 MBone을 통해 멀티캐스트되는 실시간 멀티미디어 세션을 웹상에서 시청할수 있도록 웹 브라우저의 기능을 확장한 Windows 95 플랫폼 기반의 '실시간 멀티미디어 웹 플러그인'을 구현하였다 그리고 , 본 연구에서 개발한 수신 위주의 웹 시스템의 서비스 품질을 개선하기 위하여 현 인터넷 환경을 고려한 서비스 품질 제어 기법인 '관측 기반의 수신단 QoS 제어 알고리즘'을 제안하고, 이를 적용하여 실시간세션 서비스 품질이 개선됨을 보였다.

• The KIPS Transactions:PartD
• /
• v.9D no.2
• /
• pp.313-322
• /
• 2002

Multimedia conference systems are classified into the tightly coupled conference and the loosely-coupled conference, the former is a peer-to-peer-conference or multi-point-conference based un the ITU-T H.323 standard, and the latter is the H.332 standard extended from the ITU-T H.323 standard. Loosely-coupled conference based on the H.332 standard is composed of small participants with the capacity of the send-receive mode and large participants with the capacity of the receive-only mode. In the H.332 conference, a subject and a media type of the multimedia conference is pre-announced for conquest of the difficulty of capability negotiation, and used SDP of IETF for that. Therefore, the H.332-conference can conquer a defect of the H.323 multimedia conference scalability. This paper analyzes H.332 standard for the loosely-coupled conference and describes an implementation of the H.332 multimedia conference systems.