• The Transactions of the Korea Information Processing Society
• /
• v.3 no.1
• /
• pp.127-143
• /
• 1996

Advanced countries are very active in deploying the National Information I infrastructure which provides universal service to promise fruitful quality of futuree life. Even in the distributed environment, we can closely converse, work together, and share information in a very convenient way. This is actually enabled with the help of groupware technology, which are currently focused and researched in a larger popularity. The aim of this study is to design a portable pack for group communications management to support the development of groupware application. In the paper we begin with technical survey, continue to build our own model for group communications man agement, and design its architecture and procedure. We also suggest group addressing mechanism under Internet environment such as how to create IP multicast address and IP port number dynamically and as a globally unique value for the communication session, with the help of the hierarchical and distributed address managers. We also indicate the reliable data transmission services to remedy the unreliable feature of the UDP multicast services, and finally the architecture/ applied to support the practical applications is briefly discussed for verification of the designed concept.

• Journal of KIISE:Computer Systems and Theory
• /
• v.26 no.8
• /
• pp.975-983
• /
• 1999

최근 네트워크의 속도가 빨라지고 멀티미디어 데이터를 다루기 위한 기술들이 개발됨에 따라 많은 멀티미디어 응용 프로그램들이 인터넷에 등장하고 있다. 그러나 이들 응용프로그램들은 수신자에게 전송되는 영상.음성의 품질이 낮기 때문에 기대만큼 빠르게 확산되지 못하고 있다. 영상.음성의 품질이 낮은 이유는 현재 인터넷이 실시간 응용프로그램이 요구하는 만큼 빠르고 신뢰성 있게 데이터를 전송할 수 없기 때문이다. 현재 인터넷의 내부구조를 바꾸지 않고 품질을 높이기 위해 많은 연구들이 진행되고 있는데 그 중 하나는 동적으로 변화하는 인터넷의 상태에 맞게 멀티캐스트 트래픽의 전송율을 조절하는 종단간의 흐름제어이다. 본 논문은 기존의 흐름제어 기법인 IVS와 RLM의 성능을 개선시키기 위한 두 가지 흐름제어 기법을 소개한다. IVS는 송신자가 주기적으로 측정된 네트워크 상태에 따라 전송율을 일정하게 조절한다. 송신자가 하나의 데이타 스트림을 생성하는 IVS와는 달리 RLM에서는 송신자가 계층적 코딩에 의하여 생성된 여러개의 데이타 스트림을 전송하고 각 수신자는 자신의 네트워크 상태에 맞게 데이타 스트림을 선택하는 기법이다. 그러나 IVS는 송신자가 전송율을 일정하게 증가시키고, RLM은 각자의 네트워크 상태를 고려하지 않고 임의의 시간에 하나 이상의 데이타 스트림을 받기 때문에 성능을 저하시킬 수 있다. 본 논문에서는 TCP-like IVS와 Adaptive RLM이라는 두 가지 새로운 기법을 소개한다. TCP-like IVS는 송신자가 전송율을 동적으로 결정하고, Adaptive RLM은 하나 이상의 데이타 스트림을 받기 위해 적당한 시간을 선택할 수 있다. 본 논문에서는 시뮬레이션을 통해 여러 가지 네트워크 구조에서 두 가지 방식이 기존의 방식에 비하여 더욱 높은 대역폭 이용율과 10~20% 정도 적은 패킷손실율을 이룬다는 것을 보여준다.Abstract Nowadays, many multimedia applications for the Internet are introduced as the network gets faster and many techniques manipulating multimedia data are developed. These multimedia applications, however, do not spread widely and are not fast as expected at their introduction time due to the poor quality of image and voice delivered at receivers. The poor quality is mainly attributed to that the current Internet can not carry data as fast and reliably as the real-time applications require. To improve the quality without modifying the internal structure of the current Internet, many researches are conducted. One of them is an end-to-end flow control of multicast traffic adapting the sending rate to the dynamically varying Internet state. This paper proposes two flow-control techniques which can improve the performance of the two conventional techniques; IVS and RLM. IVS statically adjusts the sending rate based on the network state periodically estimated. Differently from IVS in which a sender produces one single data stream, in RLM a sender transmits several data streams generated by the layered coding scheme and each receiver selects some data streams based on its own network state. The more data streams a receiver receives, the better quality of image or voice the receiver can produce. The two techniques, however, can degrade the performance since IVS increases its sending rate statically and RLM accepts one more data stream at arbitrary time regardless of the network state respectively. We introduce two new techniques called TCP-like IVS and Adaptive RLM; TCP-like IVS can determine the sending rate dynamically and Adaptive RLM can select the right time to add one more data stream. Our simulation experiments show that two techniques can achieve better utilization and less packet loss by 10-20% over various network topologies.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2010.10a
• /
• pp.643-645
• /
• 2010

The Internet-based multimedia services such as IPTV is being expanded with the development of technology to support the convergence of broadcasting and telecommunications technology for the control seems to be growing larger. Especially for the real-time TV broadcast multicast control technology to support the authentication and resource control, in addition to the technology services that enhance the value of technology for a variety of services in both directions seems to be developed. And, Internet-based transmission system transmit the video content for the video content delivery using RTP(Real Time Transport Protocol). Standardization body, IETF(Internet Engineering Task Force) within the RTP, according to a variety of audio and video formats only transmission format(RTP Payload Format) Establish a separate standard and scalable video content "RTP Payload Format for SVC(Switched Virtual Connection) Video" the standardization is currently processing. In this paper we are improving the quality of broadcasting and telecommunication systems, so that the upper layer by the application can react adaptively to the existing MPEG2-TS and RTP who are provided by a variety of content applied to a variety of devices consumers ETE(End- to-End) QoS(Quality of Service) for enhance the system who was designed and implemented.