• ETRI Journal
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• v.32 no.2
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• pp.281-291
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• 2010

Scalable video coding (SVC) has been standardized as an extension of the H.264/AVC standard. This paper proposes a practical real-time transport protocol (RTP) packetization scheme to transport SVC video over IP networks. In combined scalability of SVC, a coded picture of a base or scalable enhancement layer is produced as one or more video layers consisting of network abstraction layer (NAL) units. The SVC NAL unit header contains a (DID, TID, QID) field to identify the association of each SVC NAL unit with its scalable enhancement layer without parsing the payload part of the SVC NAL unit. In this paper, we utilize the (DID, TID, QID) information to derive hierarchical spatio-temporal relationship of the SVC NAL units. Based on the derivation using the (DID, TID, QID) field, we propose a practical RTP packetization scheme for generating single RTP sessions in unicast and multicast transport of SVC video. The experimental results indicate that the proposed packetization scheme can be efficiently applied to transport SVC video over IP networks with little induced delay, jitter, and computational load.

• Proceedings of the Korea Information Processing Society Conference
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• 2000.04a
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• pp.155-158
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• 2000

텔리포인팅은 지역 시스템의 공유윈도우에서 텔리포인터를 통해 발생하는 이벤트를 원격지 시스템의 공유윈도우에 동기적으로 표현하는 기법이다. 이러한 텔리포인팅은 다양한 CSCW분야에서 응용되고 있다. 본 논문은 다양한 텔리포인팅의 기능을 제공할 수 있고 응용프로그램에 상관없이 응용이 가능한 텔리포인팅 모듈을 설계하고 구현한다. 텔리포인팅 모듈은 텔리포인팅 이벤트의 송수신을 위해 RTP(Real-Time Transport Protocol)을 사용한다. 구현된 모듈이 잘 동작함을 보이기 위해 다가간의 텔리포인팅을 지원하는 화이트보드를 구현하고 인터넷 상에서 시연하였다.

• The Transactions of the Korea Information Processing Society
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• v.6 no.9
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• pp.2512-2523
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• 1999

In this paper, we present the design and implementation results of RTP(Real-time Transport Protocol) which is introduced to provide end-to-end delivery service for multimedia data with real-time characteristics and to provide QoS(quality of service) monitoring services. Conventional RTP communication modules have some problems such as poor reusability and inefficiency. Reusability problem stems from the inherent characteristics of RTP that is framework is deliberately not complete, and so, RTP should have been designed and implemented in application-imbedded from. The RTP communication module, proposed in this paper, is designed and implemented in application-independent from in order to be used in all kinds of high level applications independent of their architectures and functions.

• The KIPS Transactions:PartA
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• v.9A no.1
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• pp.29-36
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• 2002

For the efficient collaboration awareness, the CSCW (Computer-Supported Cooperative Work) systems provide functions such as telepointing, snap-shot, and document/project tracking. Among these, telepointing allows an event occurrence on the shared window of a local computing system to be presented in remote computing systems. Although telepointing has been mentioned in the literature, few research papers deal with the functional requirements and protocols for telepointing. This parer describes the functional requirements and payload format for RTP (Real-Time Transport Protocol)-based telepointing. In order to evaluate the performance of the proposed RTP-barred telepointing, we have implemented a whiteboard and compared it with that of the Microsoft Netmeeting. The evaluation results show that the proposed telepointing performs better in terms of CPU usage and the amount of network traffic.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.4
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• pp.839-844
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• 2011

IPTV contents are based on MPEG2-TS(Moving Picture Expert Group-2 Transport Stream). The Created content is delivered to each home's Set-Top box through the Internet service network, which will be visible on television through the decoding process. The using and spending of content expanding were quite limited in the other existing lines' devices. For this end, provided methods are to convert MPEG2-TS to other format and then transmit it. These methods are causing not only an overload to the system but also its increasing the hardware resource requirements. In this paper, the MPEG2-TS of IPTV's content using RTP(Real time Transport Protocol) provided by the applicable variety of devices and environmental system, was designed and implemented. Through this paper, the existing MPEG2-TS is providing other devices for distribution and consumption that can give better content to the consumers which means that the QoS(Quality of Service) was enhanced. And, the increased consumption and use of content will help the IPTV business to get more development and activation.

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

• The Transactions of the Korean Institute of Electrical Engineers P
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• v.65 no.4
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• pp.326-334
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• 2016

In this paper for developing and implementing the realtime HD level video streaming transmitter which is operated on the multi-platform in all network and client environment compared to the exist video live streaming transmitter. We design the realtime HD level video streaming transmitter supporting the multi-platform using the TMS320DM386 video processor of T.I company and then porting the Linux kernel 2.6.29 and implementing the RTSP(Real Time Streaming Protocol)/RTP(Real Time Transport Protocol), HLS(Http Live Streaming), RTMP(Real Time Messaging Protocol) that can support the multi-platform of video stream protocol of the received equipments (smart phone, tablet PC, notebook etc.). For proving the performance of developed video streaming transmitter, we make the testing environment for testing the performance of streaming transmitter using the notebook, iPad, android Phone, and then analysis the received video in the client displayer. In this paper, we suggest the developed the Realtime HD(High Definition) level Video Streaming transmitter performance data values higher than the exist products.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.305-308
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• 2000

IP 네트웍을 통해 실시간 비디오를 전송할 때 네트윅 특성을 고려하여 비디오 데이터를 부호하면 패킷 손실로 인한 품질 열화를 최소화하여 더 좋은 품질의 비디오를 얻을 수 있다. 이를 위해 현재 네트윅의 정보를 빠르고 정확하게 얻어내는 메커니즘과 부호화 변수를 네트윅 상황에 적응적으로 조절하여 패킷 손실에 강인한 압축 및 전송 메커니즘의 두 가지 기능이 요구된다. 첫번째 메커니즘은 RTP(Real Time Transport Protocol)을 통해 구현될 수 있으며, 두 번째 메커니즘을 위해 본 연구에서는 다중기술 변환부호화(Multiple Description Transform Coding) 기법을 적용한 비디오 부호화 알고리즘을 제안한다. RTP에서 제공하는 RTCP(Real Time Control Protocol) 정보를 이용하여 현재 네트웍 정보를 얻을 수 있으며, 다중기술 변환부호화 기법을 이용하여 현재의 패킷 손실률에서 최적의 품질을 보장하도록 부호화 변수를 조절할 수 있다. 본 논문에서는 다중기술 변환부호화 기법을 비디오 부호화에 적용하여 순수 비디오 정보에 추가되는 잉여 정보량과 패킷 손실에 대한 강인성 사이의 관계를 도출함으로써 다중기술 변환부호화 기법이 네트웍 적응적 부호화 방식에 적합한 방식임을 제시한다.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.1
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• pp.54-61
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• 2007

In this paper, on-demand service streaming protocol for user-created contents in digital cable broadcasting environment is considered. In order to provide variety media contents service like UCC in digital cable broadcasting environment, the same service model as RVOD(Real Video on Demand) is required and the different interface for each broadcasting platform is needed in a current OCAP environment. Using return path based RTP, we separate existing VOD stream band into broadcasting and VOD stream band. In order to broaden On-demand service, consistent expansion of the infrastructure same as live broadcasting system is inefficient in the digital cable broadcasting environment. Using existing network protocol, the service which is insensitive to the infrastructure for VOD service becomes possible. Therefore we considered the analysis of the class of download available structure in the Set-Top-Box for RTP(Real-time Transport Protocol) and designing the decoding available streaming server for UCC transcoding and transmission in the receiver. Designing a efficient VOD service and system under the broadcasting environment gives a expansion of On-Demand service and more chance to upload and utilization of contents.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.142-144
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• 2002

VoIP(Voice over IP) 시스템을 구현함에 있어서 호설정을 처리하는 여러 프로토콜이 제안되고 있는 가운데 IETF(Internet Engineering Task Force)에서 제안한 SIP(Session Initiation Protocol)는 텍스트 기반의 프로토콜로서 구현과 파싱이 쉬운 등 많은 장점을 가지고 있어 차세대 VoIP의 표준으로 자리잡고 있다. 또한, 뛰어난 확장성을 가지고 있어 다양한 서비스에 적용할 수 있는 호설정 프로토콜이다. 본 논문에서는 SIP를 이용한 VoIP 시스템을 구현함에 있어 주요 구성요소 중 하나인 UA(User Agent)를 2002년 6월에 발표된 새로운 SIP 버전에 맞추어 개발하였다. 본 UA는 플랫폼에 독립적으로 기능을 할 수 있도록 자바(Java)를 사용하여 GUI(Graphical User Interface)환경으로 구현하였다 그리고 RTP(Real-time Transport Protocol) 전송 모듈을 통하여 호설정이 이루어진 후 실제 음성과 화상통신이 이루어지는 부분을 포함하였다.