• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4A
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• pp.421-430
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• 2006

In this paper, we propose a practical implementation of network-adaptive HD(high definition) MPEG-2 video streaming with a cross-layered channel monitoring(CLM) over the IEEE 802.11a WLAN(wireless local area network). For wireless channel monitoring, AP(access point) periodically measures the MAC(medium access control) layer transmission information and sends the monitoring information to a streaming server. This makes that the streaming server reacts more quickly as well as efficiently to the fluctuated wireless channel than that of the end-to-end monitoring(E2EM) scheme for the video adaptation. The streaming sewer dynamically performs the priority-based frame dropping to adjust the video sending rate according to the measured wireless channel condition. For this purpose, our streaming system nicely provides frame-based prioritized packetization by using a real-time stream parsing module. Various evaluation results over an IEEE 802.11a WLAM testbed are provided to verify the intended QoS adaptation capability The experimental results show that the proposed system can effectively mitigate the quality degradation of video streaming caused by the fluctuations of time-varying wireless channel condition.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1997.06a
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• pp.49-54
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• 1997

For the purpose of a flexible coded video transmission over a heterogeneous network, we propose a new packetization method for coded video data. The proposed method achieves small degradation of coded picture quality in case of packet discard at the network node and does not require heavy processing load for bitrate control operation. Computer simulation results show that the bitrate reduction from 384 kb/s to 192 kb/s does not cause severe degradation in picture quality.

• Proceedings of the Korea Information Processing Society Conference
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• 2009.11a
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• pp.69-70
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• 2009

단거리에서의 MCU들의 통신을 위해서 직렬방식의 통신이 많이 사용되어지고 있다. 이러한 시리얼 통신에서도 통신 규격을 변경하면 일대일 통신뿐만 아니라 다대다 통신을 통한 네트워크 구성도 가능하다. 네트워크로 구성되어지는 시리얼 통신에서 각 MCU들의 원활한 통신을 위한 방법이 필요하기 때문에, 본 논문에서는 그 방법을 Ethernet의 패킷 데이터그램 형태를 응용하여 시리얼 통신에서의 데이터 패킷을 메시지단위로 통신하는 방법을 제안한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.6C
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• pp.471-479
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• 2008

In this paper, we propose a novel algorithm to automatically generate an RTP timestamp value that is required for the RTP packetization in order to transmit SVC video over various If networks such as Internet. Unlike the conventional single layer coding algorithms such as H.263, MPEG-4 and H.264, SVC generates a multi-layered single bitstream which is composed of a base layer and one or more enhancement layers in order to simultaneously provide temporal, spatial, and SNR scalability. Especially, in order to provide temporal scalability based on hierarchical B-picture prediction structure, the encoding (or transmission) and display order of pictures in SVC coding is completely decoupled. Thus, the timestamp value to be specified at the header of each RTP packet in video transmission does not increase monotonically according to the display time instant of each picture. Until now, no method for automatically generating an RTP timestamp when SVC video is loaded in a RTP packet has teen introduced. In this paper, a novel automatic RTP timestamp generation method exploiting the TID (temporal ID) field of the SVC NAL unit header is proposed to accommodate the SVC video transmission.

• Journal of Korea Multimedia Society
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• v.14 no.1
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• pp.54-75
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• 2011

PoC(Push-to-talk Over Cellular) is an integrated technology of group voice calls, video calls and internet based multimedia services. If a PoC user can not participate in the PoC session for various reasons such as an emergency situation, lack of battery capacity, then the user can use the PoC Box which has a similar functionality to the MM Box in the MMS(Multimedia Messaging Service). The RTSP(Real-Time Streaming Protocol) method is recommended to be used when there is a transmission session between the PoC box and a terminal. Since the existing VOD service uses a wired network, the packet size of RTSP-based VOD service is huge, however, the PoC service has wireless communication environments which have general characteristics to be used in RTSP method. Packet loss in a wired communication environments is relatively less than that in wireless communication environment, therefore, a buffering latency occurs in PoC service due to a play-out delay which means an asynchronous play of audio & video contents. Those problems make a user to be difficult to find the information they want when the media contents are played-out. In this paper, the following techniques and methods were proposed and their performance and superiority were verified through testing: cross-over dual reception buffering technique, advance partition multi-reception buffering technique, and on-demand multi-reception buffering technique, which are designed for effective picking up of information in media content being transmitted in short amount of time using RTSP when a user searches for media, as well as for reduction in playback delay; and same-priority packetization transmission method and priority-based packetization transmission method, which are media data packetization methods for transmission. From the simulation of functional evaluation, we could find that the proposed multiple receiving buffering and packetizing methods are superior, with respect to the media retrieval inclination, to the existing single receiving buffering method by 6-9 points from the viewpoint of effectiveness and excellence. Among them, especially, on-demand multiple receiving buffering technology with same-priority packetization transmission method is able to manage the media search inclination promptly to the requests of users by showing superiority of 3-24 points above compared to other combination methods. In addition, users could find the information they want much quickly since large amount of informations are received in a focused media retrieval period within a short time.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.2
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• pp.40-49
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• 1998

In this paper, we propose an efficient per-Trunk interworking mechanism between PSTN and ATM network assuming the situation ATM network interworks with PSTN during the evolution period. The proposed mechanism improves the cell payload utilization by mapping only the active charnnels of PSTN frame into ATM cell payload. Also, we propose the frame recovery mechanism to guarantee the frame sequence integrity. The proposed mechanism is compared with other possible ones in terms of cell payload utilization, cell packetization delay. We present the implementation structure of the interworking unit. the correctness of the mechanism and the feasibility of the implementation are verifid through the CAD simulation.

• 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.475-477
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• 2002

이미지 압축을 위한 새로운 기법 중의 하나가 웨이블릿 변환이다. 웨이블릿 변환된 이미지는 이미지 전체의 중요 정보가 한쪽으로 치우치는 경향을 보인다. 그러므로 웨이블 이미지를 순차적으로 네트웍상에 전송한다면 중요 정보가 포함된 패킷에 손실이 발생할 경우 이미지의 품질에 치명적인 영향을 미친다. 본 논문에서는 전송된 웨이블릿 이미지의 품질을 보장하기 위한 새로운 패킷타이징 기법을 제안한다. 즉 주파수 부대역(frequency subband)을 여러 개의 블록으로 나누고, 이 블록들을 서로 다른 패킷에 삽입하여 패킷의 중요도를 분산하는 것이다. 또한 주파수대역이 가장 낮은 부대역 정보를 중복하여 전송함으로써 이미지 전체의 품질을 보장한다. 제안된 기법의 성능을 평가하기 위해 객관적인 성능평가 단위인 PSNR과 주관적인 이미지 분석을 수행하였다. 실험결과 제안된 기법의 성능이 순차적인 블록 패킷 기법에 비하여 PSNR에서 1～4 ㏈ 높았으며, 이미지의 선명도도 높음을 알 수 있었다.

• Proceedings of the Korea Information Processing Society Conference
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• 2000.04a
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• pp.507-512
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• 2000

본 논문에서는 통합 스트리밍 프레임워크 (ISSA)[1][2]의 주요 모듈인 전송 관리자의 구현에 대한 개발 경험을 소개한다. 전송 관리자는 ISSA에서 지원하는 다양한 형식의 멀티미디어 데이터를 네트워크의 전송에 알맞게 분해(packetization), 조립(depacketization)하는 기능과 패킷화된 미디어 데이터를 실시간으로 전달하는 역할을 수행한다. 이러한 기능은 RTP/RTCP[3][4] 프로토콜을 이용하여 구현되었으며, 코드 레벨에서의 이기종 플렛폼간 호환성과 네트워크의 투명성을 보장하도록 설계 되었다.

• Journal of Broadcast Engineering
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• v.8 no.3
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• pp.259-267
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• 2003

In this paper, we propose a media processor to provide interactive services in mobile multimedia broadcasting environments. The proposed system Is designed to support various functionalities, such as generation of MPEG-4 IOD (Initial Object Descriptor)/OD(Object Descriptor)/BIFS (Binary Format for Scene) data, encapsulation of MPEG-4 AVC (Advanced Video Coding)/BSAC (Bit Sliced Arithmetic Coding) stream and generated IOD/OD/BIFS data into SL (Sync Layer) packet, packetization of SL packet into TS (Transport Stream) packet and multiplexing. The proposed media processor can provide MPEG-4 based interactive services for users.