• 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 Broadcast Engineering
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• v.17 no.4
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• pp.596-610
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• 2012

This paper proposes a method to provide the high quality 3D broadcasting service in a mobile broadcasting system. In this method, audio and video data are delivered through a heterogeneous network, consisting of a mobile network as well as a broadcasting network, due to the limited bandwidth of the broadcasting system. However, it is more difficult to synchronize the left and right video frames of a 3D stereoscopic service, which come through different types of networks. The proposed method suggests the use of the offset from the initial timestamp of RTP (Real Time Protocol) to determine the order of frames and to find the pair of a left and a right frame that must be played at the same time. Additionally, a new signaling method is introduced for a mobile device to request a 3D service and to get the initial RTP timestamp.

• 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.

• Journal of Korea Multimedia Society
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• v.12 no.5
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• pp.653-663
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• 2009

Synchronization between media is an important aspect in the design of multimedia communication-system. This paper proposes a precise media synchronization mechanism for 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. In 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.

• Journal of Korea Multimedia Society
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• v.10 no.8
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• pp.1039-1051
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• 2007

In this paper, a new lip synchronization control method is proposed to achieve very low delay in the video communication. The lip control is so much vital in video communication as delay reduction. In a general way, to control the lip synchronization, both the playtime and capture time calculated from RTP time stamp are used. RTP timestamp is created by stream sender and sent to the receiver along the stream. It is extracted from the received packet by stream receiver to calculate playtime and capture time. In this paper, we propose the method of searching most adjacent corresponding frame of the audio signal, which is assumed to be played with uniform speed. Encoding buffer of stream sender is removed to reduce the buffering delay. Besides, decoder buffer of receiver, which is used to correct the cracked packet, is resulted to process only 3 frames. These mechanisms enable us to achieve ultra low delay less than 100 ms, which is essential to video communication. Through simulations, the proposed method shows below the 100 ms delay and controlled the lip synchronization between audio and video.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2015.11a
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• pp.118-120
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• 2015

본 논문에서는 IP 망을 통한 다윈 스트리밍 미디어 전송 서비스에서 압축된 미디어를 RTP 패킷화하여 전송할 때 RTP 패킷의 헤더에 기록될 타임스탬프 (timestamp) 정보로부터 유도해 낼 수 있는 Normal Play Time 정보를 활용하여 비디오와 오디오 간에 미디어 동기화 지원 방법을 제안한다. 모의실험을 통해 제안된 미디어 동기화 알고리듬을 적용함으로써 서로 다른 미디어 간에 정확한 동기화가 제공됨을 확인할 수 있었다.