• 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.

• 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 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 Broadcast Engineering
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• v.13 no.6
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• pp.930-940
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• 2008

This paper proposes a prototype realization of multi-view HD video transport system with synchronized multiplexing over IP networks. The proposed synchronized multiplexing considers the synchronization during video acquisition and the multiplexing for the interactive view-selection during transport. For the synchronized acquisition from multiple HDV camcorders through IEEE 1394 interface, we estimate the timeline differences among MPEG-2 compressed video streams by using global time of network between the cameras and a server and correct timelines of video streams by changing the time stamp of the MPEG-2 system stream. Also, we multiplex a selected number of acquired HD views at the MPEG-2 TS (transport stream) level for the interactive view-selection during transport. Thus, with the proposed synchronized multiplexing scheme, we can display synchronized HD view.

• Journal of Broadcast Engineering
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• v.16 no.3
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• pp.530-541
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• 2011

In this paper, we propose an effective network-adaptive ARQ based error control scheme to provide video streaming services through IP networks where packet error usually occurs. If time delay and feedback channel are allowed, client can request server to retransmit lost packets through IP networks. However, if retransmission is unconditionally requested without considering network condition and number of simultaneous feedback messages, retransmitted packets may not arrive in a timely manner so that decoding may not occur. In the proposed ARQ, a client conditionally requests retransmission based on assumed network condition, and it further determines valid retransmission time so that effective ARQ can be applied. In order to verify the performance of the proposed adaptive ARQ based error control, NIST-Net is used to emulate packet-loss network environment. It is shown by simulations that the proposed scheme provides noticeable error resilience with significantly reduced traffics required for ARQ.

• Proceedings of the IEEK Conference
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• 2009.05a
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• pp.16-18
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• 2009

VoIP(Voice over IP) is the technology to transport voice and video over IP networks such as Internet. Today, VoIP technology is viewed as the right choice for provide voice, video, and data communication over next generation network. We are sure that the multipoint VoIP will help enhancing the various application services in ubiquitous environment. The paper shows multipoint VoIP system implemented with end-point mixing model and introduces various embedded systems such as UFC(Ubiquitous Fashionable Computer), tourist guide terminal and industrial terminal which use the multipoint VoIP.

• Journal of Satellite, Information and Communications
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• v.9 no.4
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• pp.38-41
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• 2014

Due to the diversity of device performances and the possibility of providing multimedia service through many networks, media consumption environment has been changed in various ways. In recent, mobile services and broadcasting services started to provided through IP network. According to this flow changes, there was great demand for various multimedia services through heterogenous networks. To solve this problem, MMT was suggested as a standard of multimedia transport which replaces MPEG-2 TS(Transport Stream) which is used for HD(High Definition) transport. MMT can be provided various broadcasting based on IP through terrestrial, satellite, cable broadcasting networks, and also deliver multimedia through many networks at the same time. MMT was risen as a method of servicing UHD broadcasting. In this paper, we present an implementation of service architecture for hierarchical UHD broadcasting over heterogenous networks using MMT.

• 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.

• ETRI Journal
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• v.38 no.5
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• pp.838-846
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• 2016

The objective of digital broadcasting has evolved from providing a plain video service to offering a realistic visual experience. Technologies such as 3DTV and UHDTV have been suggested to achieve this new objective by providing an immersive and stereoscopic visual experience. However, owing to the high bandwidth requirements of such services, the broadcasting industry has faced a challenge to find a new transport mechanism for overcoming the bandwidth limitation. The standardization organizations, the Advanced Television Systems Committee, Digital Video Broadcasting, and Telecommunications Technology Association, have been working on the integration of broadcasting and a broadband network (IP) to resolve the bandwidth issue of realistic video services. This paper introduces a frame-level timeline synchronization and transport system target decoder model for providing a stable 3DTV service over a hybrid network. The experimental results indicate that the proposed technologies can be successfully adopted as a reference model in a broadcast-broadband hybrid 3DTV service and other IP-associated hybrid broadcasting services.

• Journal of Korea Multimedia Society
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• v.10 no.3
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• pp.335-347
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• 2007

VoIP (Voice over IP) is a technology to transport video and voice traffic over IP networks such as Internet. Today, the VoIP technology is viewed as the right choice for providing voice, video, and data communication among various terminals over the next generation network. This paper discusses a multipoint VoIP implementation with end-point mixing model which can support multipoint conference without a conference bridge. The multipoint VoIP is implemented with SIP (Session Initiation Protocol), and supports STUN (Simple Traversal of UDP Through NATs) since it works in an asymmetric NAT (Network Address Translator) environment. The characteristics of this paper are as follows. It is possible that all terminals in the hierarchical conference don't receive the duplicated media information because we use the end-point mixing model with the new media processing module. And, the paper solves the problem that the hierarchical conference session should be separated into several sessions when a mixing terminal terminates the hierarchical conference session.