• 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.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.

• International Journal of Internet, Broadcasting and Communication
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• v.11 no.3
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• pp.59-63
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• 2019

VoIP technology is a technology for exchanging voice or video data through IP network. Various protocols are used for this technique, in particular, RTP(Real-time Transport Protocol) protocol is used to exchange voice data. In recent years, with the development of communication technology, there has been an increasing tendency of services such as "Kakao Voice Talk" to exchange voice and video data through IP network. Most of these services provide a service with security guarantee by a user authentication process and an encryption process. However, RTP protocol does not require encryption when transmitting data. Therefore, there is an exposition risk in the voice data using RTP protocol. We will present the risk of the situation where packets are sniffed in VoIP(Voice over IP) communication using RTP protocol. To this end, we configured a VoIP telephone network, applied our own sniffing tool, and analyzed the sniffed packets to show the risk that users' data could be exposed unprotected.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.20 no.9
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• pp.7-13
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• 2019

Unlike a typical video wall system, video wall systems used for integrated monitoring in smart city environments should be able to display various videos, images, and texts simultaneously. In this paper, we propose an Internet Protocol (IP)-based video wall system that has no limit on the number of videos that can be monitored simultaneously, and that can arrange the monitor screen layout without restrictions. The proposed system is composed of multiple display servers, a wall controller, and video source providers, and they communicate with each other through an IP network. Since the display server receives and decodes the video stream directly from the video source devices, and displays it on the attached monitor screens, more videos can be simultaneously displayed on the entire video wall. When one video is displayed over several screens attached to multiple display servers, only one display server receives the video stream and transmits it to the other display servers by using IP multicast communications, thereby reducing the network load and synchronizing the video frames. Experiments show that as the number of videos increases, a system consisting of more display servers shows better decoding and rendering performance, and there is no performance degradation, even if the display server continues to be expanded.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2004.11a
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• pp.204-204
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• 2004

• 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.13 no.1
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• pp.17-24
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• 2008

At the view of Application layer, there are many researches to achieve the cross-layer optimization with Physical layer. Scalable video coding is good example. Although it is necessary to consider the session layer which lies halfway between two layers, the research about that is insufficient. We present a feasible solution of dynamic session control for scalable video coding over IMS.

• The Journal of the Korea Contents Association
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• v.6 no.8
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• pp.20-29
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• 2006

This paper proposes a quality measurement model for video phone service over IP environment. Proposed model enhances conventional E-Model by using quality analysis and this model is suitable for ubiquitous environment. This research measures video phone quality by applying bust packet loss and recency effect. It uses delay and recency effect for compensating actual quality and recognized quality of user using NR and UR factor. Simulation results show that this model can provide more precise results than conventional model by considering recency effect of video phone service quality measurement model.

• The Journal of the Korea Contents Association
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• v.10 no.8
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• pp.59-65
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• 2010

This paper proposes an implementation of a hardware module for transmitting MPEG-2 TS data over the internet protocol (IP)-based network. This implementation consists of two modules; one is an encapsulation module which bridges between n TS packets, where $1\;{\leq}\;n\;{\leq}\;7$, and an IP packets, the other is a packet conversion module which extracts an DSM-CC PS packet from consecutive TS packets and then reconstructing an IP packet. So, these IP packets are carried over 150 megabits per second. Although overall work flow of the proposed DeMUX is based on the reference design of ALTERA, the DeMUX is enhanced by modifying it and performs more functions by adding a packet conversion module. The DeMUX is described by Verilog-HDL (hardware description language) and shows the faithful functionality and throughput through the simulation.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.54 no.8
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• pp.510-516
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• 2005

Rapid advances in telecommunication make emergency telemedicine possible that specialist offers medical care to an emergency case in moving vehicle. Although there were many telemedicine projects delivering the image or video of patient over several wireless networks, none of them considered effective solutions for optimizing video transmission over error-prone environments, such like wireless links. To alleviate the effect of channel errors on compressed video bit-stream, this paper analyzed the error resilient features of MPEG-4 standard and measured the quality of transmitted MPEG-4 encoded video over commercially available CDMA2000 1xEV-DO networks, transmitting different IP packet sizes and RM positions. we propose an error resilient transmission methods for emergency telemedicine over real 3G network.