• Journal of Broadcast Engineering
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• v.23 no.5
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• pp.682-692
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• 2018

In this paper, we propose an ARQ packet error control scheme using multiple threads in delivering massive capacity of multimedia based on MMT(MPEG Media Transport) protocol. On the sending side, each frame that constitutes an image is packetized into MMT packets based on MMT protocol. The header of the packet stores the sequence number of the frames contained in the packet and the time of presentation information. The payload of the packet stores the direct information that comprises the frame. The generated MMT packet is transmitted to the IP network. The receiving side checks if any error has occurred in the received packet. For any identified error, it controls the error through ARQ error control scheme and reconfigure the frame according to the information stored in the header of the received packet. At this point, a multi-threading based transport design is constructed so that each thread takes over a single frame, which increases the transmission efficiency of massive capacity multimedia. The efficiency of the multi-threading transport method is verified by solving the problems that might arise when using a single-thread approach if packets with errors are retransmitted.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.4B
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• pp.368-376
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• 2009

Over recent years, the development of multimedia devices has meant that a wider multimedia streaming service can be supported, and there are now many ways in which TV channels can communicate with different terminals. Generally, scalable video streaming is known to provide more efficient channel capacity than simulcast video streaming. Simulcast video streaming requires a large network bandwidth for all resolutions, but scalable video streaming needs only one flow for all resolutions. On the contrary, to preserve the same video quality, SVC(Sealable Video Coding) needs a higher bit-rate than AVC(non-layered Video Coding) due to the coding penalty($10%{\sim}30%$). In previous research, scalable video streaming has been compared with simulcast video streaming for network channel capacity, in two-user simulation environments. The simulation results show that the channel capacity of SVC is $16{\sim}20%$ smaller than AVC, but scalable video streaming is not efficient because of the limit of the present network framework. In this paper, we propose a new network framework with a new router using EDE(Extraction Decision Engine) and SVC Extractor to improve network performance. In addition, we compare the SVC environment in the proposed framework with previous research on the same way subject. The proposed network framework shows a channel capacity 50%(maximum) lower than that found in previous research studies.

• The KIPS Transactions:PartC
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• v.15C no.5
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• pp.391-398
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• 2008

To provide IPTV service over mobile networks, the mechanism that reduce packet loss and interrupt of multimedia service during the handover should be supported. Especially, buffering based mechanism is preferable for supporting IPTV services in the way of preserving streaming service using stored data and recovering non-received data after handover. But previous research doesn't consider the buffer allocation for applying various environments which can change handover time or end to end delay of relay node. This paper propose DBAHAJ mechanism that optimize buffer size of mobile nodes and relay node for supporting seamless IPTV service over mobile environments. Mobile node determines buffer size by checking handover time and maximum difference of sequence to keep playing video data. And multicast agent recovers packet loss during the handover by sending buffered data. By these two procedure, node supports seamless IPTV service on mobile networks. We confirm performance of this mechanism on NS-2 simulator.

• The Journal of the Korea Contents Association
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• v.9 no.12
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• pp.528-536
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• 2009

Recently, the importance of IPTV providing multimedia service over IP based networks managed to provide the required level of quality of service and experience, security, interactivity and reliability has been gradually increasing by the commercialization of its service. However, the original purposes of IPTV service - contents are produced and consumed anytime, anywhere, and over any device at the same time - have not been satisfied by service providers because most services offered by service providers depend on closed IPTV platform. In this paper, we propose an open IPTV service platform that an individual or small company can easily produce contents and offer IPTV service independently from the existing closed IPTV platform.. The proposed IPTV platform exploits overlay multicast scheme to offer reasonable service under the environment where network bandwidth and processor resource are limited. It also uses CDN-like scheme to ensure the quality of transmitted contents. The performance evaluation shows that the proposed IPTV platform has the possibility of being an open IPTV platform considering its results such as the quality of transmitted contents, the transmission rate, the channel zapping time, and so on. It also shows that proposed IPTV platform could be applied to real IPTV service with continuous enhancement of its functions and user interfaces.

• Journal of KIISE:Computing Practices and Letters
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• v.15 no.2
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• pp.114-124
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• 2009

In this paper, we propose a practical TCP friendly rate control scheme that considers the minimum channel bandwidth of the network when transporting SVC (scalable video coding) video over IP netowrks such as Internet. RTP and RTCP is mainly designed for use with UDP (User Datagram Protocol) for real-time video transport over the Internet. TCP-friendly rate control was proposed to satisfy the demands of multimedia applications while being reasonably fair when competing for bandwidth with conventional TCP applications. However the rate control model of the conventional TCP-friendly rate control scheme does not consider the minimum channel bandwidth of the network. Thus the estimated channel bandwidth by the conventional rate control model might be quite different from the real channel bandwidth when the packet loss ratio of the network is very large. In this paper, we propose a modified TCP-friendly rate control scheme that considers the minimum channel bandwidth of the network. Based on the modified TCP-friendly rate control, we assign the minimum channel bandwidth to the base layer bitstream of SVC video, and remaining available bandwidth is allocated to the enhancement layer of SVC video for the TCP friendly scalable video transmission. It is shown by simulations that the modified TCP-friendly rate control scheme can be effectively used for a wider range of controlled bit rates depending on the packet loss ratio than the conventional TCP-friendly control scheme. Furthermore, the effectiveness of the proposed scheme in terms of objective video quality is proved by comparing PSNR performance with the conventional scheme.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.3
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• pp.997-1005
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• 2010

Recently, there has been a growing interest in the use of WLAN technology due to its easy deployment, flexibility and so on. Examples of WLAN applications range from standard internet services such as Web access to real-time services with strict latency/throughput requirements such as multimedia video and voice over IP on wireless network environments. Fair and efficient distribution of the traffic loads among APs(Access Points) has become an important issue for improved utilization of WLAN. This paper focuses on an AP selection scheme for achieving better load balance, and hence increasing network resource utilization for each user on wireless network environments. This scheme makes use of active scan patterns and the network delay as main parameters of load measurement and AP selection. This scheme attempts to estimate the AP traffic loads by observing the up/down delay and utilize the results to maximize the link resource efficiency through load balancing. We compared the proposed scheme with the original SNR(Signal to Noise Ratio)-based scheme using the NS-2(Network Simulation.2). We found that the proposed scheme improves the throughput by 12.5% and lower the network up/down link delay by 36.84% and 60.42%, respectively. All in all, the new scheme can significantly increase overall network throughput and reduce up/down delay while providing excellent quality for voice and video services.

• The Journal of the Korea institute of electronic communication sciences
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• v.8 no.1
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• pp.85-91
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• 2013

Recently, the weight of high-quality multimedia contents is explosively growing out of the network total traffic. This steep increase of media contents to require huge traffic does not contribute to the generation of revenue streams and it leads to the situation of dumb pipe to trouble network providers into the big burden of investment on the network expansion. Accordingly, the transfer to the smart network to enable the effective delivery of large-scale media is imminently challenging issue to the network providers to seek the profitable business. The smart network revolves around the technologies to enhance end-to-end quality and fair usage with network resources and to optimize the traffic for the contents delivery over the concept of Content-Centric Network. In this paper, we propose an architecture and fundamental functions suitable for the smart network and suggest improved test results through the construction of an experimental network.

• The Transactions of the Korea Information Processing Society
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• v.7 no.11S
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• pp.3651-3667
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• 2000

Recently, in the distributed multimedia environments based on internet, as radical growing technologies, the most of researchers focus on both streaming technology and distributed object thchnology, Specially, the studies which are tried to integrate the streaming services on the distributed object technology have been progressing. These technologies are applied to various stream service mamgements and protocols. However, the stream service management mexlels which are being proposed by the existing researches are insufficient for suporting the QoS of stream services. Besides, the existing models have the problems that cannot support the extensibility and the reusability, when the QoS-reiatedfunctions are being developed as a sub-module which is suited on the specific-purpose application services. For solving these problems, in this paper. we suggested a QoS Integrated platform which can extend and reuse using the distributed object technologies, and guarantee the QoS of the stream services. A structure of platform we suggested consists of three components such as User Control Module(UCM), QoS Management Module(QoSM) and Stream Object. Stream Object has Send/Receive operations for transmitting the RTP packets over TCP/IP. User Control ModuleI(UCM) controls Stream Objects via the COREA service objects. QoS Management Modulel(QoSM) has the functions which maintain the QoS of stream service between the UCMs in client and server. As QoS control methexlologies, procedures of resource monitoring, negotiation, and resource adaptation are executed via the interactions among these comiXments mentioned above. For constmcting this QoS integrated platform, we first implemented the modules mentioned above independently, and then, used IDL for defining interfaces among these mexlules so that can support platform independence, interoperability and portability base on COREA. This platform is constructed using OrbixWeb 3.1c following CORBA specification on Solaris 2.5/2.7, Java language, Java, Java Media Framework API 2.0, Mini-SQL1.0.16 and multimedia equipments. As results for verifying this platform functionally, we showed executing results of each module we mentioned above, and a numerical data obtained from QoS control procedures on client and server's GUI, while stream service is executing on our platform.

• Journal of Broadcast Engineering
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• v.19 no.6
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• pp.767-788
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• 2014

The worldwide transition from analog to digital broadcasting has now been completed and the need to study next generation standards for Ultra High Definition TV (UHDTV) broadcasting, and broadcasting & communication convergence systems is rapidly growing. In particular, high resolution mobile broadcasting services are needed to satisfy recent consumers. Therefore, the development of highly efficient convergence broadcasting systems that provide fixed/mobile broadcasting through a single channel is needed. In this paper, a service scenario and requirements for providing 4K UHD & HD convergence broadcasting services through a terrestrial single channel are analyzed by employing the latest transmission and A/V codec technologies. Optimized transmission parameters for 6 MHz & 8 MHz terrestrial bandwidths are drawn, and receiving performances are measured under Additive White Gaussian Noise (AWGN) and time-varying multipath channels. From the results, in a 6 MHz bandwidth, the reliable receiving of HD layer data can be achieved when the receiver velocity is maximum 140 Km/h and is not achieved when the velocity is over 140 Km/h due to the limit of bandwidth. When the bandwidth is extended to 8 MHz, the reliable receiving of both 4K UHD and HD layer data is achieved under a very fast fading multipath channel.