• Journal of Digital Contents Society
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• v.14 no.1
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• pp.25-33
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• 2013

P2P(Peer-to-Peer) architecture can reduce the network bandwidth and resource on the server since peers exchange data chunks with each other, while server-client architecture causes a lot of traffic on the server. Peers receive a data more reliably when the number of participating peer increases. Currently, P2P traffic has accounted for about 65% of the world's Internet traffic and diverse P2P streaming services have launched combining to video streaming technology. However, the requirements and data chunk delivery algorithms for mobile P2P streaming service should be investigated, since the existing P2P technologies have been developed and designed for the wired network. In particular, the bandwidth fluctuation caused by user mobility, wireless packet collisions, and packet losses brings about different problems on the mobile P2P streaming service compared to existing P2P streaming service. In this paper, we analyzed the problem of mobile P2P streaming services in the 802.11n wireless LAN environment through experiments.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.4
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• pp.51-56
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• 2014

Thesedays, there are many smart device applications developed, especially on the using various sensors included in the smart device. Smart devices have several sensors which are camera, GPS, mike, and communication module for collecting ubiquitous environment, and many applications are developed by using such sensors. In this paper, we developed the multimedia stream architecture and examined the smart device applications based on open source with front and back-end server clouds for developing the conceptual architecture. Also, we examined the back-end distributed servers, realtime multimedia stream transferring, multi-media store, and media relay for other server and smart devices. We test the examined architecture on the real target environment to collect the SIP initial setup time, media stream delay, and end-to-end play time. The test results show that there have good network operation environment to provide realtime multimedia services, and we need to improve the end-to-end play time by minimizing the initial setup time.

• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.4
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• pp.379-384
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• 2010

The service quality is one of the most important issue for successful deployment of IPTV services. Currently, a number of parameters have been reported that they have an effect on the quality of IPTV services. However, it is almost impossible to monitor and manage all the parameters in real time. Key parameters, therefore, should be chosen as a service metric for efficient quality management of IPTV services. For this purpose, we implement an environment for monitoring and measuring the quality of IPTV services and try to figure out the effect of each parameter on the quality of IPTV services. In particular, we focus on a parameter called continuous count error (CCE) measured at the level of MPEG transport stream (TS) and show that it has a greater effect on the quality of IPTV services than other parameters.

• Proceedings of the Korean Institute of Navigation and Port Research Conference
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• 2004.04a
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• pp.43-48
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• 2004

RTP that is proposed supplement of real-time services on internet environment, as Real-time Transport Protocol, is the protocol that for the purpose of sending data of stream type. RTP and RTCP(Real-time Transport Control Protocol) basically work at the same time, RTCP serves with state information of network at present. RTP has important properties of a transport protocol that runs on end-to-end systems and provides demultiplexing. It also offer reliability and protocol-defined flow／congestion control that transport protocol like TCP can not provides. In this paper, we look around concept and construction of Differentiated sen1ice tint run on RTP and by setting parameters of packet transfer method be used CBQ(Class-Based Queuing) for packet transfer on Differentiated service, each service queue controls properly through packet scheduling method, such as WRR(Weighted Round Robin) and PRR(Packet-by-packet Round Robin) all service classes do not experience the starvation and confirm the performance through computer simulation to achieve fairly scheduling.

• The KIPS Transactions:PartC
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• v.13C no.6 s.109
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• pp.669-676
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• 2006

In this paper, we present a new framework and stream synchronization mechanism to effectively support developing real-time multimedia streaming services by using a real-time object model named TMO (Time-triggered Message-triggered Object). The purpose of the framework is twofold. Firstly, the framework helps developers to design complex distributed real-time multimedia streaming services. Secondly, it supports timely streaming facilities. In order to achieve these goals, our framework is consist of Multimedia Streaming TMO, MMStreaming TMO Support Library and TMO Support Middleware. The time-triggered spontaneous feature of the MMStream TMO and a global-time based synchronization scheme is used as a regulator against the irregular deliveries and processing of media units caused by QoS non-guaranteed systems and communication channels. In conclusion, timely service capability of our framework is expected to contributed to overall enhancement of the real-time multimedia streaming.

• Journal of the Korea Society of Computer and Information
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• v.15 no.11
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• pp.135-142
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• 2010

QoS in the proxy system are under heavy influence from interferences such as congestion, latency, and retransmission. Also, multi-level streaming services affects from temporal synchronization, which lead to degrade the service quality. This paper proposes a new segment-based buffer management mechanism which reduces performance degradation of streaming services and enhances throughput of streaming due to drawbacks of the proxy system. The proposed paper optimizes streaming services by: 1) Use of segment-based buffer management mechanism, 2) Minimization of overhead due to congestion and interference, and 3) Minimization of retransmission due to disconnection and delay. This paper utilizes fuzzy value $\mu$ and cost weight $\omega$ to process the result. The simulation result shows that the proposed mechanism has better performance in buffer cache control rate, average packet loss rate, and delay saving rate with stream relevance metric than the other existing methods of fixed segmentation method, pyramid segmentation method, and skyscraper segmentation method.

• Journal of Korea Game Society
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• v.12 no.4
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• pp.13-22
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• 2012

The wide-spread use of the broadband Internet service makes the cloud computing-based gaming service possible. A game program is executed on a cloud node and its live image is fed into a remote user's display device via video streaming. The user's input is immediately transmitted and applied to the game. The minimization of the time to process remote user's input and transmit the live image back to the user and thus satisfying the requirement of instant responsiveness for gaming makes it possible. However, the cost to build its servers can be very expensive to provide high quality 3D games because a general purpose graphics system that cloud nodes are likely to have for the service supports a single 3D application at a time. Thus, the server must have a technology of 'realtime multiple rendering' to execute multiple 3D games simultaneously. This paper proposes a new architecture of 2-tier servers of clouds nodes of which one group executes multiple games and the other produces game's live images. It also performs a few experimentations to prove the feasibility of the new architecture.

• Journal of Digital Convergence
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• v.9 no.5
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• pp.225-233
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• 2011

Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. Recently, the introduction of smart phones has led to a growth in social networking services and thus, the research and development of Internet telephony has been actively progressed and has the potential to become a replacement for the telephone service that is currently being used. In this paper we designed and implemented a recording system which records voice data of SIP-based Internet telephone's voice calls. It is developed on the linux system and has some features such as audio mixing of two in/out voice channels, live packet sniffing, and the ability to transfer mixed audio files to the log file server. These functions are implemented using various open source softwares. Afterwards, this VoIP recording system will be applied as a base technology to advanced services like a VoIP-based call center system.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.48 no.6
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• pp.35-47
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• 2011

A present experimental broadcasting is service compatible to delivery left view and right view of a dual stream for backward compatible with 2D device. It has a limitation of quality for a transmission of dual stream in terrestrial bandwidth 19 Mbps. This paper proposes adjunct terrestrial-3D broadcasting system which can provide a stable quality beyond a limitation of quality from fixed bandwidth. It proposes the system that composes a high quality stereoscopic video through synchronizing Real time broadcasting and Non Real Time broadcasting Regardless of the terrestrial bandwidth. So this paper tries to analyze each technique of Real-Time broadcasting and NRT broadcasting, and verify this proposal through the technique of an algorithm that we can implement NRT adjunct high quality 3DTV system and the experiments.

• Journal of KIISE:Computer Systems and Theory
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• v.28 no.12
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• pp.660-673
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• 2001

In Video-on-Demand (VoD) servers, disk throughput is an important system design parameter because it is directly related to the number of user requests that can be served simultaneously. In this paper, we propose an efficient periodic request grouping scheme for disk array-based VoD servers that reduces the disk seek time, thus improving the disk throughput of VoD disk arrays. To reduce the disk seek time, the proposed scheme groups the periodic requests that access data blocks stored in adjacent regions into one, and arranges these groups in a pre-determined order (e.g., in left-symmetric or right-symmetric fashion). Our simulation result shows that the proposed scheme reduces the average disk bandwidth required by a single video stream and can serve more user requests than existing schemes. For a data block size of 192KB, the number of simultaneously served user requests is increased by 8% while the average waiting time for a user request is decreased by 20%. We also propose an adaptation technique that conforms the proposed scheme to the user preference changes for video streams.