• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.1
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• pp.156-166
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• 2010

Development of IP network technology creates new service which applying to fusion technologies such as IPTV, VoIP, and so on. Especially, Next generation IP is called IPv6 which will solve the lack of IP. IPv6 is very important on IPTV which needs best quality of service about security, QoS, and bandwidth. In this paper, we constructed experimental network, measured PSNR which computes the peak signal-to-reconstructed video measurement in experimental network environment, analyzed PSNR value. And as a result of measurement, we, Propose optimum network environment for IPTV service provider and transition mechanism designer.

• Information and Communications Magazine
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• v.22 no.6
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• pp.73-82
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• 2005

광섬유를 통하여 가입자 댁내까지 수십- 백Mbps의 QoS(Quality of Service)가 보장된 데이터의 전송이 가능한 FTTH(Fiber To The Home)망에서는 기존의 Best effort망에서 제공이 곤란하였던 고품질 IPTV등 다양한 영상서비스의 제공이 가능하다. 본 고에서는 광대역 영상서비스의 최근동향과 KT가 광주광역시에서 제공하고있는 FTTH시험서비스의 내용에 관하여 기술하였다.

• Proceedings of the KIEE Conference
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• 2005.05a
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• pp.215-217
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• 2005

There are two types of packet loss probabilities used in both the network layer and the physical layer within the wireless transmitter such as a queueing discard probability and transmission loss probability. We analyze these loss performances in order to guarantee Quality of Service (QoS) which is the basic of the future network. The queuing loss probability is caused by a maximum allowable delay time and the transmission loss probability is caused by a wireless channel error. These two types of packet loss probabilities are not easily analyzed due to recursive feedback which, originates as a result at a queueing delay and a number of retransmission attempts. We consider a wireless transmitter to a M/D/1 queueing model. We configurate the model to have a finite-size FIFO buffer in order to analyze the real-time traffic streams. Then we present the approaches used for evaluating the loss probabilities of this M/D/1/K queueing model. To analyze the two types of probabilities which have mutual feedbacks with each other, we drive the solutions recursively. The validity and accuracy of the analysis are confirmed by the computer simulation. From the following solutions, we suggest a minimum of 'a Maximum Allowable Delay Time' for real-time traffic in order to initially guarantee the QoS. Finally, we analyze the required service rate for each type utilizing real-time traffic and we apply our valuable analysis to a N-user's wireless network in order to get the fundamental information (types of supportable real-type traffics, types of supportable QoS, supportable maximum number of users) for network design.

• Industrial Engineering and Management Systems
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• v.9 no.3
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• pp.285-293
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• 2010

To obtain visible and traceable information from the supply chain, HW/SW standards for the EPC global network, which process electronic product code (EPC) data read from Radio frequency identification (RFID) tags, are regarded as the de facto industry standard. Supply chain participants install information service systems and provide logistics information to partners by following the EPCglobal architecture framework. Although quality of service (QoS) is essential for providing dependable and scalable services as pointed out by Auto-ID Lab, only a few models for the performance analysis of QoS-related work have been developed in the context of EPC information service systems. Specifically, doing so allows alternative design choices to be tested in an easy and cost-effective manner and can highlight potential performance problems in designs long before any construction costs are incurred. Thus, in this study we construct a model of an EPC information service system for the purposes of performance analysis and designing a dependable system. We also develop a set of building blocks for analytical performance models. To illustrate how the model works, we determine the characteristics of an EPC information service system and then select a combination of these proven modeling concepts. We construct a performance model that considers the response time and shows how to derive meaningful performance values. Finally, we compare the analytical results to measurements of the EPC information service system.

• Journal of Korea Multimedia Society
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• v.10 no.5
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• pp.612-622
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• 2007

As compared with wired network environment, mobile VOD service in wireless network environment is difficult to guarantee the quality of VOD service due to the limitation of the system resources. Moreover, the wireless network has small bandwidth resources and short access range so that has limitations for providing the service to a large number of mobile clients. The limited battery capacity of a mobile device becomes an obstacle to mobile clients for enjoying VOD service for a long time. Therefore, these drawbacks should be considered in developing mobile VOD applications. This paper proposes a new technique for improving QoS of mobile VOD service by jointly use of broadcasting technique, which can provide VOD service to a large number of mobile clients simultaneously, and bursty communication technique, which can reduce the power consumption of a mobile device. With having these two techniques, streaming data are divided into segments, each of which is grouped into a particular packet and then transmitted at a time in the busty communication manner. In the performance evaluation, the power consumption on WNIC of a mobile device can be reduced by over 80%, and the initial service delay can be reduced in proportion to the number of channels compared with the existing staggered broadcasting technique.

• The KIPS Transactions:PartC
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• v.9C no.4
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• pp.505-512
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• 2002

Rerouting mechanisms must be used by connections in order to provide QoS (Quality of service) characterization of services, which provides mean for reliable and efficient transfer of services under fault generating network. Also, user's services can classily according to their QoS characterizations. In this paper, we study classification of user services according to their characterization for providing differentiated services, and propose rerouting mechanisms under fault generating network. For this, we study various rerouting mechanisms including rerouting locus of start (Source Rerouting, Link Rerouting), rerouting timing of start (Immediate Rerouting, Random Rerouting) and their orthogonal relationship, eventually we propose new rerouting mechanisms such as DRIT, DRDT which show higher performance according to priority of services than others. Our simulation shows that rerouting mechanism (DRDT), applied differentiated mechanisms is better performance to provide differentiated service.

• The Transactions of the Korea Information Processing Society
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• v.5 no.12
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• pp.3047-3062
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• 1998

This paper deseribes the desihn and the implementation of the Realtime Transport Protocol(RTP)/ Rdaltime Control Protocol(RTCP) (RFC 1889,1890) that is used to transmit the audio/video data to any destination and to feedback the Quality of Service (QoS) information of the received media data to the sender, in the teleconferencing systems proposed by ITU-T. These protocols are implemented with multi thead technique and run on top of UDP/IP-Multicast through the socket interface as the underlying protocol. The upper layer is impelmented such that in can be accessed by the H245 comference control protocol. The RTP packetizes the digitized audio/video data from the encoder info a fixed format, and multieast to the participants. The RTCP monitors RTP packets and extracts the QoS values from it such as round-trip delay, jiter and packet loss to form RTCP packets and non periokically sends them to the sender site. In this Paper, we also descritx the study of measurement and analysis for QoS factors that observed on performing teleconferencing system over Internet. The results from this experiment is indicate that RTT and Jitter value are acceptable even entwork load is high. However, it appears that packet loss rate is high in daytime and most losses periods have length one or two.

• Journal of IKEEE
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• v.17 no.2
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• pp.96-103
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• 2013

Since multimedia data takes large part of realtime transmission in wireless communication environments such as IEEE 802.11, QoS issues became one of the important problems with network performance. 802.11e MAC provides differentiated services based on priority schemes to solve existing 802.11 MAC problems. The TXOP is an important factor with the priority to improve network performance and QoS because it defines the time duration in which multiple frames can be transferred at one time for each station. In this paper, therefore frame sizes, TXOP Limit, and Priority values in accordance with the number of stations are experimented and derived for best network performance and QoS. Using 802.11e standard parameters, simulation results show the best throughput when the number of stations is 5 and TXOP Limit value is 6.016ms. For fairness, the best result is achieved at 3.008ms of TXOP Limit value and 15-31 of CW(Contention Window) that is lower priority than CW 7-15.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.1
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• pp.79-95
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• 2016

In order to improve the Quality-of-Service (QoS) of latency sensitive applications in next-generation cellular networks, multi-path is adopted to transmit packet stream in real-time to achieve high-quality video transmission in heterogeneous wireless networks. However, multi-path also introduces two important challenges: out-of-order issue and reordering delay. In this paper, we propose a new architecture based on Software Defined Network (SDN) for flow aggregation and flow splitting, and then design a Multiple Access Radio Scheduling (MARS) scheme based on relative Round-Trip Time (RTT) measurement. The QoS metrics including end-to-end delay, throughput and the packet out-of-order problem at the receiver have been investigated using the extensive simulation experiments. The performance results show that this SDN architecture coupled with the proposed MARS scheme can reduce the end-to-end delay and the reordering delay time caused by packet out-of-order as well as achieve a better throughput than the existing SMOS and Round-Robin algorithms.

• The KIPS Transactions:PartC
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• v.9C no.5
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• pp.733-742
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• 2002

It has been known that the host mobility feature has very significant impact on the QoS (Quality of Service), which is usually required to a real-time multimedia application. The existing QoS support mechanisms to provide the real-time services to fixed network environment, like as RSVP, are inadequate to accommodate the mobile hosts which can frequently change their point of attachments to the fixed network. So, MRSVP (Mobile RSVP) protocol has been proposed to reduce the impacts of host mobility on QoS guarantees, in which a mobile host needs to make advance resource reservations at multiple locations it may possibly visit during the lifetime of the connection. This paper proposes a dynamic dual anchor node (DDAN) architecture which integrates the MRSVP and RSVP tunnel, in addition to the Mobile IP Regional Registration protocol. By limiting the resource reserved in local area, it preserves the lower level of resource reservation, but provides approximately the same degree of QoS support as the existing MRSVP.