• Journal of the Korea Institute of Information and Communication Engineering
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• v.7 no.2
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• pp.228-238
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• 2003

We believe that existing wire solutions such as HomePNA2.0 using phone lines and HomePlug using power line and wireless solution such as HomeRF are the most promising solutions, because of its cost-effectiveness. However, MAC protocols of these solutions provide only Class of Service(CoS) using priority mechanism like HomePNA and HomePlug or consider only voice among real-time traffics like HomeRF. For these reasons, we perceive the needs of the new MAC protocol which is no new wire solution and provides guaranteed Quality of Service (QoS) for not only voice but also video and audio. In light of this, we present the design and software implementation of a new MAC protocol for Home Networks. Our evaluation results of software implementation verify that proposed MAC protocol can provide low delay, low loss, and low jitter to real-time traffic by reserving bandwidth.

• Journal of Communications and Networks
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• v.7 no.2
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• pp.222-230
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• 2005

This article presents a field evaluation of an IP-based architecture for heterogeneous environments that has been developed under the aegis of the Moby Dick project, covering UMTS-like (universal mobile telecommunications system) TD-CDMA (time division-code division multiple access) wireless access technology, wireless and wired LANs. The architecture treats all transmission capabilities as basic physical and data-link layers, and replaces all higher-level tasks by IP-based strategies. The Moby Dick architecture incorporates mobile IPv6, fast handovers, AAA-control (authentication, authorisation, accounting), charging and quality of service (QoS) in an integrated framework. The architecture further allows for optimised control on the radio link layer resources. It has been implemented and tested by expert users, and evaluated by real users on field trials with multiple services available.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.2 no.2
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• pp.63-81
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• 2008

WiMAX has been introduced as a competitive alternative for metropolitan broadband wireless access technologies. It is connection oriented and it can provide very high data rates, large service coverage, and flexible quality of services (QoS). Due to the large number of connections and flexible QoS supported by WiMAX, the uplink access in WiMAX networks is very challenging since the medium access control (MAC) protocol must efficiently manage the bandwidth and related channel allocations. In this paper, we propose and investigate a cost-effective WiMAX bandwidth management scheme, named the WiMAX partial sharing scheme (WPSS), in order to provide good QoS while achieving better bandwidth utilization and network throughput. The proposed bandwidth management scheme is compared with a simple but inefficient scheme, named the WiMAX complete sharing scheme (WCPS). A maximum entropy (ME) based analytical model (MEAM) is proposed for the performance evaluation of the two bandwidth management schemes. The reason for using MEAM for the performance evaluation is that MEAM can efficiently model a large-scale system in which the number of stations or connections is generally very high, while the traditional simulation and analytical (e.g., Markov models) approaches cannot perform well due to the high computation complexity. We model the bandwidth management scheme as a queuing network model (QNM) that consists of interacting multiclass queues for different service classes. Closed form expressions for the state and blocking probability distributions are derived for those schemes. Simulation results verify the MEAM numerical results and show that WPSS can significantly improve the network’s performance compared to WCPS.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.9B
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• pp.900-907
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• 2009

With the introduction of the concept such as QoS, QoE, and MOS for the multimedia services over the high-speed network the user's concern about the degradation in the perceived quality of service for the conventional data service has become a new problem to the Internet service providers. In this work we propose a method to evaluate the subjective QoE such as MOS for the high-speed Internet service, where the round trip time for the data transfer is taken into account as a performance parameter for the experienced delay. We develop a tool to evaluate an MOS that enables us to investigate the behavior of users for diverse delay times for the web access service, via which we analyze the sensitivity of users with regard to MOS as a function of delay.

• Annual Conference of KIPS
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• 2008.11a
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• pp.917-919
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• 2008

WDM 네트워크에서 QoS를 향상시키기 위해서는 가상 토폴로지를 효율적으로 재구성할 수 있어야 한다. 본 연구에서는 토폴로지를 재구성할 시점을 결정하고 적절한 파장을 할당하는 기법을 제안함으로써 효과적으로 WDM 네트워크를 운영할 수 있는 WDM 컨트롤러를 제안한다.

• Journal of the Korea Society of Computer and Information
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• v.8 no.4
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• pp.47-56
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• 2003

According to rapidly developed mobile terminals and network in the play environment for multimedia presentation, different end-to-end QoS situations appear. Then the generation of transcoding path algorithm was reviewed to transcode a source'data satisfying the needed QoS from a destination and play it considering given transcoders and network. This method used only workload as a parameter although two parameters, workload and throughput, were needed to process multimedia stream in a transcoder, Therefore generated transcoding path with this method had additional calculation to check playability whether it was safisfed the QoS of a destination or not. To solve the problem this paper suggests T algorithm with evaluation function using isochronous property that is needed for multimedia stream to arrive a destination. That means, most playable path is selected with heuristic search based isochronous property between many transcoding paths. Using the suggested algorithm, a transcoding path can be generated faster to play the multimedia data with different end-to-end QoS in real-time transmission.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.1
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• pp.9-19
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• 1999

An increase of the number of the real time applications and the Internet hosts make the Internet architecture changed. Current Internet architecture has some problems to process the real time traffics. To solve these problem, a new Internet architecture is proposed as the Integrated Service model. In the current Internet, as multicast protocols, the QoS multicast and the best-effort multicast have been studied in their separate network environments. But, the Integrated Service Packet Network is a heterogeneous network composed of the QoS delivery domain and the best-effort delivery domain. Thus, those separated multicast protocols have limitations in an ISPN. In this paper, we propose a multicast protocol for the ISPN with the QoS and the best-effort multicasting, and analyze the performance of this protocol. As a result, we find that the packet losses are same for hybrid multicast and best-effort multicast when the bandwidth is sufficient. But, if there exist some background traffics, the hybrid multicast has less packet loss than of the best-effort multicast.

• Journal of the Korean Institute of Intelligent Systems
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• v.18 no.1
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• pp.78-85
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• 2008

In this paper, we propose the extension of cooperation protocol which realizes the autonomous control of QoS and show its effectiveness with implementation and experiments of our agent-based videoconference system. The autonomous QoS control in the agent-based videoconference system means the framework of adjusting its functions/abilities by itself to cope with the changes of user requirements and system resource situations. But, the original cooperation protocol used by the agent-based videoconference system is not enough to perform cooperation closely to use the limited resources effectively. In our proposal, the cooperation protocol is offended in the following points: (1) introducing a cooperation states, (2) extension of performatives, (3) introducing compromise levels. We embedded the proposed protocol to the agent-based videoconference system and experimented the behavior of the system. According to the experiments on this prototype system, we confirmed that the flexibility of the system is improved.

• Journal of KIISE:Software and Applications
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• v.36 no.1
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• pp.1-9
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• 2009

In heterogeneous and distributed computing environments, with an increasing number of Web services providing similar functionalities, the reliability of Web services is a critical decision factor. To fulfill the open business model such as cooperation among enterprises, several Web services can be composed into the upper level business transaction. In Web services composition, the reliability of services is more and more critical. Though each unit Web service can be reliable, the reliability of the composed service is not guaranteed. Thus a way to efficiently assess and select composed Web services is needed. In this paper, we define new metrics for measuring the trust value of Web services, and propose an evaluation method to predict the trustworthy degree of the composed services based on the metrics. We also define a conceptual framework to support optimal Web services selection based on the proposed trust evaluation method. By selecting using the quantitative measurement rather than intuitive selection of the service user, it allows the service users to select the high reliable service meeting their quality requirements well.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.71-77
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• 2011

In this paper, we propose a conversational quality measurement (CQM) system for providing the objective QoS of high quality mobile VoIP voice telecommunication. For measuring the conversational quality, the VoIP telecommunication system is implemented in two smart phones connected with VoIP. The VoIP telecommunication system consists of echo cancellation, noise reduction, speech encoding/decoding, packet generation with RTP (Real-Time Protocol), jitter buffer control and POS (Play-out Schedule) with LC (loss Concealment). The CQM system is connected to a microphone and a speaker of each smart phone. The voice signal of each speaker is recorded and used to measure CE (Conversational Efficiency), CS (Conversational Symmetry), PESQ (Perceptual Evaluation of Speech Quality) and CE-CS-PESQ correlation. We prove the CQM system by measuring CE, CS and PESQ under various SNR, delay and loss due to IP network environment.