• The KIPS Transactions:PartD
• /
• v.14D no.3 s.113
• /
• pp.319-328
• /
• 2007

As the coverage of Web services become wider and the number of implementation cases is growing, the importance of applying the Web services quality model to real world is increased. For maintaining the level of Web services qualify, it should be required to study on assurance method of Web services qualify level. Assurance for Web services, which is newly proposed by OASIS TC, means the totality of activities for managing the quality level of them. For managing Web service quality, Web service associates could usually use SLA(Service Level Agreement) method in which a service consumer contracts for some service level with a service provider and gives for penalty or pays incentives according to the result of evaluation of services. But, there are some difficulties in applying SLA to Web services, because Web services have publicity, multiple users, and 3rd party for management. So, we need a new assurance method for Web service by considering the characteristics of Web services. This paper provides the new concept of committed assurance level for Web services. This concept can be defined as the set of maximum level of quality expected by each user, which provide the consistent view of Web service quality. This paper presents the method for duality associates to preserve some quality level of Web service by using this concept.

• 한국정보통신설비학회:학술대회논문집
• /
• 2007.08a
• /
• pp.92-99
• /
• 2007

현재 통신 시장은 보다 편리하고 다양한 서비스를 찾는 소비자의 욕구와 맞물려 TPS(Triple Play Service) 방향으로 급속히 나아가고 있다. 인터넷 전화의 보급은 이미 보편적 서비스가 된 초고속인터넷 서비스와 연계하여 빠르게 확산되고 있으며, 2007년 들어 방송서비스까지 결합한 TPS에 대한 통신사업자간 경쟁이 본격화되고 있는 추세이다. 트리플 플레이 서비스(Triple Play Service;이하 TPS)는 초고속 인터넷과 인터넷 전화와 방송 서비스를 하나의 회선으로 제공하는 종합 인터넷 서비스로, 광대역 통합망(BcN; Broadband Convergence Network)의 핵심 서비스이다. 액세스 스위치(Access Switch) 단은 TPS로 인해 더욱 많은 역할을 담당할 것으로 전망되는데, 디지털 방송을 인터넷 프로토콜을 통해 전송하기 위한 멀티캐스트 기능과 혼재된 이종간의 디지털 신호를 각 서비스의 품질에 맞게 보장해 주는 QoS(Quality of Service) 기능이 무엇보다 중요하게 고려 되어야 한다. 본 논문에서는 TPS의 핵심일 수 있는 방송 서비스를 위해 액세스 스위치 단에서 제공해야 할 멀티캐스트와 QoS 기능 구현 방안에 대해 제시한다

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 1999.11a
• /
• pp.401-405
• /
• 1999

본 논문은 실시간성 서비스인 버스트 트래픽을 갖는 ATM(Asynchronous Transfer Mode)망에서 실시간 연결 수락 제어를 위한 종단간의 지연보장을 알고리즘을 이용하여 실시간 서비스 트래픽 서비스 품질의 성능에 대해 분석하였다. 셀 손실 확률은 QoS(Quality of Service)의 기준이 되므로 제안된 실시간 알고리즘은 일정한 버퍼 메모리 크기와 연결 요청에 대한 수락 여부 즉, 연결 수락 여부의 결정을 위한 하나의 큐잉 서비스를 필요로 한다. 이러한 트래픽 소스 각 타입에 대한 초과된 셀에 대한 셀 손실 확률은 트래픽 소스 타입 전체에 하나만 존재하더라도 요구를 충족시킬 수 없으므로, 본 논문에서 제안된 버스트 트래픽 특성에 따라 지연보장 알고리즘을 이용한 큐 서비스 방식은 기존의 종단간 지연 보장 알고리즘들에서 사용하던 씬 정렬을 이용하지 않고 몇 개의 FIFO 큐로 구성된 라운드로빈 큐를 이용하여 복잡성을 크게 감소시켰으며 실시간 서비스와 비실시간 서비스의 서비스방식을 효과적으로 결합하여 다양한 QoS를 보장해야 하는 ATM망에 효과적으로 적용할 수 있고 제안된 알고리즘에 산출되는 각각의 셀 손실 확률에 충분한 상한선을 제공하는 전형적인 매개변수를 사용하여 성능을 분석한다.

• The Journal of the Korea Contents Association
• /
• v.5 no.4
• /
• pp.245-253
• /
• 2005

The multimedia ad hoc wireless network is quite an attractive issue since it offers a flexible solution to enable delivery of multimedia services to mobile end users without fixed backbone networks. However, with the unique design challenges of ad hoc wireless networks, it is a non-trivial issue to provide bounded delay guarantee, with fair share of resources. In this paper, we proposed distributed implementation of the delay guaranteed fair queueing (DGFQ) scheme in multimedia ad hoc wireless networks. Through the the results of performance evaluation, we can conclude that DGFQ also performs well to control bounded delay in multimedia ad hoc wireless networks.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.18 no.1
• /
• pp.149-154
• /
• 2014

Quality is the value that can be measured by observing the characteristics of the service quantity or quality. QoS is predictable service traffic to a minimum requirements what passed in network. In the course of Smart Medical Information System Development there exist some functional requirements to satisfy quality objectives. The functional smart domains of healthcare information systems consists of Patient Module, a smart sensing and communication domain, RFID Tag Readers and the behavior domain, Homecare Station Domain, Clinical Station. This study is performed on evaluation methodology of u-health service satisfaction quality of each domain. In this paper QoS metrics and the quality of medical information requirements, functional requirements are separated by. Quality parameters consists of six items and the functional requirements and quality requirements 20 details the five items and consist of 20 detailed items. On this study the quality evaluation methodology of Korean smart health information quality assessment matrix 2 - factor evaluation method is proposed. The overall framework of this paper is organizing the specific criteria of quality of medical information system and modeling quality evaluation process under all smart environment.

• Journal of KIISE:Information Networking
• /
• v.27 no.4
• /
• pp.507-521
• /
• 2000

실시간 멀티미디어 서비스에서 Quality of Service(QoS) 보장의 필요성이 증가하고 있다. 멀티미디어 서비스 제공 형태의 대다수가 될 멀티캐스트 경로설정에서도 QoS 보장은 확장성 신뢰성과 함께 매우 중요한 문제이다. QoS 기반 코어 선택 알고리즘을 제안한다. 제안 알고리즈믄 멀티캐스트 경로설정에서 코어 선택시에 다중 QoS 제약조건을 고려한다. QoS 제약조건은 최소보장 대역폭, 종단 지연, 종단 지연변이 등으로 정의한다. 모의 실험결과는 제안한 QCSA와 Maximum Centered Tree(MCT) Average Centered Tree (ACT) Initial Delay-Constrained Shared Tree(Dcinitial) Random Tree(Random)등의 기존 코어 선택 알고리즘의 성능을 각 항목별로 비교한다 멀티캐스트 그룹 멤버수와 QoS 제약조건을 인자로 한 모의 실험 결과는 제안한 QoS 기반 코어 선택 알고리즘이 기존 코어 선택 알고리즘에 비해서 다중 QoS 제약조건 보장 코어 선택 성공률에서 성능 개선 효과를 가짐을 보여준다. 제안 알고리즘이 본 논문에서 설정한 모의 실험 환경에서는 QoS 기반 코어 선택의 정도를 나타내는 성공률에서 약 10% 정도 기존 알고리즘보다 우수함을 보인다. 이 결과는 제안 알고리즘이 코어 선택 과정의 초기부터 멀티캐스트 그룹내의 모든 멤버에 대한 다중 QoS 제약조건을 고려하는 점이 QoS 기반 코어 선택에서 개선 효과를 나타냄을 보여준다.

• Journal of KIISE
• /
• v.41 no.11
• /
• pp.974-981
• /
• 2014

These days, use of multimedia streaming service and demand of QoS (Quality of Service) improvement have been increased because of development of network. QoS of streaming service is influenced by a jitter, delay, throughput, and loss rate. For guaranteeing these factors which are influencing QoS, the role of transport layer is very important. But existing TCP which is a typical transport layer protocol increases the size of congestion window slowly and decreases the size of a congestion window drastically. These TCP characteristic have a problem which cannot guarantee the QoS of UHD multimedia streaming service. In this paper, we propose a router buffer based congestion control method for improving the QoS of UHD streaming services. Our proposed scheme applies congestion window growth rate differentially according to a degree of congestion for preventing an excess of available bandwidth and maintaining a high bandwidth occupied. Also, our proposed scheme can control the size of congestion window according to a change of delay. After comparing with other congestion control protocols in the jitter, throughput, and loss rate, we found that our proposed scheme can offer a service which is suitable for the UDH streaming service.

• Journal of the Institute of Electronics Engineers of Korea SD
• /
• v.44 no.10
• /
• pp.55-60
• /
• 2007

This paper proposes a fast and scalable priority queue architecture for use in high-speed networks which supports quality of service (QoS) guarantees. This architecture is cost-effective since a single queue can generate outputs to multiple out-links. Also, compared with the previous multiple systolic array priority queues, the proposed queue provides fast output generation which is important to high-speed packet schedulers, using a special control block. In addition this architecture provides the feature of high scalability.

• The KIPS Transactions:PartC
• /
• v.17C no.3
• /
• pp.299-306
• /
• 2010

IPTV and VoIP services are considered as killer applications over Mobile WiMAX network, which provideshigh mobility and data rate. Among those which affect the quality of voice in those services, the jitter buffer or playout buffer can compensate the poor voice quality caused by the packet drop due to frequent route change or differences among routes between service endpoints. In this paper, we analyze the correlation between the sleep interval length and jitter buffer size in order to guarantee a predefined level of voice quality. For this purpose, we present an end-to-end delay model considering additional delay incurred by the WiMAX PSC-II sleep mode and a VoIP service quality requirement based on the delay constraints. Through extensive simulation experiments, we also show that the increase of jitter buffer size may degrade the voice quality since it can introduce additional packet drop in the jitter buffer under WiMAX power saving mode.

• Journal of KIISE:Information Networking
• /
• v.30 no.4
• /
• pp.535-544
• /
• 2003

Integrated services networks should guarantee end-to-end delay bounds for real-time applications to provide high quality services. A real-time scheduler is installed on all the output ports to provide such guaranteed service. However, scheduling algorithms studied so far have problems with either network utilization or scalability. Here, network utilization indicates how many real-time sessions can be admitted. In this paper, we propose service curve allocation schemes that result in both high network utilization and scalability in a service curve algorithm. In service curve algorithm, an adopted service curve allocation scheme determines both network utilization and scalability. Contrary to the common belief, we have proved that only a part of a service curve is used to compute deadlines, not the entire curve. From this fact, we propose service curve allocation schemes that result in a constant time for computing deadlines. We through a simulation study that our proposed schemes can achieve better network utilizations than Generalized processor Sharing (GPS) algorithms including the multirate algorithm. To our knowledge, the service curve algorithm adopting our schemes can achieve the widest network utilization among existing scheduling algorithms that have the same scalability.