• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.5
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• pp.146-153
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• 2009

Recently, the research relative to NGN(Next Generation Network) is progressing in 3GPP(The 3rd Generation Partnership Progect), IETF(Internet Engineering Task Force), and so on. Although user needs frequently mobility which is various service pattern, In accordance with the development of these various applications, IMS(IP Multimedia Subsystem) and hierarchical networks ie, Femtocell/WiBro/3G etc is constructed for more user demands which provide service in anytime, anywhere. It is necessary to optimum network selection criterion which consider to wireless signal quality add to user service profile and service network traffic balance. NGN also needs a method to perform heterogeneous handover and to constraint Ping-pong phenomenon when using existing terminal-based handover decision. This paper proposes scheme for network selection and heterogeneous handover procedure in hierarchical wireless multi-access network based on SIP-MIH(Session Initiation Protocol-Media Independent Handover) with IMS by using user service profile that the considerations are dealing with not only selection and registration of various access network but also easy of developing the terminal.

• Journal of Broadcast Engineering
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• v.16 no.1
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• pp.155-173
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• 2011

Recently, with the digitalization of a broadcasting system and the development of a communication technology, the existing trend of consuming media contents throughout the fixed-displayer and the dedicated channel is being changed. The existing user only could consume media contents under limited time and places because of the fixed-displayer and the dedicated channel. However, with advent of the IP-based terminals and HTTP adaptive streaming which transfer the media sequence according to the user's transmission condition, users become possible to enjoy the media content anytime anywhere. As the result of the alteration of the broadcasting surrounding, users can enjoy the media content while changing his terminals according to their preferences and circumstances. However, in case that users try to consume consecutively the content from last view-point ended in the previous terminal under current HTTP adaptive streaming environment, a user has to remember the last view-point, and then has to apply the view-point to the changed terminal. Thus, for solving this problem, this paper defines "User Profile" for describing the metadata for the chained content consume environment between the terminals. Also, for proving the proposed method, this paper try to demonstrate the proposed method throughout the realization of the system based on Smooth Streaming from Microsoft.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.41 no.5
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• pp.37-45
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• 2004

As Internet technologies are mature, many new applications that are different characteristics are emerging. Recently we see wide use of P2P(Peer to Peer) applications of which traffic shows different statistical characteristics compared with traditional application such as web(HTTP) and FTP(File Transfer Protocol). In this paper, we measured subscriber network of KT(Korea Telecom) to analyze P2P traffic characteristics. We show flow characteristics of measured traffic. We also estimate Hurst parameter of P2P traffic and compare self-similarity with web traffic. Analysis results indicate that P2P traffic is much bustier than web traffic and makes both upstream traffic and downstream traffic be symmetric. To predict parameters related QoS such as packet loss and delays we model P2P traffic using two self-similar traffic models and predict both loss probability and mm delay then compare their accuracies. With simulation we show that the self-similar traffic models we derive predict the performance of P2P traffic accurately and thus when we design a network or evaluate its performance, we can use the P2P traffic model as reference input traffic.

• Asia-Pacific Journal of Business Venturing and Entrepreneurship
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• v.5 no.3
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• pp.27-45
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• 2010

It is strongly believed that multicast will become one of the most promising services on internet for the next generation. Multicast service can be deployed either on network-layer or application-layer. IP multicast (network-layer multicast) is implemented by network nodes (i.e., routers) and avoids multiple copies of the same datagram on the same link. Despite the conceptual simplicity of IP multicast and its obvious benefits, it has not been widely deployed since there remain many unresolved issues. As an alternative to IP multicast, overlay multicast (application-layer multicast) implements the multicast functionality at end hosts rather than routers. This may require more overall bandwidth than IP multicast because duplicate packets travel the same physical links multiple times, but it provides an inexpensive, deployable method of providing point-to-multipoint group communication. In this paper we develop an efficient method applied greedy algorithm for solving two models of overlay multicast routing protocol that is aimed to construct MDST (Minimum Diameter Spanning Tree : minimum cost path from a source node to all its receivers) and MST (Minimum Spanning Tree : minimum total cost spanning all the members). We also simulate and analyze MDST and MST.

• Information Systems Review
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• v.4 no.2
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• pp.209-222
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• 2002

EDI (Electronic Data Interchange) has been widely utilized to support Business Activities since it has such advantages as fast transfer of information, less documentation work, efficient information exchange etc. Recently e-business environment has urged the traditional EDI system to be changed to ebXML framework. To apply the ebXML framework to a certain industry, it is required to implement Business Process (BP), Core Component (CC), Collaboration Protocol Profile (CPP), Collaboration Protocol Agreement (CPA), Messaging system etc. We have selected the port and logistics industry as a target domain to apply ebXML framework, since the EDI usage ratio of it is relatively higher than other industries. In this paper, we have analyzed the current status of EDI system and transaction processes in the port and logistics industry. We have defined the business process that will be registered in the registry/repository, the main component of ebXML framework, using UN/CEFACT modeling methodology. And Business Collaborations, Business Transactions, Business Document Flows, Choreography, Pattern, etc. are represented using UML according to UN/ CEFACT modeling methodology, to apply ebXML Framework to the port and logistics distribution industry. Also we have suggested the meta methodology for applying the ebXML framework to other industries.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.7
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• pp.813-826
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• 1999

실시간 멀티미디어 응용의 등장으로 멀티캐스트와 QoS(Quality of Service) 지원이 필수적인 망 서비스로 부각되고 있다. 이에, ATM 기반의 인터넷에서 IP 멀티캐스트의 효율적인 처리를 위하여 MARS(Multicast Address Resolution Server)가 제안되었고, 기존의 최선 서비스 기반의 인터넷에서 QoS(Quality of Service)를 지원하기 위하여 RSVP(Resource Reservation Protocol)가 제안되었다. 본 논문에서는 ATM 망에서 QoS가 지원되는 IP 멀티캐스트 서비스를 제공하기 위하여 MARS 구조에서 RSVP를 지원하는 두 가지 방안을 제안하고, 시뮬레이션을 통하여 그 성능을 분석하였다. 제안하는 두 가지 방법은 각각 'RSVP 전 홉 노드 방식'과 'MARS 서버 방식'이라 명명하였다. RSVP 전 홉 노드 방식은 송신원으로부터 ATM 망으로 진입하는 노드와 수신원을 향하여 ATM 망을 진출하는 노드 간에 각각 일대일 양방향 VC를 설정하여 멀티캐스트 그룹에 속하는 수신원들이 보내는 자원 예약 메시지를 ATM 망에서 전송하는 방안이다. MARS 서버 방식은 ATM 망을 진출하는 노드와 MARS 서버간에 MARS 제어 메시지 교환을 위해 존재하는 ATM VC를 사용하여 RSVP의 자원 예약 메시지를 전송하고, MARS 서버가 RSVP 자원 예약 메시지를 처리하도록 그 기능을 확장함으로써 ATM 망에서 필요로 하는 제어 VC 수를 절약할 수 있는 방안이다. 시뮬레이션을 통하여, MARS 서버 방식은 ATM 제어 VC의 수를 절약할 수 있을 뿐 아니라 경우에 따라 RSVP 자원 예약 메시지 전달 지연을 줄일 수도 있음을 볼 수 있었다. 그러나, MARS 클러스터 내에 동시에 존재하는 RSVP 흐름이 많을 때에는 MARS 서버 방식의 경우 MARS 서버에서의 병목 현상으로 인해 성능이 저하될 수 있다.Abstract Emerging real time multimedia applications require multicast service with a QoS(Quality of Service) support. An overlay service architecture MARS(Multicast Address Resolution Server) is proposed to support IP multicast over an ATM network, and a resource reservation protocol RSVP is proposed to provide QoS support in the Internet which is originally based upon best effort service only. In this paper, we propose two schemes to support IP multicast service with QoS support over ATM networks: 'RSVP Previous Hop Node(RPHN) scheme' and 'MARS server based scheme'. In RPHN scheme, the RSVP reservation messages are transported via one-to-one ATM control VC from the egress nodes to the ingress nodes of the the multicast flow set up between each pair of nodes. The RSVP message processing occurs at the ingress nodes of the multicast flow. Whereas, in the MARS server based scheme, the RSVP reservation messages are transported via the MARS control VCs between the egress nodes and the MARS server. The RSVP message processing burden is imposed at MARS server in this scheme. For MARS server based scheme, no additional ATM VC is required for RSVP reservation message transmission, while the processing burden at the MARS server is high. Simulation results show that the MARS server based scheme, may accomplish RSVP reservation message delivery with smaller delay as well as saving of the number of ATM VCs. When the number of simultaneous RSVP flows in the MARS cluster is large, however, MARS based scheme may suffer performance degradation since MARS server becomes a performance bottleneck.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.10 no.3
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• pp.213-219
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• 2017

Recently, the Internet Protocol Television (IPTV) services have become very common, enabling various Internet-based services as well as watching TV. In the IPTV system, a Set-Top box (STB) plays a key role as a network terminal device that transmits and receives realtime multimedia contents. In addition, the IPTV networks are usually supported by broadband optical fiber-base network such as fiber-to-the-home (FTTH), However, a general IPTV STB is regarded as one of the local area network (LAN)-attached devices while sharing the bandwidth of the LAN (e.g., Ethernet). In order to overcome the limited bandwidth utilization by fully facilitating the broadband bandwidth (e.g., 1 Gbps) of the optical fiber-based network, we propose a new FC (Fiber Channel)-embedded IPTV STB which can be directly attached to the optical fiber network. Then, we verify that the impacts of the proposed FC-embedded IPTV STB by organizing the the FC-AL (Fiber Channel-Arbitration Loop) network equipped with the FC-embedded IPTV We measures the average Start-up Delay, Average Reject Ratio and the Number of Concurrent Users through extensive simulations to investigate the performances of the suggested FC-AL-based IPTV network. Surprisingly, the IPTV network architecture with the proposed FC-embedded IPTV STBs has an excellent average start-up delay of less than 10 msec, an acceptable average reject ratio of less than 3 % as well as a linear increase of the number of concurrent users when extending the architecture. This reveals that the proposed FC embedded STB has a superior impacts on the performance of the entire IPTV network by effectively utilizing the broadband bandwidth of the fiber optic-based network.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.19 no.5
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• pp.106-116
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• 2018

Recently, the role of the contact center for business-to-consumer (B2C) operations is becoming more and more important as the customer contact point. In particular, an Internet Protocol (IP)-based contact center system is made up of a complicated information system in order to accommodate various customer channels, in addition to the telephone, and to respond in real time. However, until now, evaluations of contact centers have focused on customer service-based research from inbound contact centers. We used the contact center as a measure of performance, focusing on indicators that have traditionally influenced customer satisfaction, such as response rates and service levels. There is insufficient research on the characteristics of the services that a contact center should have and on the evaluation models for information systems. The role of information systems is becoming important as the latest contact center, which has moved from the TDM-driven digital phone system center to the IP-based contact center, accommodates a variety of digital channels other than voice phones. In particular, as offline branches decrease due to the development of the Internet and mobile phones, non-facing responses to customers are important, so the contact center has influenced the enterprise. Therefore, we developed an evaluation model not only in terms of customer service, but also from information system and business aspects, using the AHP and verifying the evaluation model through empirical cases. In particular, content analysis was used to ensure objectivity of AHP evaluation items.

• Journal of KIISE:Information Networking
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• v.35 no.1
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• pp.46-55
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• 2008

As a mobility support for IP have studied, Internet Engineering Task Force(IETF) standardized the Mobile IPv6(MIPv6) protocol. When a mobile node moves between subnets, MIPv6 maintains connectivity to network and supports seamless communication, and these processes are called a Handover. Whenever the mobile node moves between subnets, the Handover is performed. The mobile node can not communicate during the Handover. This period is Galled Handover latency. To reduce this latency, mipshop working group standardizes Fast Handovers for Mobile IPv6(FMIPv6), but latency which the mobile node registers its new care-of address to a home agent and a correspondent node is still long. To solve this problem, we propose a scheme that the mobile node registers the new care-of address to the home agent and initiates Return Routability procedure in advance during layer 2 handover, based on FMIPv6 and IEEE 802.11. We analyze MIPv6, FMIPv6 and the proposed scheme in term of packet transmission cost during the Handover. Compared to MIPv6 the proposed scheme gains 79% improvement, while it gains 31% improvement compared to FMIPv6.

• Journal of KIISE:Information Networking
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• v.36 no.6
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• pp.492-504
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• 2009

The delivery of multimedia that efficiently adapts its bit-rate to changing network characteristics and conditions is one of the important challenging tasks in the design of today's real-time multimedia streaming systems such as IPTV, Mobile IPTV and so on. In these work, the primary focus is on network congestion, to improve network stability and inter-protocol fairness. However, these existing works have problems which do not support QoE (Quality of Experience), because they did not consider essential characteristics of contents playback such as the media continuity. In this paper, we propose a novel rate control scheme for improving the QoE of multimedia streaming service in the Internet congestion, called NCAR (Network and Client-Aware Rate control), which is based on network-aware congestion control and client-aware flow control scheme. Network-aware congestion control of the NCAR offers an improving reliability and fairness of multimedia streaming, and reduces the rate oscillation as well as keeping high link utilization. Client-aware flow control of NCAR offers a removing the media discontinuity and a suitable receiver buffer allocation, and provides a good combination of low playback delay. Simulation results demonstrate the effectiveness of our proposed schemes.