• Proceedings of the Korea Contents Association Conference
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• 2006.11a
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• pp.26-30
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• 2006

Internet phone service is a new service technology that provides voice call services through Internet not through the pre-existing PSTN. It enables a cheap voice call service regardless of distance. We may expect that the Internet phone service may substitute for the voice call service through the PSTN, but not in a short period. There are several problems to be solved for this transition, such as, voice call quality, numbering scheme, billing, standardization, and support of several functions. In this paper, we provided and designed a UA (User Agent) that can support functions regarding voice call, such as call forwarding, auto-connection, call rejection and restriction of individual call, using SIP (Session Initiation Protocol) which is proposed by SIP-Working Group as the standard Internet phone service management protocol.

• Journal of Korea Multimedia Society
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• v.11 no.7
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• pp.1016-1024
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• 2008

In this paper, we designed and implemented the ACE(ACtive Emergency call service system) system for emergency call service actively. ACE system has two physical components; E-Device(Emergency Mobile Device) and E-Server(Emergency Server). The role of E-Device is the mobile device in order to call emergency by using mild handi-capped, the elderly and children who are able to communicate theirs intention to another. E-Server is the server for management E-devices with realtime monitering. E-Device will be developed to the portable size for easily mild handi-capped, the elderly and children. When they need the service of emergency call, the button of E-device can be used and the call signal is transmitted to the emergency office and the guardian through Internet and CDMA. E-server should be developed the integrated control system for management of E-Devices basically. And it also supported to realtime monitoring of E-devices with respect to high quality of emergency call service for rise the efficiency.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.8
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• pp.1958-1969
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• 1996

This paper proposed a new approach to adaptive call admission control based on a neural network for multiple service classes with different quality of service (QoS) in the ATM-based Broadband Integrated Services Digital Networks. the proposed method extend Hiramatsu's neural network based "leaky pattern table" method for the single QoS[1, 2, 3] to deal with multiple services with different QoS by constructing multiple pattern tables based on each service's acceptance or rejection at the call set-up requests, and by simultaneously controlling each service's QoS according to the target QoS of the service and the trunk capacity. Computer simulation results on two service classes with different traffic characteristics and different cell loss rates as QoS, highlight good performance and effectiveness of the proposed call admission controller for multiple service classes.e classes.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 1987.04a
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• pp.67-71
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• 1987

No 1A ESS must operate in real time. Call processing capacity is the maximm number of originating plus incoming calls which the processor can process in some fixed interval of real time shile all service criteria have been satisfied. In this paper. we discuss the postcutower method of determining call processing capacity of No 1A ESS which can be applicable to an in service office and explain the usuage mithod of estimated call processing capacity of No 1A ESS in service

• IE interfaces
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• v.25 no.3
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• pp.327-337
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• 2012

The development of information and communication technology affects people's life in social, cultural, and economic aspects. When this happens in public sector, it gives way more benefits than in private sector because of its high accessibility by public. Among the technological public services in Korea, call center service which provides administrative services by telephone and internet had been spotlighted as a new type of communication between people in demand and public service provider. Public call center service is expected to be continuously diffused in years due to its accessibility and convenience for public users. This study analyzes diffusion pattern of public call center service in Korea using Bass model and tries to suggest appropriate diffusion strategies. For practical cases, three most popular public call centers in Korea are analyzed in light of diffusion pattern and operating strategies. Our analyses identify that public call centers in Korea are facing continuous diffusion in two years and there exist certain strategies to efficiently expedite the diffusion.

• Proceedings of the IEEK Conference
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• 2001.06c
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• pp.225-228
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• 2001

There are two standards currently compete for the dominance of IP telephony architecture. Those are H.323 protocol suit by International Telecommunication Union Sector T(ITU-T) and Session Initiation Protocol/Session Description Protocol(SIP/SDP) by International Engineering Task Force(IETF). This paper has been studied a adaption VoIP in home networking, Especially SIP-based call screening service in home gateway. And then this paper has designed SIP-based call screening service in home gateway working protocol, verified them.

• Journal of KIISE:Information Networking
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• v.30 no.5
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• pp.621-630
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• 2003

In this paper, we implemented SIP protocol that supports IPv6 and differentiated call processing scheme for NGN(Next Regeneration Network). In NGN, SIP processes call signaling among various application services. A softswitch and SIP server must give priority to sensitive services such as Fax, network management and home networking that require a fast call setup time. Also, the support of IPv6 is needed under consideration of All-IP. We proposed differentiated call processing scheme. The differentiated call processing scheme supports differentiated call processing as priority of service class on call processing in SW server We defined three service classes and use the Flow Label field of the IPv6 header for setting service class. Through the performance analysis, we proved that it improves throughput for call message with the high priority. The result of performance analysis demonstrates that differentiated call processing scheme gives better performance for the service requiring a fast session establishment in NGN.

• Journal of Information Technology Services
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• v.16 no.3
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• pp.185-203
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• 2017

Call center service is enabled by the use of a firm's various information technologies (IT) and accumulated knowledge. IT and knowledge resources have been considered a fundamental infrastructure for developing a firm's business capabilities. Recognizing this, this study examines whether a firm's IT and customer knowledge resources enhance its customer service process capability and thereby contribute to creating superior customer service, at the level of customer service representatives (CSRs). That is, constructs in this study were measured on a basis of CSRs' perception. Moreover, this study verifies the moderating role of CSRs' computer self-efficacy on the relationships between IT and customer knowledge resources and customer service process capability. To test the proposed hypotheses, this study conducted partial least squares (PLS) analysis with a total of 234 data which were collected on CSRs working at four call centers. The key findings are as follows: first, IT infrastructure integration and customer knowledge integration are positively associated with customer service process capability. Second, customer service process capability is a key determinant of customer service performance. Finally, CSRs' computer self-efficacy has a moderating effect on the relationship between IT infrastructure integration and customer service process capability. The details of the findings and implications are presented.

• The Journal of the Korea Contents Association
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• v.7 no.3
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• pp.34-41
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• 2007

The presence service which provides user's presence information by subscription and notification is one of SIP(session initiation protocol) extension services, and it is used importantly in VoIP(Voice over IP) and Instant Messaging service. In this paper, we propose a new method in which users can combine and control presence service and call processing services in various ways by extending call processing language, and only changed parts of the presence information are published instead of full presence information document. Each user registers full presence information document with his own call processing script during the first publication to a presence server. The presence server executes these call processing scripts, so it can provide various services with combination of presence service and call processing services during the presence subscriptions and notifications. Afterwards, users publish only changed parts of the presence information and the presence server notify only these changed parts to watchers. Therefore the efficiency of the overall system can be improved. The performance of our proposed model is evaluated by experiments.

• Journal of Communications and Networks
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• v.15 no.1
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• pp.15-24
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• 2013

Provisioning of quality of service (QoS) is a key issue in any multi-media system. However, in wireless systems, supporting QoS requirements of different traffic types is a more challenging problem due to the need to simultaneously minimize two performance metrics - the probability of dropping a handover call and the probability of blocking a new call. Since QoS requirements are not as stringent for non-real-time traffic, as opposed to real-time traffic, more calls can be accommodated by releasing some bandwidth from the already admitted non-real-time traffic calls. If the released bandwidth that is used to handle handover calls is larger than the released bandwidth that is used for new calls, then the resulting probability of dropping a handover call is smaller than the probability of blocking a new call. In this paper, we propose an efficient call admission control algorithm that relies on adaptive multi-level bandwidth-allocation scheme for non-realtime calls. The scheme allows reduction of the call dropping probability, along with an increase in the bandwidth utilization. The numerical results show that the proposed scheme is capable of attaining negligible handover call dropping probability without sacrificing bandwidth utilization.