• Proceedings of the IEEK Conference
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• 2003.11c
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• pp.153-156
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• 2003

In this paper, we propose QoS guarantee system fur multi-functional VoIP end Terminal. This system guarantees low delay of voice data for Internet telephony in VoIP end terminal that has various kinds of Internet dependant application. QoS system we propose support low delay transmission in VoIP terminal interface.

• Proceedings of the IEEK Conference
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• 2000.11c
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• pp.133-136
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• 2000

In this paper, a new service architecture for IP Telephony, based on the ITU-T standard H.323［1］, is proposed. This architecture uses mobile Agents and existing architectural concepts taken from Intelligent Network［IN］. This IP service architecture enables telecom services deployed through mobile service agents on a per user basis, which results in several advantages when compared to centralized service architecture. The paper demonstrates that the flexible and extensible architecture can accommodate a wide variety of future services.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.6
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• pp.63-69
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• 2010

Today, Internet Telephony Services based on VoIP are gaining tremendous popularity for general user. Therefore a various demands of the user keep up increase, the most important requirements of these is voice security about telephony system. It is needed to ensure secret of voice call in a special situation. Due to the fact that many users can connect to the internet at the same time, VoIP can always be in a defenseless state by hackers. Therefore, in this paper, we have developed VPN IP-PBX for the voice security and measured conversation quality by adopting VPN IPsec based on SIP and using tunnel method in transmitting voice data to prevent eavesdrop of voice data. This VPN IP-PBX that is connected Soft-phone provide various optional services.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.19 no.3
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• pp.95-105
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• 2009

Spam call is one of the main security threat in VoIP services. In this paper, we have designed simulation model for performance evaluation of VoIP spam defense mechanism. The simulation model has functions for performance evaluation such as calls generation and input/output comparison. Four representative caller models have been developed for performance evaluation and each model has its own characteristics as statistical parameters. The target mechanism of performance evaluation is SPIT(Spam over Internet Telephony) level decision algorithm, and we have derived SPIT levels of caller models. The performance evaluation model is designed using the DEVS formalism and DEVSJAVA$^{TM}$ is exploited for development and execution of simulation models.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.2
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• pp.947-950
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• 2005

Voice of IP(VoIP) technology provides voice service as well as data service via Internet. It has been a promising technology as Internet grows fast and the requirements are increasing. Recently, serveral protocols have been created to allow telephone calls to be made over IP networks, notably, SIP and H.323. Due to introducing SIP and H.323, In this paper, we consideration interoperability of internet telephony service between ITSPs(internet telephony service provider)based on H.323.

• TTA Journal
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• s.67
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• pp.154-157
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• 2000

• Journal of KIISE:Information Networking
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• v.32 no.3
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• pp.350-358
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• 2005

Recently, VoIP technology is being accepted as are the most promising Internet telephony service, due to the substitution effect of traditional telephony service. Two standards, i.e, . H.323 and SIP. have emerged for signaling and control for Intemet telephony, of which SIP provides far lower complexity and rich extensibility. It is important to secure components of SIP in order to develop various services. Generally, open source codes provide basic functions of SIP as well as complicated structure, but are difficult to extend. In this thesis, we focused on offering interface mechanism between application and SIP User Agent to easily extend for various VoIP services. This thesis describes what function is needed for SIP User Agent, how to define the internal data structure, and how to define the internal processing procedure. The check iist derived through participating the interoperability event for stabilized SIP User Agent is also suggested.

• The Transactions of the Korea Information Processing Society
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• v.6 no.10
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• pp.2740-2751
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• 1999

Critical to more widespread use of Internet telephony are the smooth interoperability with the existing telephone network and the improved quality of voice connections. Of these requirements, this interoperability comes through the use of Internet Telephony Gateway's which perform protocol translation between an IP network and the Public Switched Telephone Network. In this paper, we have focused on the necessity and possibility of interoperability, and furthermore derives the necessary requirements for interoperability between IP networks and PSTN. For this purpose, we have analyzed the signaling protocols for gateway system. Then, we have modelled the inter-working part using the Petri-Net model. Through reachability trees of the Petri-Net model, we have confirmed that interoperability is possible, and that characteristics of deadlock, liveness, and boundness are satisfied.

• IE interfaces
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• v.15 no.4
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• pp.391-400
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• 2002

The internet call systems using CTI(Computer Telephony Integration) functions are designed with system configuration, DFD(Data Flow Diagram) and ERD(Entity Relationship Diagram) in this paper. The internet call systems are constructed to cooperate with conventional CTI call center. The internet phone calls occurred from the web browser of customer can be connected throughout VoIP gateway and PBX to many counselors. The internet call systems can provide various services; customer information service, escorted browsing service, text chatting service, text sharing service, conference service, and statistical analysis service.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.556-559
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• 2001

VoIP(Voice over IP) Technology, turn voice services over traditional telephone network into internet, is highlighted because of easy adopting the value added services related voice In this paper, we described the user agent system architecture for internet telephony services based on SIP (Session Initiation Protocol)