• Journal of Communications and Networks
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• v.7 no.3
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• pp.353-366
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• 2005

In future telecommunications systems, behaviour will be defined by inexperienced users for many different purposes, often by specifying requirements in the form of policies. The call processing language (CPL) was developed by the IETF in order to make it possible to define telephony policies in an Internet telephony environment. However, user-defined policies can hide inconsistencies or feature interactions. In this paper, a method and a tool are proposed to flag inconsistencies in a set of policies and to assist the user in correcting them. These policies can be defined by the user in a user-friendly language or derived automatically from a CPL script. The approach builds on a pre-existing logic programming tool that is able to identify inconsistencies in feature definitions. Our new tool is capable of explaining in user-oriented terminology the inconsistencies flagged, to suggest possible solutions, and to implement the chosen solution. It is sensitive to the types of features and interactions that will be created by naive users. This tool is also capable of assembling a set of individual policies specified in a user-friendly manner into a single CPL script in an appropriate priority order for execution by telecommunication systems.

• 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.

• 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.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.9 no.4
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• pp.929-935
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• 2008

As internet telephony services based on Session initiation Protocol (SIP) enter the spotlight as marketable technology, many products based on SIPs have been developed and utilized for home and office telephony services. The call connection of an internet phone is classified into specific call connections and group call connections. Group call connections have a forking function which delivers the message to all of the group members. This function requires excessive message control for a call connection and creates heavy traffic in the network. In the internet cail system model. most of the call-setup messages are directed to the proxy server during a short time period. This heavy message load brings an unwanted delay in message processing and. as a result, call setup can not be made. To solve the delay problem, we simplified the analysis of the call-setup message in the proxy server, and processed the forking function distributed for the group call-setup message. In this thesis, a new system model to minimize the load is proposed and the subsequent implementation of this model demonstrates the performance improvement.

• International journal of advanced smart convergence
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• v.2 no.1
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• pp.24-26
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• 2013

In the process of a mobile network evolution to the All-IP, it is inevitable to experience a transient period embracing both circuit and packet based data traffics. At the stage of those hybrid networks, it is important to build them in an efficient manner in terms of resource utilization which is closely related to the overall system operation cost. Especially, the multimedia telephony is one of the essential services in the advanced packet based mobile networks. In this paper an effective method of system operation is proposed for building up the multimedia telephony service while the legacy network co-exists. The proposed solution is based on the careful investigation of the usage pattern of the multimedia services in the evolving networks. This method is also expected to be a useful guideline for the network resource planning.

• 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.

• 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.

• Journal of Information Processing Systems
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• v.18 no.1
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• pp.1-11
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• 2022

Online video telephony systems have been increasingly used in several industrial areas because of coronavirus disease 2019 (COVID-19) spread. The existing session initiation protocol (SIP)-based video call system is being usefully utilized, however, there is a limitation that it is very inconvenient for users to transmit additional information during conversation to the other party in real time. To overcome this problem, an enhanced scheme is presented based on augmented real-time reality (AR). In this scheme, augmented information is automatically searched from the Internet and displayed on the user's device during video telephony. The proposed approach was qualitatively evaluated by comparing it with other conferencing systems. Furthermore, to evaluate the feasibility of the approach, we implemented a simple network application that can generate SIP call requests and answer with AR object pre-fetching. Using this application, the call setup time was measured and compared between the original SIP and pre-fetching schemes. The advantage of this approach is that it can increase the convenience of a user's mobile phone by providing a way to automatically deliver the required text or images to the receiving side.

• 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.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2004.07a
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• pp.183-183
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• 2004