• Journal of Internet Computing and Services
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• v.5 no.6
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• pp.31-43
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• 2004

Voice over Internet Protocol (VoIP) is currently a popular research topic as a real time voice packet transmission method. But current Internet environment do not guarantee the quality of voice when we take a side view of delay, jitter and loss. Up to now, many voice based evaluation algorithms have been used to measure speech quality of VoIP systems. However, these algorithms have the defects that their results are different according to voice samples and some algorithms can not take network environment for speech transmission path. The E-model can be used to solve the problems of these algorithms. In this paper. we introduce VoIP planning guidelines through the various analysis of E-model which can model impairments of network quality as well as VoIP equipment quality systematically, We, also, show the evaluation method and results of speech transmission quality.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.8 no.6
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• pp.21-28
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• 2008

In this paper, we proposed an algorithm that reduces call setup time between the IP PBXs installed in the main office and several branch offices. In the IP PBXs installed in branch offices, we prepare some internal memory spaces, and they keep the frequently used extension numbers and the corresponding IP PBX's addresses. In order to call an extension number registered to the other remote IP PBX from an extension number registered to the local IP PBX, the local PBX searches for the destined extension from its internal memory first. If the extension is found, then the local PBX uses its corresponding IP PBX address stored along with the extension, instead of making a query to the main office's IP PBX. By this technique, we could shorten the average call setup time and verified it by an experiment.

• Journal of Internet Computing and Services
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• v.9 no.2
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• pp.139-145
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• 2008

CRM(Customer Relationship Management) improves the competitiveness of small companies to large enterprises. E-mail, SMS (Short Message Service), telephony service, DM (Direct Mail) are used for customer response in many companies. In order to improve company's profit, business diversification is in progress with protecting the succession of existing customers, and the only of new customers. Futhermore, BPR (Business Process Re-engineering) and BPM (Business Process Management) of the CRM can be easily found in many companies, because it is the first contact point to the company at the customer's view. This paper discusses the development of CRM, and proposes a call object routing agent that is the main engine of the CRM. Result of performance evaluation of proposed method show that the system is effective and powerful enough to use at the multi-channel environment.

• Journal of the Korea Society of Computer and Information
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• v.8 no.1
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• pp.97-102
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• 2003

In this paper it Is to suggest that beyond the Previous communication concept of exchange, transmission mounting. and terminal , the call control system function should be added to the call control exchange server, to make an external telephone call available through the internet by connecting the interior line and the Internet. The function evaluation of call control system is found to become stable when the callers are 10 rather than 20 above, as a result of the more than ten times repeated concurrent calls. Furthermore, the evaluation shows that the program responds Quickly within a second, even when there are more than 100 callers as well as secures a stable system operation and mutual inter-feasibility due to a fast connectivity and a superb function.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.1
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• pp.7-16
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• 2012

In this paper we describe a VoIP transcript system which is able to record call conversation between counselor and customer in an IP contact center based on IP telephony environment. The transcript system, designed and implemented in this paper, uses packet sniffering to capture packets without imposing network overhead on overall system. It can decode H.323 and SIP which are used to setup call sessions in VoIP environment and captures voice data and record without any loss of contents. Implemented transcript system can be integrated with CTI system in that it can manage and record call more effectively. It is designed generically so that it is implemented both on Windows and Linux environment.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.637-640
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• 2011

Recently, rapid spreading of smart-phone make an expectation for increase of practical application for MANET(Mobile Ad-Hoc Network) which is not used infrastructure like as base-station. One of important application for MANET will be VoIP(Voice over Internet Protocol) known as Internet telephony. On the other hand, information intrusions causing serious problems is not allowed exception of MANET. Especially, there are some dangerous problems of intrusions to MANET, differently to other networks, because of it's usage of military purpose or emergency application of rescue. In this paper, to analyze this intrusion problem on MANET with blackhohe attacks, effect of intrusion to transmission performance is studied. VoIP traffic is assumed as an application service on MANET, computer simulation with NS-2 is used for this analysis.

• Electronics and Telecommunications Trends
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• v.15 no.2 s.62
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• pp.1-14
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• 2000

인터넷 트래픽의 급격한 증가와 인터넷 서비스 보급의 대중화로 인해 인터넷전화 기술이 전세계 전기통신사업자 및 인터넷 서비스사업자들의 주요 관심사가 되고 있다. 인터넷전화란 인터넷망을 통해 전화 혹은 음성 서비스를 제공하는 기술을 의미한다. 인터넷전화는 음성전화 서비스 외에도 음성사서함, 호출, 팩스 등의 부가서비스뿐만 아니라 전자상거래, 화상회의, 홈쇼핑 등의 멀티미디어 서비스에 대한 핵심 기반 기술이라는 점에서 사업자 및 사용자들의 관심이 크다. 이에 본 글에서는 인터넷전화 관련 시장 및 표준화 동향 그리고 국내 서비스 보급 현황 및 문제점 등을 분석함으로써 인터넷전화 보급 촉진 및 기술 개발을 위해 나아가야 할 방향을 제시하고자 한다.

• 한국경영정보학회:학술대회논문집
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• 2007.06a
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• pp.652-657
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• 2007

인터넷의 발전으로 많은 시스템과 서비스가 변화를 거듭하고 있는 가운데, 당초 큰 기대를 모았던 인터넷전화는 저렴한 이용료 이외의 많은 장점에도 불구하고 활성화가 더디게 진행되어 왔다. 주요 원인으로 기존 통신사업자들의 입장 및 정책적인 규제 등이 지적되면서 다양한 측면에서의 활성화 방안이 논의되고 있다. 한편 기존의 통신사업자 이외에 인터넷포탈 및 방송사업자들이 시장에 새롭게 진입하면서 인터넷전화 시장은 새로운 변화를 맞이하고 있으며, 그로 인해 인터넷전화 시장은 보다 복잡한 경쟁관계를 보이고 있다. 본 연구에서는 인터넷전화 시장의 발전과정에 대한 분석을 통해 새롭게 변화하고 있는 인터넷전화 시장의 경쟁 속에서 인터넷전화 사업자들에게 전략적인 제언을 하고자 한다.

• Journal of Communications and Networks
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• v.12 no.5
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• pp.480-491
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• 2010

Along with the wide acceptance of IEEE 802.11 wireless local area network (WLAN), new applications such as Internet protocol (IP) telephony over WLAN are fast emerging today. For battery-powered IP phone devices, the life time extension is a key concern for the market acceptance while today's 802.11 is not optimized for snch an operation. In this paper, we propose a novel idle mode operation, which comprises paging, idle handoff, and delayed handoff. Under the idle mode operation, a mobile host(MH) without any active session does not need to perform handoff within a predefined paging area (PA). Only when it enters a new PA, an idle handoff is performed. The proposed idle mode allows an MH without traffic to extend its life time. We develop a new analytical model in order to comparatively evaluate our proposed scheme. The numerical resnlts demonstrate that the proposed scheme outperforms the existing schemes with respect to power consumption.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.442-444
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• 2002

본 논문은 인터넷 전화 (Internet Telephony)의 대표적 프로토콜인 SIP의 이동성과 효율적인 네트워크 대역 이용에 관한 연구이다. 이동성을 제공하기 위해 모바일 IP를 사용하기에는 삼각 라우팅, 캡슐화 오버헤드와 같은 단점이 있다고 알려진 것에 비해 이러한 모바일 IP 기반 SIP 애플리케이션의 가능성과 적합성을 보인 기존의 연구 〔1〕에서 더 나아가 무선통신에서 발생하는 핸드오프 동안에도 패킷 손실을 최소화 할 수 있는 패킷 당 오디오 프레임 수의 상관관계를 설계하고, 무선통신 종단 간 물리적 거리 변화에 따라 동적으로패킷 당 프레임 수를 조절하여 적용할 수 있는 적응적 애플리케이션 개발을 위한 표준을 제시한다.