• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.1
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• pp.134-144
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• 2009

The Ubiquitous/Unified Communication Platform supports various multimedia communication tools such as VoIP, Email, Unified Messaging, Instant Messaging, Web Conferencing, Audio/Video Communication etc. In this paper we introduced the main functions and architecture of the Unified Communication Platform and we researched on the function analysis and design of the VoIP System that supports PC-to-PBX/PSTN Phone and PBX/PSTN Phone-to-PC communications through the connectivity and interoperation with PSTN.

• Proceedings of the Korean Information Science Society Conference
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• 2001.04a
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• pp.457-459
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• 2001

인터넷 전화라고도 불리는 VoIP(Voice over IP) 시스템은 기존의 전화망에 비해서 장거리 전화 서비스를 저렴하게 제공할 수 있고 인터넷 망의 다양한 멀티미디어 서비스를 음성 전화 서비스에서 수용할 수 있다는 장점을 가지고 있다. 또한 이러한 VoIP 시스템은 현재의 다양한 망 기술들이 IP 기술 기반의 차세대 All IP망으로 효과적으로 발전하기 위한 기반 기술이 된다는 중요한 의미도 가지고 있다. 본 고에서는 IETF(Internet Engineering Task Force)에서 멀티미디어 세션의 생성, 수정, 종료를 이해 제안한 SIP(Session Initiation Protocol)를 이용하여 이 같은 VoIP 시스템을 설계, 구현한 결과를 설명하고자 한다. 본 VoIP 시스템은 플랫폼에 독립적이며 멀티 쓰레드 프로그래밍이 가능한 자바(Java)를 이용해 구현되었다. 따라서 뛰어난 확장성과 포괄성 등의 특징을 가지고 있다. 또한 구현된 VoIP 시스템은 추후 다양한 멀티미디어 관련 서비스를 포함시키는 한편 관련 기술들을 보완하여 All IP망 형태의 테스트베드로 확장될 계획이다.

• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.445-447
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• 2003

최근 무선네트워크의 급속한 확대로 무선인터넷 접속 또한 크게 증가하고 있다. 무선인터넷을 이용하는 Voice over IP 서비스는 IP 기반의 인터넷과 셀룰러 네트워크를 합쳐 놓은 것과 유사하며 모바일 노드의 이동성 확보가 핵심 기술이다. 특히 VoIP 서비스 이용자는 지연이나 끊어짐에 대하여 매우 민감하므로 가급적 지연시간이 적은 핸드오프 기법이 필요하다. 본 연구에서는 무선네트워크를 이용하는 VoIP 서비스 프로토콜 중 하나인 SIP를 기반으로 이동성 관리를 위한 신호의 흐름을 도시하여 발생 가능한 지연의 구성요소를 분석하였으며 핸드오프 지연을 줄이기 위한 Prediction Shadow Registration을 제안하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.758-761
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• 2002

Voice of U(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, There are many change at internet telephony service. Internet telephony enables a wealth of new service possibility Users can control telephony service directly. In this paper, we design and implementation CPL client based on SIP system.

• Journal of the Korea Society of Computer and Information
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• v.17 no.3
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• pp.113-120
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• 2012

VoIP provides voice data service using existing IP networks and has received much attention recently. VoIP service has a variety of security vulnerabilities. Types of main attacks on VoIP service are tapping/interception, DoS attacks, spam, misuse of service attacks and the like. Of these, confidential information leak because of tapping/interception has been considered as a critical problem. Encryption techniques, such as SRTP and ZRTP, are mostly used to prevent tap and intercept on VoIP media information. In general, VoIP service has two service scenarios. First, VoIP service operates within a single private network. Second, VoIP service operates between different private networks. Both SRTP and ZRTP for VoIP media information within a single private network can perform encryption. But they can not perform encryption between different private networks. In order to solve this problem, in this paper, we modify SRTP protocol. And then, we propose an encryption method that can perform encryption of VoIP media information between the different private networks.

• Journal of Digital Contents Society
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• v.12 no.1
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• pp.85-94
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• 2011

QoS(Quality of Service) is defined as "The collective effect of service performance which determines the degree of satisfaction of a user of the service" by ITU-T Rec. E.800. While the use of VoIP(Voice Over Internet Protocol) has been widely implemented, persistent problems with QoS are a very important sue which needs to be solved. This research is finding the assignment of VoIP QoS to deduct how to manage the control system and presenting the QoS control process and framework under NGN(Next Generation Network) environment. The trial framework is the modeling of the QoS measurement metrics, instrument, equipment, method of measurement, the series of cycle & the methodology about analysis of the result of measurement. This research underlines that the vulnerability of the VoIP protocol in relation to its QoS can be guaranteed when the product quality and management are controlled and measured systematically. Especially it's very important time to maintain the research about VoIP QoS measurement and control because the big conversion of new network technology paradigm is now spreading. In addition, when the proposed method is applied, it can reduce an overall delay and can contribute to improved service quality, in relation to signal, voice processing, filtering more effectively.

• Electronics and Telecommunications Trends
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• v.19 no.3 s.87
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• pp.136-144
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• 2004

인터넷의 발달 및 VoIP의 보급으로 인해 VoIP 서비스의 품질에 대한 관심이 증가하고 있다. 그 동안은 망사업자 관점에서 망의 품질을 개선하기 위한 MPLS, Diffserv, RSVP 등의 연구가 진행되어 왔으나, 실제로 서비스 품질은 망뿐만 아니라 단말 등의 품질에도 영향을 받기 때문에 망 사업자의 관점에서 보는 서비스 품질 기준이 아닌, 고객의 관점에서 인식 가능한 수준에서의 종단간 서비스 품질을 다룰 필요가 있다. 본 고에서는 서비스 품질이란 무엇인지 살펴보고, 국제표준단체의 서비스 품질 관련 연구 및 VoIP 서비스를 위한 음성 품질 평가 기술에 대하여 살펴본다.

• Information and Communications Magazine
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• v.21 no.4
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• pp.65-73
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• 2004

인터넷전화 서비스를 제공하기 위한 대표적인 기술이 VoIP(Voice over Internet Protocol)이다. VoIP 기술에 의한 전화 서비스는 기존의 PSTN(Public Switched Telephone Networks)서버에 비하여 경제적이고, 향후 멀티미디어 서비스 지원 등의 특징을 가지기 때문에 보급의 확산이 기대된다. VoIP 기술은 유선에서 뿐만 아니라, 무선에서도 VoIP 기술을 채택하여 유무선 통합의 핵심기술로서 IETF, ITU 등에서 작업이 추진되고 있다［1］. 이와 같이 VoIP 서비스의 확대가 예상됨에 따라 사용자의 인증, 메시지 보호 등 보안 서비스의 중요도가 증대되고 있다［2］.(중략)

• Proceedings of the Korea Contents Association Conference
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• 2006.11a
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• pp.75-79
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• 2006

In a VoIP(Voice over IP) system which support the realtime speech service, speech quality is deteriorated by the delay, the jitter, the loss, and the reversed packet order. In this thesis, APBC for receiver site is proposed, which compensate the jitter by the adaptive playout algorithm and conceal the packet loss, and align the packet order. Also, a VoIP application system is implemented, and the performance of APBC is verified on the implemented system by measuring the processing speed and the speech quality. From the result, processing speed is 257$\mu$sec, which is fast enough to deal with packet being received in realtime. Also, the speech quality by MOS(Mean Opinion Score) is improved as 18 percent compared with algorithm of fixed playout delay.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.2
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• pp.634-645
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• 2014

VoIP (Voice over Internet Protocol) is a technology of transmitting and receiving voice and data over the Internet network. As the telecommunication industry is moving toward All-IP environment with growth of broadband Internet, the technology is becoming more important. Although the early VoIP services failed to gain popularity because of problems such as low QoS (Quality of Service) and inability to receive calls as the phone number could not be assigned, they are currently established as the alternative service to the conventional wired telephone due to low costs and active marketing by carriers. However, VoIP is vulnerable to eavesdropping and DDoS (Distributed Denial of Service) attack due to its nature of using the Internet. To counter the VoIP security threats efficiently, it is necessary to develop the criterion or the model for estimating the information security level of VoIP service providers. In this study, we developed reasonable security indicators through questionnaire study and statistical approach. To achieve this, we made use of 50 items from VoIP security checklists and verified the suitability and validity of the assessed items through Multiple Regression Analysis (MRA) using SPSS 18.0. As a result, we drew 23 indicators and calculate the weight of each indicators using Analytic Hierarchy Process (AHP). The proposed indicators in this study will provide feasible and reliable data to the individual and enterprise VoIP users as well as the reference data for VoIP service providers to establish the information security policy.