• Annual Conference on Human and Language Technology
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• 2010.10a
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• pp.141-142
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• 2010

최근 컴퓨터와 통신 기술의 발달로 인해 세계 구석구석을 연결하는 인터넷(Internet)이 대중화되었고, 실시간 멀티미디어를 이용한 다양한 응용 서비스들이 출현하고 있다. 응용 서비스들 중의 하나인 VoIP(Voice over Internet Protocol)기반의 인터넷 전화 서비스는, 일반 사용자에게 인터넷 전화에 대한 인식을 확산시켰으며 기존의 음성 통신 시장에 새로운 변화를 가져왔다. 본 연구에서는 음성 데이터를 인터넷 프로토콜 데이터 패킷으로 변화하여 인터넷망에서 통화를 하는 통신 서비스 기술인 VoIP방식의 인터넷폰의 시험환경을 구축하여 향후 실질적인 활용을 통해 고품질 S/W의 개발을 촉진함으로써 높은 부가가치를 창출하고 경쟁력을 갖춘 제품의 개발 지원이 가능하다고 본다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.89-92
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• 2001

In this paper, it is proposed that RSVP and WFQ must be a good way of a better service for the better quality for Mobile If Network. For the Performance Analysis of working it was composed of Mobile IP and VoIP Network model, and further more test of the postpone and QoS was implemented. The results of the test is as follows, Before the movement of mobile agent was 2ms, after that, 3ms, And before QoS was adapted the value was 30ms, after being adapted, analyzed as 10ms. This research that the problem of put off was improved by adaping QoS in the mobile IP Network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.2A
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• pp.12-19
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• 2005

The VoIP(Voice over IP) has been worldwide used and already put to practical use in many fields. However, it is needed to ensure secret of VoIP call in a special situation. It is relatively difficult to eavesdrop the commonly used PSTN in that it is connected with 1:1 circuit. However, it is difficult to ensure the secret of call on Internet because many users can connect to the Internet at the same time. Therefore, this paper suggests a new model of Internet telephone for eavesdrop prevention enabling VoIP(using SIP protocol) to use the VPN protocol and establish the probability of practical use comparing it with Internet telephone.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.12C
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• pp.1238-1244
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• 2002

In this paper, H.235-based security mechanism for H.323 multimedia applications was implemented. H.235 covers authentication using HMAC, Diffie-Hellman key exchange, session key management for voice channel, and encryption functions such as DES, 3DES, RC2. Extra encryption algorithms such as SEED, and AES were also included for possible use in the future. And, we also analyzed the quality of service (QoS), the requirement of implementation, and interoperability to the result in this study. The results could be applied to secure simple IP phone terminals, gateways, or gatekeepers.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.1
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• pp.130-140
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• 2010

SIP (Session Initiation Protocol) is a signaling protocol to provide IP-based VoIP (Voice over IP) service. However, many security vulnerabilities exist as the SIP protocol utilizes the existing IP based network. The SIP Malformed message attacks may cause malfunction on VoIP services by changing the transmitted SIP header information. Additionally, there are several threats such that an attacker can extract personal information on SIP client system by inserting malicious code into SIP header. Therefore, the alternative measures should be required. In this study, we analyzed the existing research on the SIP anomaly message detection mechanism against SIP attack. And then, we proposed a Co-occurrence based SIP packet analysis mechanism, which has been used on language processing techniques. We proposed a association rule generation and an attack detection technique by using the actual SIP session state. Experimental results showed that the average detection rate was 87% on SIP attacks in case of using the proposed technique.

• 한국정보통신설비학회:학술대회논문집
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• 2009.08a
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• pp.99-103
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• 2009

Recently, as wired and wireless communication services have rapidly developed and multimodal mobile devices which have various characteristics have widely spread, the need for new convergence services increases. The growing population of VoIP technologies and the high communication expense yield that the market of IP based telephony such as WiFi phone and IP phone is substituted for one of the conventional PSTN telephony. With the help of this trend, the wireline network operators desire to find a market in mobile networks. Therefore, they focus on Fixed Mobile Convergence (FMC) service as one of the key factors to accomplish this goal. FMC services are able to provide the mobility of voice services between circuit switched and packet switched networks. IP Multimedia Subsystem (IMS) based Voice Call Continuity (VCC) is one of the schemes to embody FMC services. As Application Server (AS) which has this VCC function provides seamless handover of services between heterogeneous networks, FMC subscribers can communicate seamlessly with others m WiFi domain and COMA domain using WiFi-COMA dual phone. Most of enterprises have already introduced IP network infrastructure and IP-PBX (Private Branch eXchange) for telephony. However, the problems of high communication cost and work inefficiency due to frequent outside jobs or business trips have remained. In order to solve these problems, demands for enterprise FMC services increase. In this paper, we introduce a new IP-PBX based VCC model that can provide seamless handover of voice services between WiFi and COMA networks for enterprise users and we investigate some interworking and security issues between Soft Switch (SSW) and IMS, or between IMSs. In addition, we introduce a new service that can provide the continuity of voice sessions as well as video sessions using Multimedia Session Continuity (MMSC) technology which has evolved from VCC. This service is expected to be one of the next-generation personalized services based on user's context.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.682-685
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• 2002

VoIP(Voice over IP) Technology is highlighted because of easy adopting the value added services related voice In this paper, we described the Internet telephony service based on SIP. Especially, we described the extension for re-INVITE function. Re-INVITE function is useful for cail transfer service or conference service.

• KIISE Transactions on Computing Practices
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• v.20 no.12
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• pp.688-693
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• 2014

With the advent of ubiquitous environments, the smart phone has come into wide use and the demand for various content increases. Thus, in order to efficiently utilize limited resources cognitive radio technology is regarded as a possible solution. Besides spectrum sensing or access schemes, the provision of VoIP traffic service for secondary users with limited spectrum resources is a very important issue. In this paper, a resource allocation and control system for VoIP QoS provision in cognitive radio networks is proposed. Firstly, as the system model, the time structure of the network is addressed and, according to the structure, a bandwidth broker is proposed. In addition, based on available bandwidth estimated by the bandwidth broker, a connection admission control for secondary users is developed. It is demonstrated that the provision of VoIP QoS is greatly affected by channel utilization, the number of channels, and the length of timeslot.

• Proceedings of the Korean Information Science Society Conference
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• 2008.06d
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• pp.110-115
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• 2008

음성 데이터를 IP기반의 패킷망을 통해 전송하는 기술인 VoIP(Voice over Internet Protocol) 기술은 음성 데이터를 기존의 PSTN(Public Switched Telephone Network)망을 통해 전송하는 방식에 비해 비용 절감 등의 장점을 가지고 있다. 그러나 VoIP가 기존의 PSTN망을 대체하기 위해서는 QoS(Quality of Service)의 보장과 보안이 제공되어야 한다는 문제점을 가지고 있다. VoIP망에서 보안을 위해서는 사용자간에 전송되는 음성 데이터에 대한 보안과 초기의 세션 연결 시 사용자를 인증하는 과정이 고려되어져야 한다. 실질적인 대화 내용인 음성 데이터의 보안도 중요한 부분이지만 대화에 참여하는 사용자를 인증하는 과정이 선행되어야 한다. VoIP에서는 세션 연결 설정을 위해 H.323과 SIP를 사용하고 있으며, 최근에는 H.323에 비해 간단한 SIP가 주목을 받고 있다. RFC3261에서는 SIP를 이용해 세션 연결을 하는 과정에서 사용자를 인증하기 위한 몇 가지 인증 메커니즘을 제시하고 있다. 본 논문에서는 SIP를 이용하여 세션을 연결하는 과정에서 사용자의 인증을 위해 사용되는 인증 메커니즘 중 한 가지인 HTTP Digest Authentication의 취약점을 분석하고, 이를 보완하기 위한 새로운 인증 메커니즘을 제시한다.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.124-126
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• 2002

Voice over Internet Protocol(VoIP) is expected to be a major application on the Internet in the future This paper propose an approach to VoIP that uses Differentiated Service and Multi-protocol Label Switching(MPLS) to provide quantitative QoS guarantees over an IP network. An algorithm that determines QoS-constrained routes is proposed and a framework that uses such an algorithm for traffic engineering is outlined. the key component of this framework is a Centralize Resource Manager(CRM) responsible for monitoring and managing resources within the network and making all decisions to route／reroute traffic according to QoS requirement