• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.273-276
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• 2011

Voice over Internet Protocol(VoIP) compared to the traditional PSTN communications costs and because of the ease of use has been widespread use of VoIP. Broadband Convergence Network (BCN) as part of building with private Internet service provider since 2010, all government agencies are turning to the telephone network and VoIP. In this paper, we used the Internet on your phone in the IETF SIP-based IP-PBX is a hacking attack analysis studies. VoIP systems are built the same way as a test bed for IP-PBX hacking attacks and vulnerabilities by analyzing the results yielded. Proposes measures to improve security vulnerabilities to secure VoIP.

• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.8
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• pp.1408-1414
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• 2008

Development of internet and IP network technology creates various fusion services such as IPTV, VoIP and so on. Next generation IPv6 which will solve lack of IP is very important on IPTV which needs best quality of service about security, QoS and bandwidth. In this paper, we suggest a routing protocol standard in which we can service best quality of image using PSNR which is most commonly used as a measure of quality of reconstruction in image on IPv6 network.

• Journal of the Korea Society of Computer and Information
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• v.14 no.12
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• pp.147-156
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• 2009

This paper proposes a SRTP key exchange scheme using split transfer of divided RSA public key in SIP-based VoIP environment without PKI. The existing schemes are hard to apply to real VoIP environment, because they require a PKI and certificates in the end devices. But in case of ZRTP. which is one of existing schemes, it's able to exchange SRTP Key securely without PKI, but it is inconvenient since it needs user's involvement. To solve these problems, the proposed scheme will split RSA public key and transmit them to SIP signaling secession and media secession respectively. It can defend effectively possible Man-in-The-Middle attacks, and it is also able to exchange the SRTP key without the user's involvement. Besides, it meets the requirements for security of SRTP key exchange. Therefore, it's easy to apply to real VoIP environment that is not available to construct PKL.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4A
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• pp.377-384
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• 2006

According to fast VoIP technology development, more and more companies change voice network into IP based network among branch offices. IP PBX, which is deployed up to now, composed of IP phone and VoIP Gateway. Every telphone has replaced with If phone which support VoIP and VoIP gateway is installed in PBTN connection point to relay voice data. It can reduce the communication expense of International call, long distance call and call between a headquater and a trance because it uses internet line. In this paper, IP PBX is implemented that can distribute call using PBX network only usig personal terminal without Proxy Server. Depending on Role, terminal can be registered Master, Server and Client and it is verified in terms of performance and validation.

• 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.409-411
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• 2002

VoIP(Voice over IP)는 저렴한 비용과 다양한 응용서비스 기능으로 기존의 전화서비스를 보조 또는 대체할 서비스로 인식되고 있다. 특히 SIP(Session Initiation Protocol)는 텍스트 기반으로 다양한 애플리케이션과 쉽게 연동될 수 있다는 장점 때문에 차세대 VoIP프로토콜로 급성장하고 있으며 이를 이용한 VoIP 망 구성 및 부가서비스에 대한 관심이 증대되고 있다. 부가서비스는 콜의 발생 여부에 따라 사용자의 콜을 처리하여 서비스를 제공하는 형태와 서버가 직접 콜을 발생시켜 서비스를 제공하는 형태의 두 가지 유형으로 나눌 수 있다. 현재 SIP기반 VoIP에서는 사용자의 콜을 처리하는 서비스를 위한 CPL(Call Processing Language)이 표준화 중이며 이를 기반으로 여러 서비스도 구현되어 있다. 그러나 특정시간에 콜을 발생시켜 사용자에게 서비스를 제공하는 형태의 서비스는 논의되지 않고 있으며 구현된 사례도 아직 없는 상황이다. 따라서 본 논문에서는 이러한 유형의 새로운 서비스의 하나로 예약통화 시스템을 제안한다. 예약통화 시스템은 예약된 시간에 콜을 발생시켜 두 사용자간의 통화를 연결해 주는 예약 시간 기반의 서비스를 개발하는데 이용될 수 있다.

• Journal of the Korea Society for Simulation
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• v.10 no.2
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• pp.15-24
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• 2001

In this paper we study the delay patterns of voice traffic for end-to-end users Caused by serving the whole bunch of applications traffic at the same time on the Voice over Internet Protocol (VoIP) network. Given the current situation that voice traffic is served along with other application services on the VoIP network, it is quite necessary to figure out how and by what the voice traffic requiring high QoS is delayed. We compare the delay performance of voice traffic on the VoIP network under FIFO with the one under Weighted Fair Queuing(WFQ), and discover the differences of the delay performance resulting from the use of different voice codec algorithms. The results of our study show that using the voice codec algorithm with a higher coding rate nd the queuing algorithm of WEQ can provide users with high-quality voice traffic.

• The KIPS Transactions:PartC
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• v.15C no.4
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• pp.245-252
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• 2008

An emergency service over Voice over IP (VoIP) network is an essential condition, like the existing telecommunication services. To support for the emergency services, standardization works have been performed. The National Emergency Number Association (NENA) has been developing the framework and procedures for an emergency service for Non-IP based network, rather than protocols. In contrast, the Internet Engineering Task Force (IETF) has been only focused on end-to-end IP-based emergency calls. The NENA architecture is incompatible with the IETF protocols. To solve the problem, we design and implement a SIP-based VoIP emergency system by adopting the NENA architecture and by applying IETF protocols, for both IP-based Pubic Safety Answering Point (PSAP) and PSTN-based PSAP. It is implemented and tested under UNIX environment.