• Proceedings of the Korea Information Processing Society Conference
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• 2011.11a
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• pp.626-628
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• 2011

인터넷이 발전함에 따라 기존의 PSTN(Public Switch Telephone Network)망이 감소하고 VoIP 서비스가 증가하고 있다. VoIP 서비스가 기존의 인터넷을 기반으로 서비스가 되어 보안문제까지 같이 떠안게 되었다. 이에 VoIP상의 다양한 공격에 대한 분석 및 효율적인 탐지 방법이 연구 되고 있다. 본 연구에서는 공격 중에서 SIP 상에서의 INVITE Flooding 공격에 대해 분석하고, 기존의 탐지 알고리즘을 연구하여 오탐율이 개선된 탐지 알고리즘을 제안한다.

• Journal of KIISE:Computing Practices and Letters
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• v.15 no.8
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• pp.582-585
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• 2009

VoIP technology for transmitting voice over IP network such as packet-based network has a lot of benefits by integrating services and reducing costs. The network is different from PSTN-based communications in some aspect such as transmitting not only voice but also text, image, multimedia data. In addition, portable terminals like a mobile phone, and ubiquitous communicator can easily access the internet for VoIP. Therefore, To prevent illegal users, offering certificate services is necessary, This study proposes a solution of user certification for a VoIP environment.

• Journal of Korea Society of Digital Industry and Information Management
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• v.14 no.4
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• pp.117-125
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• 2018

Recently, VoIP technology is widely used in our daily life. Even VoIP has become a technology that can be easily accessed from services such as home phone as well as KakaoTalk.[1] Most of these Internet telephones use the RTP protocol. However, there is a vulnerability that the audio data of users can be intercepted through packet sniffing in the RTP protocol. So we want to create a tool to check the security level of a VoIP network using the RTP protocol. To do so, we capture data packet from and to these VoIP networks. For this purpose, we first configure a virtual VoIP network using Raspberry Pi and show the security vulnerability by applying our developed sniffing tool to the VoIP network. We will then analyze the captured packets and extract meaningful information from the analyzed data using the Google Speech API. Finally, we will address the causes of these vulnerabilities and possible solutions to address them.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.10
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• pp.1994-2005
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• 2015

Voice over IP (VoIP) is a technology for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks. Instead of being transmitted over a circuit-switched network, however, the digital information is packetized, and transmission occurs in the form of IP packets over a packet-switched network which consist of several layers of computers. VoIP Service that used the various techniques has many advantages such as a voice Service, multimedia and additional service with cheap cost and so on. But the various frauds arises using VoIP because VoIP has the existing vulnerabilities at the Internet and based on complex technologies, which in turn, involve different components, protocols, and interfaces. According to research results, during in 2012, 46 % of fraud calls being made in VoIP. The revenue loss is considerable by fraud call. Among we will analyze for Toll Bypass Fraud by the SIM Box that occurs mainly on the international call, and propose the measures that can detect. Typically, proposed solutions to detect Toll Bypass fraud used DPI(Deep Packet Inspection) based on a variety of detection methods that using the Signature or statistical information, but Fraudster has used a number of countermeasures to avoid it as well. Particularly a Fraudster used countermeasure that encrypt VoIP Call Setup/Termination of SIP Signal or voice and both. This paper proposes the solution that is identifying equipment of Toll Bypass fraud using those countermeasures. Through feature of Voice traffic analysis, to detect involved equipment, and those behavior analysis to identifying SIM Box or Service Sever of VoIP Service Providers.

• Journal of Internet Computing and Services
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• v.10 no.6
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• pp.27-36
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• 2009

PSTN (public switched telephone network) provides voice communication service, whereas IP network provides data oriented service, and we can use IP network for multimedia transport service (e.g. voice over IP service) with economic price. In this paper, we propose RoIP (robot on IP) service scenario, signaling call flow, and implementation to provide home automation and monitoring service for remote site users. In our scheme, we used a extended SIP (session initiation protocol) for signaling protocol between remote site users and home robots. For our bearer transport control, we implemented H.263 video codec over RTP (real-time transport protocol) and additionally DTMF (dual tone multi-frequency) transport for robot actuator control. We implemented our scheme on home robots and experimented with KTF operator network, and it shows good communication quality (average MOS = 9.15) and flexible robot controls.

• Journal of the Korea Computer Industry Society
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• v.5 no.9
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• pp.1017-1024
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• 2004

In this paper, we propose the progressed interworking method between H.323 and SlP, then explain the improved property. The VolP(Voice over Internet Protocol) technology which is able to use a voice service through internet is more cheaper then existing telephone charges, and is easil)· accept the various of multimedia services from internet. Previous connectionmethod of VoIP used H.323 protocol, but it is very complex to connection establishment. so, the SIP(Session Initiation Protocol) protocol that propose in SIP-Working Group is in use recently. Therefore, we need new interworking methodology between H.323 and SIP Products. In this thesis, the progress interworking method between H.323 and SIP are Propose, then interpret unnecessary packet delay for call setup and improved feature of message exchange.

• Proceedings of the Korean Information Science Society Conference
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• 2001.10c
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• pp.601-603
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• 2001

기존의 전화망과 달리 VoIP(Voice over IP)기술은 저렴한 통화 서비스를 가능하게 해주며, 다양한 멀티미디어 서비스를 제공해 줄 수 있다는 장점을 가지고 있어 현재 이와 관련한 개발이 활발히 진행 중에 있다. 그런데, 대부분의 VoIP시스템이 ITU-T에서 제안한 H.323을 표준으로 삼아 구현되었으나, H.323은 랜(LAN)환경에서 개발된 기술 방식이므로 이에 따른 구조적인 한계점을 가지고 있다. 반면, IETF에서 제안만 SIP(Session Initiation Protocol)의 경우는 내용이 간단하고 구현이 쉬울 뿐 아니라 확장성과 포괄성 측면에서 구조적인 문제점을 해결할 수 있는 대안으로 제시되어 지고 있다. 본 논문에서는 SIP을 기반으로 한 User Agent와 부재중 음성 메일 서비스의 설계 및 구현에 대해 설명한다. User agent간에는 실시간 음성 및 화상 통화가 가능하며 상대방이 부재중일 경우에는 상대에게 음성 메일을 전송할 수 있다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.2
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• pp.331-337
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• 2004

Multiparty conference service based on SIP supported by VoIP network is gradually increased in use and the continuous development and standardization works on SIP are in the process of advancing. But SIP used in currently does not support identity discovery and distribution of each participant for multipath conference. In this paper, a SIP extension method for guaranteeing security from the multiparty conference based on SIP is proposed. We design a new SIP header and method for discovering and distributing a participant's identity in closed multiparty conference when the call initiation is established. And it can ensure that each participant is notified before a new participant joins.

• Journal of KIISE:Computing Practices and Letters
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• v.11 no.4
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• pp.312-333
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• 2005

SIP(Session Initiation Protocol) has been proposed for session control protocol of Internet multimedia communication system like VoIP(Voice over IP). SIP has complicated session control steps to support various kinds of audio and video formats and to assure service quality of real time data communication. Up until now, existing protocol analyzers can not provide such detailed information of SIP based communication system. In this paper, therefore, we propose a new protocol analyzer as a tool that can analyze and diagnose SIP based multimedia communication system throughout the session initiation, data exchange and session change steps. The propose traffic analyzer, which is called STAT(SIP based Traffic Analysis Tool), Is implemented on Winder's environment so that it is generally usable and extensible. Since STAT analyze low level packets captured via Ethernet broadcasting property, it is able to provide session status and real time traffic monitoring information without any affection to the communication system. The STAT which is implemented in this paper. therefore, is expected to be a useful tool for developing and managing of a SIP based multimedia communication system.