• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.4B
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• pp.411-423
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• 2004

This paper defines test methodologies and performance indices for SIP server system, and describes elements that can have influence on the test of SIP network equipments. Moreover, we implement a tester to evaluate the performance of SIP Servers such as Registrar and Proxy server. The performance indices for testing SIP servers are message processing rate, transaction delay, and call success probability. The parameters that can have an effect on the performance of SIP servers are user population, transport protocol, method of database access, method of DNS, call creation pattern, definition of transactions, and size of packets. We tested several SIP servers that act as Registrar, Proxy, and Redirect server using the implemented SIP tester, and, as a result, verified functions of the tester and performance indices and input parameters defined in this paper. Performance indices and methodologies presented in this paper can be used to evaluate SIP servers in NGN

• Journal of Advanced Navigation Technology
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• v.17 no.4
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• pp.421-428
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• 2013

Voice over IP refers to technology that enables routing of voice conversations over the Internet or a TCP/IP network. VoIP communication costs cheaper than traditional analog phone. Phone calls can be made to anywhere / anyone: Both to VoIP numbers as well as people with normal phone numbers. VoIP protocol equipment available today follows the SIP standard. Older VoIP equipment though would follow H 323, MGCP, Megaco/H.248. A SIP server is the main component of an IP PBX, dealing with the setup of all SIP calls in the TCP/IP network. A SIP server is also referred to a Asterisk IP-PBX. A VoIP telephone, also known as a SIP phone or a softphone, allows the user to make phone calls to any softphone, mobile or PC by using App store. A VoIP telephone can be a simple software-based softphone. However, the SIP Server and the program is vulnerable to VoIP attacks. In this paper, eavesdropping attacks tested by using the Asterisk SIP server. Eavesdropping attacks and TLS security methods apply to VoIP system. TLS can be applied to determine whether the eavesdropping available for VoIP Environments.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.04b
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• pp.1587-1590
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• 2002

차세대 네트워크 및 서비스를 위한 프로토콜로 IETF (Internet Engineering Task Force)의 SIP (Session Initiation Protocol)가 각광을 받고 있다. SIP는 PC, PDA, IP Phone과 같은 VoIP (Voice over IP) 단말간에 호 설정 프로토콜로 사용된다. SIP는 기본적으로는 양 단말간 호설정 프로토콜이지만 응용, 인터넷 단말기, 네트워크 장치에 구성요소로 구성할 수 있어 쉽게 적용 가능하기 때문에 모든 응용의 호설정 프로토콜로서 넓게 채택되어지고 있다. 그러나 SIP는 텍스트 기반 프로토콜로서 구현은 쉬우나 실제 표준에 맞게 구현하였는지는 판단하기가 어렵다. 따라서 구현된 SIP 프로토콜이 표준에 맞게 구현하였는지를 시험할 필요가 있다. 이를 해결하기 위해서, 본 논문에서는 SIP 시뮬레이터를 구현하였다. SIP 시뮬레이터는 구현된 SIP 제품을 인터넷상에서 시험할 수 있을 뿐만 아니라 시험 시나리오를 선택할 수 있고, 시험 과정을 그래픽하게 볼 수 있으며, 시험 결과를 확인할 수 있다. SIP 시뮬레이터는 사용자 인터페이스인 Testing User Agent와, 테스트 시나리오를 수행하는 Test Server로 구성된다. 사용자 인터페이스는 모든 플랫폼에 적용 가능한 Java를 사용하였으며, Test Server는 Linux 환경하에서 C++을 사용하여 구현하였다.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2004.07a
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• pp.161-161
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• 2004

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.7
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• pp.1854-1873
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• 2012

Owing to its simplicity and flexibility, the session initiation protocol (SIP) has been widely adopted as a major session-management protocol for Internet telephony or Voice-over IP (VoIP) services. However, SIP has faced various types of security threats. Call-disruption attacks are some of the most severe threats they face, and can greatly inconvenience consumers. In this paper, we analyze such SIP call-disruption attacks, and propose a method for detecting and counteracting them by extending the SIP INFO method with authentication. Using the proposed method, both the target user and the SIP server can detect the existence of a call-disruption attack on a user and counteract the attack. We demonstrate the effectiveness of the proposed method from the viewpoint of computational complexity by configuring a test-bed with an Asterisk SIP proxy server and an SIP performance (SIPp) emulator.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2004.11a
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• pp.180-180
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• 2004

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.4
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• pp.51-56
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• 2014

Thesedays, there are many smart device applications developed, especially on the using various sensors included in the smart device. Smart devices have several sensors which are camera, GPS, mike, and communication module for collecting ubiquitous environment, and many applications are developed by using such sensors. In this paper, we developed the multimedia stream architecture and examined the smart device applications based on open source with front and back-end server clouds for developing the conceptual architecture. Also, we examined the back-end distributed servers, realtime multimedia stream transferring, multi-media store, and media relay for other server and smart devices. We test the examined architecture on the real target environment to collect the SIP initial setup time, media stream delay, and end-to-end play time. The test results show that there have good network operation environment to provide realtime multimedia services, and we need to improve the end-to-end play time by minimizing the initial setup time.