• 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.

• 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.

• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.3
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• pp.309-316
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• 2016

Recently, the voice service over a mobile communication system tends to be provided based on the packet-based technology. Even though the sufficient transmission rate is supported by LTE mobile communication system, the quality of VoIP service that is experienced by the user can be degraded by the change in the transmission conditions and the terminal mobility. This paper has established an environment on which experiments are conducted for the different values of the major parameters that represent the transmission conditions. The result can contribute to the decision of the requirement that the mobile system should meet for maintaining the quality of VoIP service.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.862-865
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• 2002

VoIP was simply used to transfer voice call between IP host. VoIP provides efficiently end-to-end voice communication, however it needs to supply value-added services to users. This raper describes design of SIP communication for registering CPL.

• Proceedings of the Korea Information Processing Society Conference
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• 2009.11a
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• pp.447-448
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• 2009

VoIP 서비스에서 긴급 통화 서비스를 제공하기 위해서는 PSTN 망에 있는 소방서와 같은 긴급 서비스 센터에게 호를 연결한다. PSTN 망에 있는 긴급 서비스 센터를 통해 제공되는 VoIP 긴급통화 서비스는 VoIP 서비스가 가지고 있는 여러 멀티미디어 서비스를 사용할 수 없다. 이러한 문제를 해결하기 위해서는 모든 구성요소가 IP 망에서 제공되어야 한다. 본 논문에서는 모든 구성요소가 IP 망에서 제공되는 All-IP 기반 VoIP 긴급통화 서비스를 제공하기 위한 프로토타입을 구현하였다.

• The Journal of the Korea institute of electronic communication sciences
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• v.5 no.4
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• pp.385-390
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• 2010

VoIP Service is provided voice & image call using internetwork unlike traditional call. VoIP usage is becoming generalization & development. As a result, threats of security are steadily increasing. Regarding this situation, we will investigate the security problem of VoIP in various aspects. Also We will investigate main technology of security solution method for solve this problem.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.8
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• pp.1679-1684
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• 2004

This paper designs and implements a VoIP module for embedded system. Many other VoIP modules for desktop environment has widely implemented but the one for embedded system is not prevalent yet. The implemented VoIP module uses the OpenH.323 and is operated on the WinCE PDA. We use the TCP-friendly adaptation for congestion control. The experiment results show smooth communication but a little noise. Future works include the minimization of noise.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.12
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• pp.2555-2562
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• 2011

Binary CDMA is a new standard technology for wireless communication developed by our country that makes high speed communications and good quality of services. In this paper we researched the development methods and implementation of a wireless VoIP System based on the Binary-CDMA technology that makes it freely installed in any place without phone cables and laying works. We designed the structure of the Binary-CDMA Wireless VoIP System and defined messages and data structures for call control. And then we designed and implemented call control message flow between VoIP terminals for user's communication.

• The Journal of the Korea institute of electronic communication sciences
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• v.6 no.4
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• pp.493-498
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• 2011

In this paper, Transmission performance over MANET(Mobile Ad-hoc Networks) under DDoS Intrusions is evaluated. Intrusion counterplan requirement, which have to be used for MANET under DDoS intrusions, is suggested through this evaluation. VoIP simulator based on NS-2 network simulator is used for performance measurement. MOS, network delay, packet loss rate and call connection rate is measured with this simulation. Finally, requirement of intrusion continuing time shorter then 10 seconds is suggested for VoIP service over MANETs under DDoS intrusions.

• International Journal of Internet, Broadcasting and Communication
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• v.11 no.3
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• pp.59-63
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• 2019

VoIP technology is a technology for exchanging voice or video data through IP network. Various protocols are used for this technique, in particular, RTP(Real-time Transport Protocol) protocol is used to exchange voice data. In recent years, with the development of communication technology, there has been an increasing tendency of services such as "Kakao Voice Talk" to exchange voice and video data through IP network. Most of these services provide a service with security guarantee by a user authentication process and an encryption process. However, RTP protocol does not require encryption when transmitting data. Therefore, there is an exposition risk in the voice data using RTP protocol. We will present the risk of the situation where packets are sniffed in VoIP(Voice over IP) communication using RTP protocol. To this end, we configured a VoIP telephone network, applied our own sniffing tool, and analyzed the sniffed packets to show the risk that users' data could be exposed unprotected.