• Journal of Advanced Navigation Technology
• /
• v.17 no.4
• /
• pp.421-428
• /
• 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 Korean Institute of Information and Commucation Sciences Conference
• /
• 2010.05a
• /
• pp.719-721
• /
• 2010

VoIP over MANET is an important IP phone technique used in mobile ad-hoc networks. VoIP system over wired networks is based on SIP, SIP UA and SIP server which process the users invite, register and so on. Since no base stations in MANETs to perform as SIP servers, VoIP service on MANETs is being done with tiny SIP server which is done on each node. In this paper, VoIP system is implemented with pseudo SIP server which can process 1 call based on modification of standard SIP server which is used on communication environment such as wired networks. The propesed system is performed well and compatible with SIP and SIP UA.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.40 no.12
• /
• pp.18-26
• /
• 2003

In this paper, we design and implement a QoS control function in the SIP-based VoIP system. As a network infrastructure for VoIP service, we select the Intserv over Diffserv architecture where the network resources are managed by a call admission control mechanism. The SIP protocol extended to support QoS signaling procedure is modulized to operate independently with the infrastructure. The performance of the QoS-enabled VoIP system is verified by experiments.

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

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.7 no.1
• /
• pp.75-82
• /
• 2003

We can foresee the rapidly growing of users by development of VoIP service what can transmit a audio traffic with low cost among a lots application using internet. VoIP needs a standardized protocol that is able to do signaling for offering high quality of services such as mobility, universal number, multiparty conference, voice mail, automatic call distribution. At the present time, a base composition elements of SIP(Session Intiation Protocol) are developing for offering VoIP based SIP in the inside and outside of the country, because SIP of IETF which has a strength from 'fast connection', 'parsing' & 'easy to compile' points of view. This paper suggests a type of Hybrid SIP Server for providing some services as 'a reducing load of SIP server that process a request method from users', 'efficiency of managing networks', 'offering services to many users'.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.29 no.9B
• /
• pp.799-807
• /
• 2004

In this paper, a security mechanism for a VoIP system based on SIP was implemented. This was satisfied sec security requirement of RFC 3261. The SIP standard proposes a HTTP digest authentication for user authentication mechanism, TLS for hop-by-hop security and S/MIME for end-to-end security. SRTP draft was implemented for media security. We also analyzed security of proposed SIP standard.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.27 no.9C
• /
• pp.842-852
• /
• 2002

The fast advance in the VoIP technologies gives a rich opportunity to create different kind of VoIP applications such as IP telephony services. The application level call signaling protocols such as ITU-T H.323 and IETF SIP provide the communication functions of end-to-end call setup and release. Currently, there is a lot of H.323 based VoIP products in the market, however SIP is considered as a suitable protocol for supporting applications in IP environments, so SIP based VoIP products and services begin to appear. In this paper, firstly we present the characteristics of some possible SIP based applications and describe the design and implementation of a VoIP example service named PC-to-PC Internet telephony service using the developed SIP network components. The PC-to-PC Internet telephony service and User Agent are developed in MS window 98/2000 using visual C/C++, and Proxy server and Registrar in Linux 7.0 using C, respectively.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.12 no.6
• /
• pp.1022-1025
• /
• 2008

Session Initiation Protocol (SIP) has become the call control protocol of choice for Voice over IP (VoIP) networks because of its open and extensible nature. However, the integrity of call signaling between sites is of utmost importance, and SIP is vulnerable to attackers when left unprotected. Currently a herby-hop security model is prevalent, wherein intermediaries forward a request towards the destination user agent sewer (UAS) without a user agent client (UAC) knowing whether or not the intermediary behaved in a trusted manner. This paper presents an integrated security model for SIP-based VoIP network by combining hop-by-hop security and end-to-end security.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.34 no.6B
• /
• pp.575-584
• /
• 2009

In this paper, we design and implement a VoIP traffic generator for simulating call processing in IP contact center systems. Creating a VoIP call based on H.323 and SIP and generating RTP traffic which uses G.711 codec, the generator lets many users simulate situations on which they call each other. With this tool, which is named VoIPTG, users can combine H.323 or SIP session control protocol, the number of users, time variation, and voice codecs and then direct various situations for simulation. This traffic generator can be used for testing functions of an IP contact center and especially it is necessary for testing the quality of IP based call recording systems.

• Proceedings of the Korea Institutes of Information Security and Cryptology Conference
• /
• 2006.06a
• /
• pp.471-474
• /
• 2006

기존 전화에 비해 저렴한 가격으로 서비스의 제공이 가능한 VoIP 서비스의 증가는 SIP 스팸이라는 역기능을 낳았다. SIP은 표준 VoIP 프로토콜로 현재 SIP 기반의 VoIP 서비스의 개발이 활발하게 진행 중에 있다. 이에 본 논문은 SIP 기반 VoIP 환경에서의 스팸 유형을 살펴본 후 이를 해결하기 위한 스팸 대응 기술로 기존의 이메일 스팸 대응 기술을 비교, 분석한다. 또한 이메일 스팸 대응 기술을 기반으로 제안된 현재 SIP 스팸 대응 기술을 알아보고 앞서 분석한 대응 기술의 SIP 기반 VoIP 환경에서의 적용 가능 여부를 생각하여 가장 적합한 스팸 대응 기술을 제시한다.