• 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 Korean Institute of Information and Commucation Sciences Conference
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• 2010.05a
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• pp.719-721
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• 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 Korea Institute of Information and Communication Engineering
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• v.7 no.1
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• pp.75-82
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• 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'.

• Journal of Internet Computing and Services
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• v.9 no.6
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• pp.37-48
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• 2008

VoIP service is a transmission of voice data using SIP protocol on IP based network, The SIP protocol has many advantages such as providing IP based voice communication and multimedia service with cheap communication cost and so on. Therefore the SIP protocol spread out very quickly. But, SIP protocol exposes new forms of vulnerabilities on malicious attacks such as Message Flooding attack and protocol parsing attack. And it also suffers threats from many existing vulnerabilities like on IP based protocol. In this paper, we propose a new Virtual Proxy Server system in front of the existed Proxy Server for anomaly detection of SIP attack and stateful management of SIP session with enhanced security. Based on stateful virtual proxy server, out solution shows promising SIP Message Flooding attack verification and detection performance with minimized latency on SIP packet transmission.

• Proceedings of the Korean Information Science Society Conference
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• 2004.10c
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• pp.331-333
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• 2004

본 논문에서는 SIP(Session initiation Protocol) 기반의 VoIP(Voice over Internet Protocol) 시스템에 효율적인 call-control을 위해 필요한 헤더와 파라미터를 추가한 Extension SIP를 제안하였다. 또한 이 제시된 Extension SIP에 따르는 SIP Proxy Server와 User Agent(User Agent Client, User Agent Server)를 리눅스 시스템에서 C언어를 통해 구현하였고, 이 구현된 Extension SIP 시스템을 통해 기존의 SIP 시스템과 cail-control을 위한 packet traffic을 비교.분석 함으로써 제안한 Extension SIP의 효율성 을 증명하였다.

• The Journal of the Korea Contents Association
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• v.7 no.3
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• pp.17-24
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• 2007

SIP-based tightly coupled conference, which has a centrally located conference server for controlling and management, can be classified several models according to location of focus and mixer. These are centralized server model, endpoint server model, media server component model and distributed mixing model. However each model has its strength and weakness. In this paper, we propose and implement a SIP-based effective hybrid-type conference model which decreases amount of SIP signaling messages, lowers load of server media mixer, and can be easily expandable to large scale conference. In this model, when the number of participants exceeds a pre-defined limit, the conference server selects some participants which posses specific functions and let them share functions of notifications of conference state event package and media mixing. When each participant subscribes conference state event package to the server, it can indicates its possession of such functions by a specific header message. The server stores the indication to the conference information database, and later uses it to select participants for load sharing. The performance of our proposed model is evaluated by experiments.

• 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 the Korea Convergence Society
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• v.9 no.10
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• pp.69-74
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• 2018

IP and SIP public broadcasting systems developed in Korea and abroad are developed in a Windows or Linux server environments and are installed in a server-rack structure, have high power consumption, and are difficult to remotely respond to system failures. In this paper, IoT platform is designed to connect IoT device and gateway to IoT service server by using internet service structure. We also designed a server based on embedded OS that can provide a variety of public safety management services according to the order of the server with built-in call processing and broadcasting function that can handle emergency calls and emergency broadcasts in public places using this structure. This server is interoperable with a variety of SIP-based call and broadcast devices that support the standard SIP and can be integrated with an in-house phone and on-premises system.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.146-149
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• 2001

In order to provide VoIP service that using SIP on the Internet, users must register their current location to receive call from other cal]or. 13y using REGISTER method, users register his or her SIP URL address and currently contactable addresses to registrar server. The registered addresses are not only SIP URI but also mai]to, http, tel, and so on. In these procedures, we must consider authentication mechanism whether he/she has authority to handle the record or not. In this paper, we will briefly describe authentication mechanism that used in SIP and design of registrar server that support the 3rd party registration.

• The Journal of the Korea Contents Association
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• v.6 no.2
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• pp.17-26
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• 2006

SIP(Session Initiation Protocol)-based centralized conference service model has advantage of easiness in conference management and service as compared to other models. However when media server which is one of the components of conference server is included in the conference server, this model shows disadvantage of high server work load with increasing numbers of conferences and participants. In this paper, to improve this problem, we have suggested and implemented a new conference service model in which an UA(User Agent) who first make the conference acts as a media server instead of conventional conference server and, the conference server with conference event package takes only part of management of the conference and its participants. Therefore, many more conferences can be held and managed compared to the conventional centralized conference service model because load of the conference server decreases in our suggested model md, furthermore the participants needs only make SIP call connections to the UA who first make the conference for establishing media session.