• Journal of Korea Multimedia Society
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• v.10 no.3
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• pp.335-347
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• 2007

VoIP (Voice over IP) is a technology to transport video and voice traffic over IP networks such as Internet. Today, the VoIP technology is viewed as the right choice for providing voice, video, and data communication among various terminals over the next generation network. This paper discusses a multipoint VoIP implementation with end-point mixing model which can support multipoint conference without a conference bridge. The multipoint VoIP is implemented with SIP (Session Initiation Protocol), and supports STUN (Simple Traversal of UDP Through NATs) since it works in an asymmetric NAT (Network Address Translator) environment. The characteristics of this paper are as follows. It is possible that all terminals in the hierarchical conference don't receive the duplicated media information because we use the end-point mixing model with the new media processing module. And, the paper solves the problem that the hierarchical conference session should be separated into several sessions when a mixing terminal terminates the hierarchical conference session.

• The KIPS Transactions:PartC
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• v.14C no.6
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• pp.481-492
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• 2007

The ubiquitous environment allows users to communicate with others and to have access to computing service any time, any place. Multipoint VoIP is expected to be one of the loading applications in this environment. The paper compares various conference models for efficient VoIP service in ubiquitous environment and shows why the endpoint mixing model performs effectively in this environment. However, the endpoint mixing model has its own drawbacks as well and we propose solutions to solve the problems.

• Proceedings of the IEEK Conference
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• 2009.05a
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• pp.16-18
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• 2009

VoIP(Voice over IP) is the technology to transport voice and video over IP networks such as Internet. Today, VoIP technology is viewed as the right choice for provide voice, video, and data communication over next generation network. We are sure that the multipoint VoIP will help enhancing the various application services in ubiquitous environment. The paper shows multipoint VoIP system implemented with end-point mixing model and introduces various embedded systems such as UFC(Ubiquitous Fashionable Computer), tourist guide terminal and industrial terminal which use the multipoint VoIP.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.8 no.8
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• pp.1640-1646
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• 2004

In this study, we propose the multiple VoIP service networks for P2P network environments based on portable internet. We found that voiceservice using portable internet is possible for real time services. After we designe the multiple VoIP service networks for P2P network environments based on portable internet, evaluate the performances. from the performance evaluations, we found satisfactory results and so we can expect the possibility of the portable internet services.

• Proceedings of the Korea Information Processing Society Conference
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• 2014.11a
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• pp.218-220
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• 2014

VoIP(Voice over Internet Protocol)는 음성 및 화상과 같은 멀티미디어 세션을 인터넷과 같은 IP 기반 네트워크를 통해 통신하는 기술이다. 최근에는 기존의 PC 시스템 이외에 이동통신기기와 다양한 무선네트워크 기반 휴대용 기기들의 보급으로 VoIP 의 사용량은 크게 증가하고 있다. 하지만, 무선네트워크는 그 특성과 환경적 요인으로 NAT 에서의 차단, 지연, 유실등과 같이 통신의 연속성을 보장해 주지 못하는 문제가 발생할 수 있다. 본 논문에서는 무선네트워크에서 통신할 때 발생할 수 있는 이런 문제들에 대응하는 해결 방안을 제시하고 RTP 미디어 재생의 연속성을 보장하는 ICE 기반 다자간 VoIP 시스템 설계와 구현모델에 대해서 기술하고자 한다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.176-179
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• 2003

The use of Multiparty Conference service based on SIP for VoIP provides is gradually magnified, and the work for continuous development and standardization on SIP is in the process of advancing. But, currently it is impossible for SIP to support identity discovery and distribution of each participant for multiparty conference. In this paper, we propose a SIP extension for guaranteeing security on the multiparty conference using SIP by adding new method and reconstructing header informations. With this, it is also possible to identify discovery and to distribute each participant using SIP extension when a call is established for closed multiparty conference.

• Proceedings of the Korea Information Processing Society Conference
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• 2013.11a
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• pp.257-258
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• 2013

VoIP(Voice over Internet Protocol)는 인터넷과 같은 IP 망에서 음성과 영상을 전송하기 위한 기술이며, 차세대 망에서의 음성, 테이터 통신을 위한 기술로 부상되고 있다. 따라서 VoIP 응용은 인터넷 망을 이용하는 다양한 단말기들 사이의 음성 및 영상 통신을 위하여 더욱더 많이 사용되어 질 것으로 예상된다. 본 논문에서는 VoIP 기술을 여러 분야에 적용할 수 있는 응용성과 실제 다자간 음성통신의 구현 방법에 대해서 기술하겠다.

• 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.

• Proceedings of the Korea Multimedia Society Conference
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• 2002.05c
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• pp.455-458
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• 2002

단말간 호설정 기능을 제공하는 SIP (Session Initiation Protocol)는 VoIP 표준 기술로 인식되고 있다. 그러나 SIP는 단대단 프로토콜이므로 새로운 응용으로 다자간 회의 시스템에 적용하기 위해서는 새로운 메소드 추가 및 새로운 헤더 추가 등이 요구된다. 특히, 멀티미디어 회의 시스템을 위해서는 오디오는 물론 비디오 관련 정보도 협상되어야 한다. 본 논문에서는 현재 IETF에서 진행되고 있는 SIP 기반 다자간 멀티미디어 회의 시스템을 분석한다.

• The Journal of the Korea institute of electronic communication sciences
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• v.15 no.5
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• pp.857-864
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• 2020

The multi-functional VoIP phone supports the Ethernet protocol in the TIPS(: Tactical IP Switch), which is one of the sub-systems of the tactical information and communication system (TICN). It provides secured voice / video calls in conjunction with VoIP exchanges and supports differential services such as multi-party calls and command functions. In this paper, improving methods have been proposed to reduce power supply defects in the field of multi-functional VoIP phones. The power supply part was improved by applying TVS of the output voltage inlet of the dedicated adapter of the multi-functional VoIP phone, TVS of the PoE module input, adding blocking diodes, and adding DC / DC converters behind the poly-switch. Also, functional and environmental tests were performed to verify the validity of the proposed methods.