• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• v.9 no.2
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• pp.947-950
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• 2005

Voice of IP(VoIP) technology provides voice service as well as data service via Internet. It has been a promising technology as Internet grows fast and the requirements are increasing. Recently, serveral protocols have been created to allow telephone calls to be made over IP networks, notably, SIP and H.323. Due to introducing SIP and H.323, In this paper, we consideration interoperability of internet telephony service between ITSPs(internet telephony service provider)based on H.323.

• Korean Journal of Human Ecology
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• v.15 no.6
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• pp.961-971
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• 2006

The purpose of this study was firstly to provide fundamental data on the activation for the IP-based video phone for the special consumer related to the physically handicapped; secondly to inform empirical data for the consumer public policy in the information technology market, specially for the deaf people. The results of study showed that consumer needs extend to not only simple voice communication for general consumers but also special demands for both the handicapped and the elderly. This study also indicated that VoIP's characteristics of technology would be easily applied to the TRS or VRS which can be adapted to the special consumer market so that VoIP service would be optimal technology for the special consumers like the deaf. In order to successfully implement TRS & VRS business, the paper proposed as follows; 1) the provision of VoIP service enable to satisfying consumers in special market such as the deaf market and the elderly market, 2) the necessity of supporting policy by the related law, and 3) the construction of the system inducing interests from the market participants.

• Proceedings of the Korea Information Processing Society Conference
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• 2006.05a
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• pp.1243-1246
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• 2006

초고속 통신망의 확대 적용으로 인터넷의 빠른 성장과 함께 음성과 비디오 그리고 데이터를 통합하고자 하는 노력이 시도되고 있다. VoIP(Voice over IP)는 IP를 이용하여 음성과 데이터를 패킷 형태로 통합하여 실시간으로 전송하는 기술이다[1]. 패킷 네트워크에서 VoIP 시그널링 기술을 이용하면 망 자원으 효율적 이용 및 PSTN에 가까운 음질 그리고 인터넷과 연계한 다양한 음성서비스 지원이 가능하다. 콜센터에서도 VoIP를 사용하게 됨에 따라 VoIP망에서의 녹취 시스템이 필요하다. VoIP 녹취 시스템은 상담원과 고객 간의 통화 내용을 자동으로 녹음하여 보관함으로써 고객의 요구사항을 명확하게 파악할 수 있으며 녹취 데이터의 통계 자료 제공으로 효율적인 관리가 가능하고, 선택 녹취, 스케쥴링 녹취, 상담원의 평가 자료를 제공하여 고객 관리의 질적인 향상을 지원한다. 본 논문에서는 성능에 큰 영향을 주지 않고 기존의 VoIP 녹취 시스템의 문제점을 해결한 에이전트를 포함한 VoIP 녹취 시스템을 제안한다.

• Proceedings of the KIEE Conference
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• 2001.11c
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• pp.130-133
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• 2001

VoIP(voice over internet protocol) technology is based on IP protocol. The IP protocol can be involved in two types of communication: unicasting and multicasting. Unicasting is the communication between one sender and one receiver. It is one-to-one communication. Multicasting is one-to-many communication. So that, many receivers can get same data from one sender simultaneously. and, the different protocol are proposed for VoIP; H.323, SIP and MGCP. MGCP is perfect server-client protocol, so MGCP is very attractive VoIP protocol to ISP. This paper uses MGCP and offers modified MGCP for conference call. So that, Modified MGCP is compatible to MGCP, and supports conference call using IP-multicast.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2002.11a
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• pp.758-761
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• 2002

Voice of U(VoIP) technology Provides voice service as well as data service via Internet. It has been a promising technology as Internet grows fast and the requirements are increasing. Recently, serveral protocols have been created to allow telephone calls to be made over IP networks, notably, SIP and H.323. Due to introducing SIP and H.323, There are many change at internet telephony service. Internet telephony enables a wealth of new service possibility Users can control telephony service directly. In this paper, we design and implementation CPL client based on SIP system.

• The KIPS Transactions:PartC
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• v.17C no.3
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• pp.299-306
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• 2010

IPTV and VoIP services are considered as killer applications over Mobile WiMAX network, which provideshigh mobility and data rate. Among those which affect the quality of voice in those services, the jitter buffer or playout buffer can compensate the poor voice quality caused by the packet drop due to frequent route change or differences among routes between service endpoints. In this paper, we analyze the correlation between the sleep interval length and jitter buffer size in order to guarantee a predefined level of voice quality. For this purpose, we present an end-to-end delay model considering additional delay incurred by the WiMAX PSC-II sleep mode and a VoIP service quality requirement based on the delay constraints. Through extensive simulation experiments, we also show that the increase of jitter buffer size may degrade the voice quality since it can introduce additional packet drop in the jitter buffer under WiMAX power saving mode.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.3B
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• pp.398-407
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• 2010

Due to increase of user requirement for various traffics and the advance of network technology, each distinct network has converge into FMC(Fixed Mobile Convergence) networks. However, we need to research the performance analysis of VoIP(Voice over Internet Protocol) in the FMC network to provide QoE for the voice user of FMC network. Therefore, this paper introduces the scenario which is the situation of voice quality degradation when a user uses VoIP to communicate with other users in the FMC network. Especially, this paper presents scenario in terms of the component of the network and finds the improvement point of voice quality. In the simulation results, three improvement points of voice quality are found as following: voice quality degradation by packet loss in the physical layer of the HSDPA network, by utilizing GGSN without QoS parameter mapping mechanism which is gateway between 3GPP and IP backbone, and by using non-QoS AP in the WLAN network.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.281-284
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• 2011

Smart phone is being used in the job as the ubiquitous society will Without being restricted by the time and place and devices. The rapid increase in the use of smart phones has brought the activation of the mobile job. And government agencies have brought in the transition to a smart society. In this paper, using a Voice over Internet protocol(VoIP) service for your smart phones to enhance security is the study of encryption technologies. External and internal signals, and call encryption and security standards of administrative agencies is the study of VoIP. Smart phone VoIP service is a study that security of equipment certificate, the internal signal and call encryption. This paper will contribute what using smart phone VoIP security and usability In smart generation.

• Journal of Korea Society of Digital Industry and Information Management
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• v.7 no.2
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• pp.39-46
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• 2011

Privacy IP hacking problems such as invasion of privacy, password cracking, voice wiretapping and internet over charged occurred, because VoIP internet voice phone service gradually spread. This thesis attempted to attack the VoIP service network by application. First use application to spoof IP address then attempted wiretap the VoIP service and sends a lot of messages to disturb service movement. At this point, we connected VoIP soft terminal, so we can operate real-time filtering operator to block the SIP Flooding offence by monitor the traffic and detect the location where it got attacked. This thesis used experiment to prove it is possible to detect the offence and defend from SIP Flooding offence.

• Proceedings of the Korea Contents Association Conference
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• 2006.11a
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• pp.75-79
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• 2006

In a VoIP(Voice over IP) system which support the realtime speech service, speech quality is deteriorated by the delay, the jitter, the loss, and the reversed packet order. In this thesis, APBC for receiver site is proposed, which compensate the jitter by the adaptive playout algorithm and conceal the packet loss, and align the packet order. Also, a VoIP application system is implemented, and the performance of APBC is verified on the implemented system by measuring the processing speed and the speech quality. From the result, processing speed is 257$\mu$sec, which is fast enough to deal with packet being received in realtime. Also, the speech quality by MOS(Mean Opinion Score) is improved as 18 percent compared with algorithm of fixed playout delay.