• The Journal of the Acoustical Society of Korea
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• v.28 no.2E
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• pp.35-42
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• 2009

In this paper, we present criteria of objective measurement of speech quality to provide the mobile-VoIP services efficiently over wireless mobile internet. The mobile-VoIP service, which is based on mobility and is error-prone compared to conventional VoIP over wired network, is about to be launched, but there have not been adequate quality indexes and the Quality of Service (QoS) standards for evaluating speech quality of Mobile-VoIP. In addition, there are many factors influencing on the speech quality in packet network of which packet loss contribute directly to the overall voice communication quality. For this reason, we adopt the Gilbert-Elliot Channel Model for modeling packet network based on IP and assess the voice quality through the objective speech method of ITU-T P. 862 PESQ and ITU-T P. 862.1 MOS-LQO under various packet loss rates in the transmission channel environments. Our simulation results address the specific criteria and QoS for the mobile-VoIP services in terms of the various packet loss environments.

• 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.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2001.10a
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• pp.556-559
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• 2001

VoIP(Voice over IP) Technology, turn voice services over traditional telephone network into internet, is highlighted because of easy adopting the value added services related voice In this paper, we described the user agent system architecture for internet telephony services based on SIP (Session Initiation Protocol)

• 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.

• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.97-101
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• 2008

Voice over IP is a telephone service which sends and receives the voices through the Internet. Because the infrastructure of wireless and mobile communication networks such as 3G, Wi-Fi and WiMAX has expanded, the study about Mobile VoIP, which provides the voice service from wireless network, has been actively in progress. Since Rei 6 HSPA in 3GPP and Rev A lxEVDO in 3GPP2, VoIP through the data channel is more efficient than circuit switch. It is predicted that VoIP over 4G will be more effective and 4G mobile VoIP business will be vitalized in the future. In addition, there are businesses which offer VoWLAN by using software such as Skype and Fring. They provide services which cheapen the price of international calls and long distance calls. This paper will present the Korean and other countries' mobile VoIP trends, its classification along the network connection, the study on techniques, and conditions of mobile VoIP. It also will be described a view of terminal convergence and service convergence.

• 한국정보통신설비학회:학술대회논문집
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• 2009.08a
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• pp.91-96
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• 2009

In Public Switched Telephone Networks(PSTN) number portability is implemented by utilizing Intelligent Network(IN) functions for number mapping. And voice over IP(VoIP) and IP Multimedia Subsystem(IMS) networks can deploy number portability by using E.164 Number Mapping(ENUM). This paper discuss the possibility of using E.164 Number Mapping(ENUM) for number portability in voice over IP/IP Multimedia Subsystem and Public Switched Telephone Networks, eliminating the need for Number Portability Database(NPDB) for number portability routing data in Public Switched Telephone Networks.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2018.10a
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• pp.568-570
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• 2018

Currently, with rapid progress and supply of mobile communication technology, LTE(Long Term Evolution) technology is expanded and widely used to industrial and emergency communications beyond earlier smart-phone based service. In this paper, transmission performance of voice traffic, one of railway communication service based on LTE-R as an application field of LTE technology, is analyzed. This study is performed performance analysis with level of application service and consider effects of satisfaction level for users. Computer Simulation based on ns(Network Simulation)-3 is used for analysis and VoIP(Voice over Internet Protocol) specification is used for voice traffics. Results of this paper is used to implement LTE-R networks and develope application services over LTE-R network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.8B
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• pp.683-690
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• 2003

In this paper we present a theoretical framework for the network design with delay QoS guarantee to a voice at the packet level. Especially, we propose a method for estimating the bandwidth at the ingress edge routers accommodating the voice connections and data sessions in the next-generation If network. First, we describe network architecture for VoIP (Voice over IP) services in the NGN (Next Generation Network). After that, we propose a procedure for dimensioning the bandwidth at the output port of a router that accommodates voice and data traffic using the non-preemptive queuing system with strict priority service scheme. Via numerical experiments we illustrate the implication of the proposition.

• Venture DIGEST
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• s.103
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• pp.8-11
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• 2007

인터넷전화(VoIP)는 "Voice over Internet Protocol"의 약자로 기존의 회선교환망(Circuit Network)이 아닌 인터넷망(IP Network)을 통해 패킷단위로 전송하여 통화권 구분없이 음성등을 송신하거나 수신하는 새로운 방식의 전화서비스이다.

• The Transactions of the Korea Information Processing Society
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• v.7 no.11S
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• pp.3668-3674
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• 2000

Recently Technology regarding VoIP (Voice over IP) is emerging over the market of the IP network. So far nothing is unfortunately there any attempt to try any research with respect to the development of the protocol stack relating to such control of gateway as MGCP, MEGACO, SIP, SDP. The reasons come from the low level of infrastructue, the shortage of the time and technology required at the moment, and so on. In this regards, this paper is focused on developing a protocol stack made with encoder/decoder, the generator of the header file etc, based on the protocol grammars of MGCP, SDP supported by IETF. For the sake of it, first develops the syntax analyzer, encoder/decoder, header file generator for encoding/decoding as applying the method of syntax-directed to each protocol grammar.