• IE interfaces
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• v.15 no.4
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• pp.391-400
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• 2002

The internet call systems using CTI(Computer Telephony Integration) functions are designed with system configuration, DFD(Data Flow Diagram) and ERD(Entity Relationship Diagram) in this paper. The internet call systems are constructed to cooperate with conventional CTI call center. The internet phone calls occurred from the web browser of customer can be connected throughout VoIP gateway and PBX to many counselors. The internet call systems can provide various services; customer information service, escorted browsing service, text chatting service, text sharing service, conference service, and statistical analysis service.

• The Journal of the Korea Contents Association
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• v.4 no.4
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• pp.123-131
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• 2004

Internet is absolutely contributed to information technology revolution nowadays. Internet services such as voice and data, etc. are provided home or small office via home gateway. The development of communication equipment via home gateway is implemented rapidly, and its product various. This paper presents the design and implementation of the VoIP equipment including the telephone function based on the embedded environment and being connected to the home gateway and the PC because of taking 2-ethernet LAN ports. As developing environment, the STLC1502 developed at ST Microelectronics as single chip solution, VxWorks as RTOS, and C language as coding mechanism are used. The verification of the developed systems for the voice test is carried out for the gatekeeper via Internet. The performance parameter is considered as the call processing capacity measuring the time of the call setup and clearing, and the data processing capacity for the file transfer. As a call setup and clearing is about 95ms, the call processing capacity is about 10 calls per second. The data processing capacity is 5.7Mbps in case of file transfer of server and client environment. Therefore the performance result is satisfied in the aspect of the call processing time and the data transfer time in Internet.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4A
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• pp.377-384
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• 2006

According to fast VoIP technology development, more and more companies change voice network into IP based network among branch offices. IP PBX, which is deployed up to now, composed of IP phone and VoIP Gateway. Every telphone has replaced with If phone which support VoIP and VoIP gateway is installed in PBTN connection point to relay voice data. It can reduce the communication expense of International call, long distance call and call between a headquater and a trance because it uses internet line. In this paper, IP PBX is implemented that can distribute call using PBX network only usig personal terminal without Proxy Server. Depending on Role, terminal can be registered Master, Server and Client and it is verified in terms of performance and validation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.4B
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• pp.593-603
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• 2000

In this paper, a dynamic route control scheme to increase speed of data communication over public network is presented. And a new structure of gateway which is suitable for the public network environments is proposed and analyzed. For dynamic allocation of transmission routing resources of the gateway and improvement of call processing ratio of the system, virtual queue is used. As results, route utilization ratio of the proposed scheme can be improved up to 60% as compared to the conventional fixed route control scheme. For the case of limited number of routes, it provides high speed data communication while maintaining delay time as low as possible. In addition, as the allocation period at call access node can be varied with the traffic in the proposed scheme, various traffics can be applied easily. Thus, it can be expected that the presented dynamic route control gateway may be suitable for application in high speed data communication over public network.

• MALSORI
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• no.43
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• pp.113-125
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• 2002

As there are needs to search the Web information via wire or wireless telephones, VoiceXML forum was established to develop and promote the Voice eXtensible Markup Language (VoiceXML). VoiceXML simplifies the creation of personalized interactive voice response services on the Web, and allows voice and phone access to information on Web sites, call center databases. Also, it can utilize the Web-based technologies, such as CGI(Common Gateway Interface) scripts. In this paper, we have developed the voice portal service platform based on VoiceXML called TeleGateway. It enables integration of voice services with data services using the Automatic Speech Recognition (ASR) and Text-To-Speech (TTS) engines. Also, we have showed the various services on voice portal services.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.15 no.4
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• pp.99-104
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• 2015

Service platform which can offer various multimedia communication as the video, audio, voice and data is IMS(IP Multi-Media Subsystem). It is effective in the company is introducing only such convergence IMS services to be required to provide various multimedia services at the lowest cost and existing communication environment while keeping the maximum Therefore, in this study, we had developed IMS 코어 main system that not more than 1,000 employees of companies can effectively establish IMS solutions. This system is located at the middle between IMS terminal and CSCF(Call Session Control Function) in line with IMS services and provides CSCF in response to the IMS terminal and IMS terminal in response to the CSCF. As well, corded telephone and SIP phone which were used as terminal is linked with gateway.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.11
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• pp.2381-2387
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• 1997

The Communication Processing Systems(CPS) is a gateway that supports 01410 hitel service by connecting the public switched telephone network(PSTN) with the public switched data network(PSDN). The CPS is made the meausred rate accounting method for the biling after extracting the originating numbers from the nearest local PSTN. In this paper, a numerical simulation is carried out on the offered traffic per user on the telephone access part of the CPS according to change of call loss probability and also propose the proper number of channels based on the simulation results.

• ETRI Journal
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• v.10 no.2
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• pp.13-19
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• 1988

This paper describes on the interworking services. interworking issues and the design concepts in the Interworking System (IWS) between Pubilc Switched Telephone Network (PSTN) and Packet Switched Data Network (PSDN), which has been developed at the Electronics and Tedecommunications Research Institute (ETRI). As a gateway between PSTN and PSDN, IWS provides the interconnection between PSTN and PSDN. In addition to the provision of basic interconnection services (i.e. dial-in/dial-out services), IWS provides the Maintenance and Administration (M&A) services for system itself and the supplementary services such as the call barring, direct call and closed user group. The architecture of IWS is based on the multiprocessor system with single system bus. And a multiprocessor real time kernel is used to support the inter-board communication as well as the inter-process commmunications. Especially, the operation and maintenance functions of IWS are treated in detail.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.5B
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• pp.809-819
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• 2000

This paper deals with an optimal capacity allocation problem and performance analysis in Advanced Information Communication Processing System(AICPS). AICPS is a gateway system interconnection PSTN(Public Switched Telephone Network), ISDN(Intergrated Services Digital Network), PSDN(Packet Switched Data Network), internet, Frame Relay and ATM together. This study considers not only ISDN and Internet but also user behavior of On-line service which is analyzed by Markov process. A call blocking probability of TNAS and INAS is computed by Erlang's formula. Then, PNAS and WNAS's call blocking probability are computed by Stochastic knapsack modeling. The result is compared with result of simulation. Finally, we allocate an optimal capacity minimizing total call blocking probability.

• Journal of Korea Society of Digital Industry and Information Management
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• v.11 no.1
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• pp.19-25
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• 2015

In this time, as an industrial society developed to a welfare society, more and more people want their quality of lives upgraded and specially customized service by mass production/mass consumption. And it leads to an increase of requiring service. There're active developments and studies on various IT equipment as the requirement made IT devices used in service. In this paper, we try to design and realize radio paging DID by using smart device to be used as a receiver of a radio pager which is broadly used at face to face service. Firstly, we used MCU to design and implement Wireless Calling Gateway which change radio calling signal of ISM band to smart device. A receiver of wireless caller used original receiver module. Also it used bluetooth module to communicate with smart device. It was possible to have satisfactory communication since radio paging signal converter and smart device were linked in 3M. Secondly, to indicate various paging information delivered from a radio pager, we realized DID application program by using Smart PAD. As a result, we could indicate various information compared to an original receiver which only could indicate letters or numeric data. Secondly, we implemented the DID app for wireless calls that can display a variety of information sent from a wireless pager. Was implemented using the Smart Pad. As a result, it is shown that can display a variety of information than the existing receiver.