• Journal of Advanced Marine Engineering and Technology
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• v.27 no.1
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• pp.65-75
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• 2003

The internet telephony service is one of the successful internet application services. VoIP is the key technology for the service to come true. VoIP uses H.323 or SIP as the standard protocol for the distributed multimedia services over the internet environment, in which QoS is not guaranteed. VoIP carries the packetized voice by using the RTP/UDP/IP protocol stack. The UDP-based internet services cause the data transmission problem to the users behind the internet firewall. So does the internet telephony service. The users are not able to listen the voices of the counter-parts on the public internet or PSTN. It makes the problem more difficult that the internet telephony service addressed in this paper uses only one UDP port number to send the voice data of all sessions from gateway to terminal node. In this paper, two schemes including the usage of dummy UDP datagrams, and the protocol conversion are suggested. The implementation of one of the schemes, the protocol conversion, and the performance evaluation are described in detail.

• 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.

• Journal of the Korea Society of Computer and Information
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• v.12 no.1 s.45
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• pp.137-144
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• 2007

Advances in networking technology digital media, and codecs have made it possible for the Internet evolves into a Broadband convergence Network (BcN) and provides various services including Voice over Internet Protocol (VoIP) and IPTV over their high-speed IP networks. In order for the Internet to make a profit as traditional Public Switched Telephone Network (PSTN), it must provide high qualify VoIP services. Therefore, real time qualify measurement framework is the most important requisite to provide VoIP service. For this, IETF (Internet Engineering Task Force) defined RTCP-Extended Reports (RTCP-XR) that extend RTCP (Real-Time Transport Protocol Control Protocol). However, procedure and method tot actually VoIP qualify measurement did not recommended nothing but defined item to measure voice quality. Our objective in this paper is to describes a practical measuring framework for end-to-end QoS of switched voice packet in an IP environment. It includes concepts as well as step-by-step procedures for measuring packetized voice streams. It also proposes new formats that extend RTCP-XR's concept.

• Proceedings of the Korean Society of Precision Engineering Conference
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• 2001.04a
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• pp.582-585
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• 2001

Requirements for device networks differ greatly from those of data(business) networks. Consequently, any control network technology which uses a fieldbus protocol is, in general, different from IP network protocol TCP/IP. One needs to integrate fieldbus protocol and TCP/IP to realize distributed control over IP network or internet. This paper suggests a basic concept that can be applied to distributed control over IP network or internet. Specifically, LonWorks technology that uses LonTalk protocol is reviewed as device network. LonWorks technology provides networked intelligent I/O and controllers which make it a powerful, expandable solution. It is also addressed that many hardwired PLCs can be replaced by LonWorks devices. Connecting these remote LonWorks networks to the Internet can provide a powerful, integrated, distributed control system.

• Proceedings of the Korea Information Processing Society Conference
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• 2013.11a
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• pp.793-796
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• 2013

스마트폰, IPTV(Internet Protocol Television), VoIP(Voice over Internet Protocol) 등의 인터넷을 활용한 서비스의 증가는 IPv4(Internet Protocol Version4)의 주소 부족문제를 야기시켰으며 이를 해결하기 위한 장기적인 해결방안으로는 IPv6(Internet Protocol Version6)가 제시되었고 단기적으로는 NAT(Network Address Translator)가 제안되었다. NAT는 사설 IP 주소를 공인 IP 주소로 활용하여 네트워크에 접속할 수 있도록 지원하며 주소 부족 문제를 해결하고 내부 네트워크를 보호하는 긍정적인 기능도 하지만 역으로 해커들에게 숨은 공간을 제공하는 역할을 하기도 한다. 본 논문에서는 NAT 기능을 활용한 IP 공유기를 통해 내부 보안 프로세스를 우회할 수 있는 단말기를 검출하는 탐색 알고리즘을 분석하고 이를 병렬화하여 정확도를 높일 수 있는 검출 시스템을 연구하고자 한다.

• Proceedings of the KIEE Conference
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• 2002.07d
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• pp.2376-2378
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• 2002

Redundant Programmable Logic Controllers are used in practice with the aim of achieving higher degree of availability or fault tolerance. In the paper a proposed Multiplex-PC is controled for fault tolerance design. In order to select a Different-Vender by a Supervisor computer, a Multiplex-PC is designed by TCP/IP internet protocol[1]. Because TCP/IP internet protocol can not connect to CPU. We use TCP/IP internet protocol to join CPU. We have to use TCP congestion control to select Different-Vender [4]. The reliability of the fault tolerant is measured with MTBF(Mean Time Between Failures)[3].

• Journal of Advanced Navigation Technology
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• v.21 no.1
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• pp.113-118
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• 2017

As interest in mobile devices and networks has increased recently, voice over internet protocol (VoIP) service, which is a technology for transmitting voice data using an existing internet protocol (IP) network, has rapidly spread, Cheap voice call service has become possible. As the digital broadcasting service becomes popular, hybrid fiber coaxial (HFC) network technology, which uses broadband cable network through fusion of broadcasting and communication, utilizes existing communication system and network equipment to provide various new services such as interactive broadcasting service. Therefore, if UGS-AD is applied to VoCM and RTPS is applied to MTA in order to guarantee the quality of voice data in actual HFC Internet service network, it is possible to smoothly perform voice data transmission in narrow upstream band which is a problem in actual commercial HFC network We also proposed a method to improve VoIP service by improving QoS of voice data in HFC Internet service network.

• International Journal of Internet, Broadcasting and Communication
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• v.11 no.3
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• pp.59-63
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• 2019

VoIP technology is a technology for exchanging voice or video data through IP network. Various protocols are used for this technique, in particular, RTP(Real-time Transport Protocol) protocol is used to exchange voice data. In recent years, with the development of communication technology, there has been an increasing tendency of services such as "Kakao Voice Talk" to exchange voice and video data through IP network. Most of these services provide a service with security guarantee by a user authentication process and an encryption process. However, RTP protocol does not require encryption when transmitting data. Therefore, there is an exposition risk in the voice data using RTP protocol. We will present the risk of the situation where packets are sniffed in VoIP(Voice over IP) communication using RTP protocol. To this end, we configured a VoIP telephone network, applied our own sniffing tool, and analyzed the sniffed packets to show the risk that users' data could be exposed unprotected.

• 제어로봇시스템학회:학술대회논문집
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• 2005.06a
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• pp.1391-1394
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• 2005

The VoIP (Voice over IP) is being used world-widely and already put to practical use in many fields. However, it is needed to ensure the security of VoIP call in special situations. It is relatively difficult to eavesdrop commonly used PSTN network in that a 1:1 circuit connects it. However, it is difficult to ensure the security of a call on Internet because many users are connected to the Internet concurrently. This paper suggests a new model for Internet telephony to prevent eavesdrops, using VoIP (using SIP protocol) with the use the VPN protocol and establish the feasibility of its practical use comparing it with the conventional Internet telephony.

• 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.