• The Journal of the Korea institute of electronic communication sciences
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• v.11 no.12
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• pp.1187-1192
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• 2016

Performance analysis results are limited to of network level, because network level transmission parameters are used for performance measure and analysis of network design, construction and operation on underwater MANET, With this way of performance analysis based on network level, it is not easy to analyze transmission performance related with user level transmission quality. In this paper, transmission performance focused on application traffic be required by user is investigated to supplement weakness of performance analysis based on network level. Voice traffic, which is expected to be increasingly used on underwater MANET, is considered as application service, Some conditions for underwater MANET will be proposed to support transmission quality, MOS, CCR and EED, etc.. A computer simulation based on NS-2 is used for performance measure, voice traffic is generated as VoIP specification.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.10 no.6
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• pp.63-69
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• 2010

Today, Internet Telephony Services based on VoIP are gaining tremendous popularity for general user. Therefore a various demands of the user keep up increase, the most important requirements of these is voice security about telephony system. It is needed to ensure secret of voice call in a special situation. Due to the fact that many users can connect to the internet at the same time, VoIP can always be in a defenseless state by hackers. Therefore, in this paper, we have developed VPN IP-PBX for the voice security and measured conversation quality by adopting VPN IPsec based on SIP and using tunnel method in transmitting voice data to prevent eavesdrop of voice data. This VPN IP-PBX that is connected Soft-phone provide various optional services.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.9C
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• pp.842-852
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• 2002

The fast advance in the VoIP technologies gives a rich opportunity to create different kind of VoIP applications such as IP telephony services. The application level call signaling protocols such as ITU-T H.323 and IETF SIP provide the communication functions of end-to-end call setup and release. Currently, there is a lot of H.323 based VoIP products in the market, however SIP is considered as a suitable protocol for supporting applications in IP environments, so SIP based VoIP products and services begin to appear. In this paper, firstly we present the characteristics of some possible SIP based applications and describe the design and implementation of a VoIP example service named PC-to-PC Internet telephony service using the developed SIP network components. The PC-to-PC Internet telephony service and User Agent are developed in MS window 98/2000 using visual C/C++, and Proxy server and Registrar in Linux 7.0 using C, respectively.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.10 no.4
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• pp.71-77
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• 2011

In this paper, we propose a conversational quality measurement (CQM) system for providing the objective QoS of high quality mobile VoIP voice telecommunication. For measuring the conversational quality, the VoIP telecommunication system is implemented in two smart phones connected with VoIP. The VoIP telecommunication system consists of echo cancellation, noise reduction, speech encoding/decoding, packet generation with RTP (Real-Time Protocol), jitter buffer control and POS (Play-out Schedule) with LC (loss Concealment). The CQM system is connected to a microphone and a speaker of each smart phone. The voice signal of each speaker is recorded and used to measure CE (Conversational Efficiency), CS (Conversational Symmetry), PESQ (Perceptual Evaluation of Speech Quality) and CE-CS-PESQ correlation. We prove the CQM system by measuring CE, CS and PESQ under various SNR, delay and loss due to IP network environment.

• Journal of the Korea Society of Computer and Information
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• v.13 no.4
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• pp.161-167
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• 2008

A method to compute the size of packet and the optimal way to transmit the packets are proposed in this work for the VoIP communication using the MIL-STD-188-220C, military wireless Ad-hoc protocol which is used for the amicable communications of both speeches and data between several radiotelegraph. The expected time of data transmission is estimated beforehand, and then the size of package and transmission method are decided in the consideration of VoIP speech quality for the users as well as the data transmission quality of radiotelegraph.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.9
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• pp.2078-2083
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• 2012

Tactical communication environment evolving for the purpose of providing services such as voice, video and text based on ALL-IP. Therefore, To be able to guarantee QoS level which meets the required level of subscriber for these services in the constrained tactical communication infrastructure, it is required to take the characteristics such as wireless transmission link, mobility of troops or personal into service quality scheme. In this paper, to support differentiated QoS for each individual or mission in the tactical communication environment, we presents a technique that can provide same QoS level which was served originally regardless of the situation to user's move through dynamically determining the QoS level to be provided at the time of the service request on VoIP-Switch.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.6
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• pp.1022-1025
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• 2008

Session Initiation Protocol (SIP) has become the call control protocol of choice for Voice over IP (VoIP) networks because of its open and extensible nature. However, the integrity of call signaling between sites is of utmost importance, and SIP is vulnerable to attackers when left unprotected. Currently a herby-hop security model is prevalent, wherein intermediaries forward a request towards the destination user agent sewer (UAS) without a user agent client (UAC) knowing whether or not the intermediary behaved in a trusted manner. This paper presents an integrated security model for SIP-based VoIP network by combining hop-by-hop security and end-to-end security.

• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.7
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• pp.1568-1576
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• 2015

This paper proposes a prediction reputation system for the anti-SPIT solution in real-time VoIP environment. Increased accuracy of the determination as to whether spam or not by deriving a threshold based on SPIT presence in the existing paper. The existing schemes need to get the user's feedback and/or have experienced the time delay and overload as session initiates due to real-time operation. To solve these problems, the proposed scheme predicts the reputation through the statistical analysis based on the period of session initiation of each caller and the call duration of each receiver. As per the second mentioned problem, this scheme performs the prediction before session initiation, therefore, it's proper for real-time VoIP environment.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.295-297
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• 2008

Session Initiation Protocol (SIP) has become the call control protocol of choice for Voice over IP (VoIP) networks because of its open and extensible nature. However, the integrity of call signaling between sites is of utmost importance, and SIP is vulnerable to attackers when left unprotected. Currently a hop-by-hop security model is prevalent, wherein intermediaries forward a request towards the destination user agent server (UAS) without a user agent client (UAC) knowing whether or not the intermediary behaved in a trusted manner. This paper presents an integrated security model for SIP-based VoIP network by combining hop-by-hop security and end-to-end security.