• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2009.10a
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• pp.795-797
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• 2009

Voice over Internet Protocol(VoIP) service becomes more and more popular nowadays. As such, it is developed over many kinds of network models, especially wireless networks. Mean Opinion Score(MOS) computes the QoS of VoIP service which should be supported by robust network environment. However, MANET is not stable enough to supply high MOS values for VoIP service. In this paper, VoIP service over MANET is simulated using ns-2(Network Simulation 2). In oder to get different MOS values in the results, we differentiate between network environments by adjusting the parameters of MANET.Through comparing the results we can know how to improve the QoS.

• Journal of KIISE:Computing Practices and Letters
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• v.14 no.3
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• pp.261-265
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• 2008

As increasing the user of VoIP service using ITSP(Internet Telephony Service Provider), the VoIP spam becomes a big problem. The spam used in the existing public telephone is detected by using the pattern inspection of call behavior because it is difficult to filter contents for the characteristic of real-time voice communication. However there is a false-positive problem. The threat on spam remains where spam with low threshold can't be detected or users share one number. In this paper, we propose anti-spam for VoIP based on luring test. The proposed method gives a user luring test and he/she can connect to a receiver if passing turing test. A ticket is given to a user that pass luring test and it reduces overhead of luring test in re-dial. The proposed method is implemented on ASUS WL-500G wireless router and Asterisk IP-PBX. Experimental results show the effectiveness of the proposed method.

• Journal of the Korea Society for Simulation
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• v.15 no.3
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• pp.87-91
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• 2006

With the growth of the internet, the number of wireless internet users has increased continuously up to date. However, mobile communications could not support high speed transmission rate with cheap communication fee and wireless LAN has problems in providing terminal mobility and wide area connectivity, respectively. So the WMAN standard has been newly designed to make up for these limits. The initial 802.16 specification effectively offers a solution for providing fixed users with high speed wireless communication but it does not offer terminal mobility. So the 802.16d and 802.16e have been developed as the next generation solution that can support various PHY layer (SC, SCa, OFDM, OFDMA) and offer the terminal mobility. In this paper, we propose an effective uplink scheduling algorithm for VoIP with using UGS, and we show that our proposed algorithm is superior in view of average delay and management of uplink bandwidth to conventional rtPS algorithm and the scheme in reference, with using NS-2 network simulator.

• The KIPS Transactions:PartD
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• v.14D no.5
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• pp.565-572
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• 2007

In this paper, we developed a VoWLAN(Voice over WLAN) system based on an open source software. The system aims to provide VoIP service over wireless LAN with an IP-PBX server. The features of system presented in this paper are as follows. First, the initial cost for the development is reduced since the system is developed based on open source software. Second, the system provides various additional services such as Voice Mail, Conference Call, and Interactive Voice Response with a software IP-PBX server. Third, the VoWLAN terminal provides high-level user applications with minimal system resources using lightweight open software solutions. Finally, it is highly scalable since it is based on the open source software.

• Journal of The Institute of Information and Telecommunication Facilities Engineering
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• v.2 no.3
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• pp.9-16
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• 2003

In this paper, we propose the burst error reduction (BER) algorithm for VoIP service in the wireless LAN network. In end point device, this BER algorithm can be achieved packet loss bounded QoS provisioning using interleaving in buffering and FEC (Forward Error Correction) through transmitting voice packet. BER algorithm can reduced the voice packet loss rate 5.5%-60% in VoIP network using wireless LAN.

• Proceedings of the KIEE Conference
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• 2004.11c
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• pp.237-239
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• 2004

Wireless LAN Voice over IP(VoIP) equipment needs Quality-of-Service(QoS) with priority for processing real-time traffic. This paper shows transmit function implementation of wireless LAN(WLANs) media access control(MAC) support VoIP, and it has an advantage of guarantee of QoS and is adaptable to VoIP or mobile wireless equipment. The IEEE 802.11e standard in progress has four queues according to four access categories(AC) for transmit and the MAC transmits the data based on EDCA. The value of AC is from AC0 to AC3 and AC3 has the highest priority. The transmit method implemented at this paper ensure QoS using one transmit FIFO in hardware since real-time traffic data and non real-time traffic data has the different priority. The device driver classifies real-time data and non real-time data and transmit data to hardware with information about data type. The hardware conducts shorter backoff and selects faster AIFS slot for real-time data than it for non real-time data. Therefor It make give the real-time traffic data faster channel access chance than non real-time data and enhances QoS.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.05a
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• pp.750-752
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• 2010

MANET has some interest as a wireless mobile network supporting node mobility and having no additional infrastructure. Speed of mobile node on MANET is various from just slow speed of human's walk to vehicle's moving speed. In this study, VoIP performance of MANET is evaluated in the view of speed of mobile node. VoIP performance is studied on walking speed, vehicle speed and higher speed on MANET. PDR(Packet Delivery Ratio), MOS(Mean Opinion Score) are used as performance parameters.

• Proceedings of the Korea Contents Association Conference
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• 2004.11a
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• pp.125-128
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• 2004

Although the current VoIP service is limited within the wire network, the future network will be a united network of wire/wireless. It is required that the study on the granting plan of numbering system by not considering the simple VoIP service but considering the entire network of wire/wireless. In this paper, we proposed a desirable VoIP service activation plan in mobile network suited to the future network by considering on VoIP service and mobile VoIP service.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.8
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• pp.1339-1346
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• 2006

This thesis outlines a preparation plan for e-Trade business service regarding tendency development in super-highway information network including internet, cable, and wireless communication. It also explains two perspectives in e-Trade market: changes of circumstances and consumerism, and revitalization devices of Von(Voice over Internet Protocol) service technology for creating new market in rapidly changing IP(Internet Protocol) environment. Plus it illustrates what core competence and progress business organizations must have in current situation, forecasts turns of future e-Trade market, and analyzes technological applications of VoIP service in an extended viewpoint of corporate strategy.

• The Journal of the Korea institute of electronic communication sciences
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• v.9 no.6
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• pp.719-726
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• 2014

This paper addresses quality of experience(QoE) and how to measure and evaluate QoE including its subjective aspects. Adopting the real measurements on the field, a various quality metric have been measured for VoIP(voice over IP) service provided through a wireless interface of WiBro(Wireless Broadband). By analyzing the measured values and correlation between the metrics, we attempt to find a method to evaluate QoE of the VoIP service in a objective way. As a result, it has been shown that QoE of the VoIP service through WiBro network has close relation to the packet-level end-to-end delay, and the delay has close relation to received signal strength indicator(RSSI).