• Journal of the Korea Institute of Information and Communication Engineering
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• v.14 no.5
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• pp.1109-1116
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• 2010

In this paper, some performance characteristics of VoIP(Voice over Internet Protocol) for MANET(Mobile Ad-hoc Networks) with simulation is studied and appropriate condition for implementation of VoIP service is suggested. VoIP simulator is implemented with NS(Network Simulator)-2. VoIP traffic for simulation is generated with some codecs of G.711, G.723.1, G.726-32, G.729A, GSM.AMR and iLBC. As simulation results for traffic transmission under $670{\times}670m$ 50node MANET environment, performance data for MOS(Mean Opinion Score), network delay, packet loss rate and transmission bandwidth are measured. Normalized analysis about measured results shows that maximum VoIP connection satisfying VoIP service quality condition is 15.

• Proceedings of the Korea Contents Association Conference
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• 2004.05a
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• pp.342-347
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• 2004

The VoIP(Voice over Internet Protocol) technology which is able to use a voice service through internet is more cheaper then existing telephone charges, and is easily accept the various of multimedia services from internet. Previous connection method of VoIP used H.323 protocol, but it is very complex to connection establishment. so, the SIP(Session Initiation Protocol) protocol that propose in SIP-Working Group Is in use recently. Therefore, we need new interworking methodology between H.323 and SW products. In this thesis, the progress interworking method between H.323 and SIP are propose, then interpret unnecessary packet delay for call setup and improved feature of message exchange.

• Journal of Internet Computing and Services
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• v.10 no.6
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• pp.27-36
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• 2009

PSTN (public switched telephone network) provides voice communication service, whereas IP network provides data oriented service, and we can use IP network for multimedia transport service (e.g. voice over IP service) with economic price. In this paper, we propose RoIP (robot on IP) service scenario, signaling call flow, and implementation to provide home automation and monitoring service for remote site users. In our scheme, we used a extended SIP (session initiation protocol) for signaling protocol between remote site users and home robots. For our bearer transport control, we implemented H.263 video codec over RTP (real-time transport protocol) and additionally DTMF (dual tone multi-frequency) transport for robot actuator control. We implemented our scheme on home robots and experimented with KTF operator network, and it shows good communication quality (average MOS = 9.15) and flexible robot controls.

• The Journal of the Acoustical Society of Korea
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• v.33 no.2
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• pp.133-138
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• 2014

Packet arrival-delay variation, so-called 'jitter' is one of the main factors that degrade the quality of voice in mobile devices at the Voice over Internet Protocol (VoIP). To resolve this issue, a playout scheduling based on adaptive jitter estimation for enhancing VoIP speech quality is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. The experimental results have shown that the proposed algorithm delivers high voice quality in unstable network environment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.10
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• pp.1994-2005
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• 2015

Voice over IP (VoIP) is a technology for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks. Instead of being transmitted over a circuit-switched network, however, the digital information is packetized, and transmission occurs in the form of IP packets over a packet-switched network which consist of several layers of computers. VoIP Service that used the various techniques has many advantages such as a voice Service, multimedia and additional service with cheap cost and so on. But the various frauds arises using VoIP because VoIP has the existing vulnerabilities at the Internet and based on complex technologies, which in turn, involve different components, protocols, and interfaces. According to research results, during in 2012, 46 % of fraud calls being made in VoIP. The revenue loss is considerable by fraud call. Among we will analyze for Toll Bypass Fraud by the SIM Box that occurs mainly on the international call, and propose the measures that can detect. Typically, proposed solutions to detect Toll Bypass fraud used DPI(Deep Packet Inspection) based on a variety of detection methods that using the Signature or statistical information, but Fraudster has used a number of countermeasures to avoid it as well. Particularly a Fraudster used countermeasure that encrypt VoIP Call Setup/Termination of SIP Signal or voice and both. This paper proposes the solution that is identifying equipment of Toll Bypass fraud using those countermeasures. Through feature of Voice traffic analysis, to detect involved equipment, and those behavior analysis to identifying SIM Box or Service Sever of VoIP Service Providers.

• International Journal of Computer Science & Network Security
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• v.22 no.2
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• pp.175-184
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• 2022

Voice over Internet Protocol (VoIP) has grown in popularity as a low-cost, flexible alternative to the classic public switched telephone network (PSTN) that offers advanced digital features. However, additional security vulnerabilities are introduced by the VoIP system's flexibility and the convergence of voice and data networks. These additional challenges add to the normal security challenges that a VoIP system's underlying IP data network infrastructure confront. As a result, the VoIP network adds to the complexity of the security assurance task faced by businesses that use this technology. It's time to start documenting the many security risks that a VoIP infrastructure can face, as well as analyzing the difficulties and solutions that could help guide future efforts in research & development. We discuss and investigate the challenges and requirements of VoIP security in this research. Following a thorough examination of security challenges, we concentrate on VoIP system threats, which are critical for present and future VoIP deployments. Then, towards the end of this paper, some future study directions are suggested. This article intends to guide future scholars and provide them with useful guidance.

• Journal of Internet Computing and Services
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• v.5 no.6
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• pp.31-43
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• 2004

Voice over Internet Protocol (VoIP) is currently a popular research topic as a real time voice packet transmission method. But current Internet environment do not guarantee the quality of voice when we take a side view of delay, jitter and loss. Up to now, many voice based evaluation algorithms have been used to measure speech quality of VoIP systems. However, these algorithms have the defects that their results are different according to voice samples and some algorithms can not take network environment for speech transmission path. The E-model can be used to solve the problems of these algorithms. In this paper. we introduce VoIP planning guidelines through the various analysis of E-model which can model impairments of network quality as well as VoIP equipment quality systematically, We, also, show the evaluation method and results of speech transmission quality.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.2
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• pp.245-252
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• 2011

VoIP transmits voices over IP-based networks and it is the abbreviation of Voice over Internet Protocol. Recently, VoIP provides various services in addition to voices. Since VoIP services' provision is extending, VoIP service quality management is becoming an important issue. Therefore, in this paper, we study VoIP service quality management. We examine VoIP technology, service types, and network architecture. Then, we investigate key quality indicators(KQIs)/key performance indicators(KPIs) in terms of customers, not network service providers. Toward this, we also study the concept of general service quality management as well as the concept of telecommunication related service quality management. Moreover, we apply $\bar{x}$ and R charts to show how to use statistical quality control techniques in real telecommunication companies with one KQI.

• The Journal of the Acoustical Society of Korea
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• v.28 no.3E
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• pp.99-108
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• 2009

In this paper, we suggest criteria for objective measurement of speech quality in mobile VoIP (Voice over Internet Protocol) services over wireless mobile internet such as mobile WiMAX networks. This is the case that voice communication service is available under other networks. When mobile VoIP service users in the mobile internet network based on packet call up PSTN and mobile network users, but there have not been relevant quality indexes and quality standards for evaluating speech quality of mobile VoIP. In addition, there are many factors influencing on the speech quality in packet network. Especially, if the degraded speech with packet loss transfers to the other network users through the handover, voice communication quality is significantly deteriorated by the transformation of speech codecs. In this paper, we eventually adopt the Gilbert-Elliot channel model to characterize packet network and assess the voice quality through the objective speech quality method of ITU-T P. 862. 1 MOS-LQO for the various call scenario from mobile VoIP service user to PSTN and mobile network users under various packet loss rates in the transmission channel environments. Our simulation results show that transformation of speech codecs results in the degraded speech quality for different transmission channel environments when mobile VoIP service users call up PSTN and mobile network users.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.3B
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• pp.161-167
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• 2007

In this paper, we evaluate the capacity of voice over internet protocol (VoIP) services over high-speed downlink packet access (HSDPA), in which frame-bundling (FB) is incorporated to reduce the effect of relatively large headers in the IP/UDP/RTP layers. Also, a modified proportional pair (PF) packet scheduler design supporting for VoIP service is provided. The main focus of this work is the effect of FB on system outage based on delay budget in radio access networks. Simulation results show that VoIP system performance with FB scheme is highly sensitive to delay budget. We also conclude that HSDPA is attractive for transmission of VoIP if compared to the circuit switched (CS) voice that is used in WCDMA (Release'99).