• 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 the Institute of Electronics Engineers of Korea TC
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• v.40 no.12
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• pp.9-17
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• 2003

In this paper we proposea method for estimating the bandwidth in next-generation If network. Especially, we concentrate on the edge routers accommodating the VoIP connections as well as a group of data connections. Bandwidth dimensioning is carried out at call level and packet level for voice traffic in the next-generation IP network. The model incorporates the statistical estimation approach at a call level for obtaining the number of voice connections simultaneously in the active mode. The call level model incorporates a statistical technique to compute the statistics of the number of active connections such as the mean and variance of the simultaneously connected calls in the network. The packet level model represents a load map for voice and data traffic by using non-preemptive M/G/1 queuing model with strict priority for voice over data buffer, From the proposed traffic model, we can derive a graph for upper bounds on the traffic load in terms of bandwidth for voice and data connections. Via numerical experiments we illustrate the implication of the work.

• 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.

• 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.

• ETRI Journal
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• v.38 no.3
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• pp.444-454
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• 2016

Voice service is very demanding in cognitive radio networks (CRNs). The available spectrum in a CRN for CR users varies owing to the presence of licensed users. On the other hand, voice packets are delay sensitive and can tolerate a limited amount of delay. This makes the support of voice traffic in a CRN a complicated task that can be achieved by devising necessary considerations regarding the various network functionalities. In this paper, the support of secondary voice users in a CRN is investigated. First, a novel packet scheduling scheme that can provide the required quality of service (QoS) to voice users is proposed. The proposed scheme utilizes the maximum packet transmission rate for secondary voice users by assigning each secondary user the channel with the best level of quality. Furthermore, an analytical framework developed for a performance analysis of the system, is described in which the effect of erroneous spectrum sensing on the performance of secondary voice users is also taken into account. The QoS parameters of secondary voice users, which were obtained analytically, are also detailed. The analytical results were verified through the simulation, and will provide helpful insight in supporting voice services in a CRN.

• Journal of Convergence for Information Technology
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• v.11 no.6
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• pp.33-39
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• 2021

Speech recognition technology is being combined with deep learning and is developing at a rapid pace. In particular, voice recognition services are connected to various devices such as artificial intelligence speakers, vehicle voice recognition, and smartphones, and voice recognition technology is being used in various places, not in specific areas of the industry. In this situation, research to meet high expectations for the technology is also being actively conducted. Among them, in the field of natural language processing (NLP), there is a need for research in the field of removing ambient noise or unnecessary voice signals that have a great influence on the speech recognition recognition rate. Many domestic and foreign companies are already using the latest AI technology for such research. Among them, research using a convolutional neural network algorithm (CNN) is being actively conducted. The purpose of this study is to determine the non-voice section from the user's speech section through the convolutional neural network. It collects the voice files (wav) of 5 speakers to generate learning data, and utilizes the convolutional neural network to determine the speech section and the non-voice section. A classification model for discriminating speech sections was created. Afterwards, an experiment was conducted to detect the non-speech section through the generated model, and as a result, an accuracy of 94% was obtained.

• The Journal of the Acoustical Society of Korea
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• v.33 no.1
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• pp.75-80
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• 2014

The quality of real-time Voice over Internet Protocol (VoIP) network is affected by network impariments such as delays, jitters, and packet loss. This paper proposes a packet loss concealment algorithm based on voice classification for enhancing VoIP speech quality. In the proposed method, arriving packets are classified by an adaptive thresholding approach based on the analysis of multiple features of short signal segments. The excellent classification results are used in the packet loss concealment. Additionally, linear prediction-based packet loss concealment delivers high voice quality by alleviating the metallic artifacts due to concealing consecutive packet loss or recovering lost packet.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.2011-2016
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• 2004

Study of human-robot communication is one of the most important research areas. Among various communication media, any useful law we find in voice communication in human-human interactions, is significant in human-robot interactions too. Control strategy of most of such systems available at present is on/off control. These robots activate a function if particular word or phrase associated with that function can be recognized in the user utterance. Recently, there have been some researches on controlling robots using information rich fuzzy commands such as "go little slowly". However, in those works, although the voice command interpretation has been considered, learning from such commands has not been treated. In this paper, learning from such information rich voice commands for controlling a robot is studied. New concepts of the coach-player model and the sub-coach are proposed and such concepts are also demonstrated for a PA-10 redundant manipulator.

• Journal of the Korea Institute of Military Science and Technology
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• v.24 no.1
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• pp.107-114
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• 2021

By analyzing the voice communication performance of the existing tactical FM radios, the performance target of the newly developing TICN combat network radio system VHF band tactical voice communication waveform was derived. In addition, a vocoder and modulation method that can satisfy the performance target and additional requirements are presented, and the expected voice communication quality is analyzed.

• Journal of Korea Multimedia Society
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• v.8 no.6
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• pp.836-843
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• 2005

The Wireless Internet services widely spread out with the developments of CDMA(Code Division Multiple Access) networks and wireless units. In contrast to the telecommunication network, WPAN (Wireless Personal Area Network) enables to transmit data and voice in personal area. Although WPAN technologies are commercially utilized, the combined services with COMA network are not so poplar up to now. Various services can be provided using the combination between COMA and WPAN. This paper presents the practical and united model between COMA and WPAN. Specially, the main focus of this research lies on the design of the Middle-ware system of a handset which could be managing both COMA and WPAN. This system used Bluethooth by WPAN. For the devices with the proposed WPAN Middle-ware, service areas of the COMA network can be expanded to WPAN, various services can be realized by the transmission of data and voice, and consequently, the user computing environment could be improved.