• Proceedings of the Korean Institute of Intelligent Systems Conference
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• 2000.11a
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• pp.471-475
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• 2000

Voice and data are transmitted over a single packetized data communications network which is designed for data communications. The public switched telephone network for voice and the packet data network for data are merging into a single data network to get efficiency and to reduce operational cost. However, integrating voice and data transmission over a single data network is not easy because voice should be transmitted without delay but data should be transmitted without error. Advances in technology begin to overcome basic differences. Several integration methods in voice and data will be examined and reviewed here. Moreover, trends and problems on integration will be also discussed.

• Journal of the Korea Society for Simulation
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• v.10 no.2
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• pp.15-24
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• 2001

In this paper we study the delay patterns of voice traffic for end-to-end users Caused by serving the whole bunch of applications traffic at the same time on the Voice over Internet Protocol (VoIP) network. Given the current situation that voice traffic is served along with other application services on the VoIP network, it is quite necessary to figure out how and by what the voice traffic requiring high QoS is delayed. We compare the delay performance of voice traffic on the VoIP network under FIFO with the one under Weighted Fair Queuing(WFQ), and discover the differences of the delay performance resulting from the use of different voice codec algorithms. The results of our study show that using the voice codec algorithm with a higher coding rate nd the queuing algorithm of WEQ can provide users with high-quality voice traffic.

• 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.

• Proceedings of the Korea Society for Industrial Systems Conference
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• 2003.11a
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• pp.721-727
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• 2003

This study designed and implemented a system to extract of vowel from a word. The system is comprised of a voice feature extraction module and a neutral network module. The voice feature extraction module use a LPC(Linear Prediction Coefficient) model to extract a voice feature from a word. The neutral network module is comprised of a learning module and voice recognition module. The learning module sets up a learning pattern and builds up a neutral network to learn. Using the information of a learned neutral network, a voice recognition module extracts a vowel from a word. A neutral network was made to learn selected vowels(a, eo, o, e, i) to test the performance of a implemented vowel extraction recognition machine. Through this experiment, could confirm that speech recognition module extract of vowel from 4 words.

• Journal of the Institute of Convergence Signal Processing
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• v.6 no.3
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• pp.127-133
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• 2005

In this paper, we propose speech interlace using VoiceXML in home network system Existing home network uses Bluetooth, IrDA, wireless LAN and Home RF but these was able to use a long distance such as outdoors or these was difficult to using method. The proposing VoiceXML speech interlace is supported with home network services more than other interface technology in a long distance also speech interlace controls home server using a wire and a wireless phone and is informed of problems to direct calling for user through VoiceXML server. In this paper, such speech interlace is able to use the aspect of home network and supports to practical remote gauge examination, remote control services. And on the basic of that, we evaluate efficiency of purposed method.

• IEMEK Journal of Embedded Systems and Applications
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• v.5 no.2
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• pp.57-66
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• 2010

In this paper, we propose an enhanced voice recognition algorithm using adaptive MFCC(Mel Frequency Cepstral Coefficients) and neural network. Though it is very important to extract voice data from the raw data to enhance the voice recognition ratio, conventional algorithms are subject to deteriorating voice data when they eliminate noise within special frequency band. Differently from the conventional MFCC, the proposed algorithm imposed bigger weights to some specified frequency regions and unoverlapped filterbank to enhance the recognition ratio without deteriorating voice data. In simulation results, the proposed algorithm shows better performance comparing with MFCC since it is robust to variation of the environment.

• 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.

• Journal of information and communication convergence engineering
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• v.14 no.1
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• pp.9-13
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• 2016

With the introduction of the data-oriented plan for LTE service, one may concerned with the background of the ISP's policy in charging for LTE services. In this work we investigate the latest usage patterns of voice and data applications for customers over the current mobile network, via which we investigate why mobile operators introduced data-oriented plans. To be specific, we collected the real-field data for the volume of voice and data traffic from the LTE network before the data-oriented plans were introduced. From the collected data we compute the absolute volume as well as the proportion of voice and data applications. From these observations we infer mobile operators' reasoning behind the decision to introduce data-oriented plans with unlimited voice calls over the mobile network.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.4
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• pp.1006-1025
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• 2012

Improving the quality of service in IP networks is a major challenge for real-time voice communications. In particular, packet arrival-delay variation, so-called "jitter," is one of the main factors that degrade the quality of voice in mobile devices with the voice-over Internet protocol (VoIP). To resolve this issue, a receiver-based enhanced timing recovery algorithm combined with active jitter estimation 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. Extensive simulations have shown that the proposed algorithm delivers high voice quality by pursuing an optimal trade-off between average buffering delay and packet loss rate.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.1 no.2
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• pp.137-143
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• 1997

From now on, the network is being developed into PSTN(public switched telephone network) and PDN(public data network), that is depend on the form of data. The former one pursues sending voice, and the latter one pursues sending data. But it causes big loss of the economy and efficiency. So, ISDN, processing voice and data at same time, gives a big profit to user. To enlarge the ISDN at the narrow area, it is necessary that study to send the mixture form of voice and data in LAN environment. So, this paper proposes the algorithm about the mixture form of voice and data in ethernet and token-ring. that is widely used in these days.