• Journal of the Korea Computer Industry Society
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• v.3 no.5
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• pp.631-640
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• 2002

The internet phones that are currently available in use adopt packet exchange system, transferring through various routes and lacking sufficient band width with a result that there is an accompanied delay for packet transmission since the traffic is increased, accordingly affecting a lot in sound quality and speed. Two solutions for such troubles are suggested in this study to improve sound quality of internet phones. Firstly, we minimize the delay and damage regarding packet size based on traffic size by using the data algorithm from variable packets in order to supplement decreased sound quality due to the delay and damage of sound data. The second suggestion is to employ a method of Jitter compensation by giving an appropriate initial delay time with regenerating buffers to bypass troubles from Jitter, From employing the Jitter compensation method, we found that there is a sound quality improvement due to the less stoppage phenomenon.

• The Journal of the Acoustical Society of Korea
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• v.25 no.4E
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• pp.154-158
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• 2006

In this paper, we suggest a noise estimation method for speech enhancement in nonstationary noisy environments. The proposed method consists of the following two main processes. First, in order to receive fewer affect of variable signals, a best fitting regression line is used, which is obtained by applying a least squares method to coefficient magnitudes in a node with a uniform wavelet packet transform. Next, in order to update the noise estimation efficiently, a differential forgetting factor and a correlation coefficient per subband are used, where subband is employed for applying the weighted value according to the change of signals. In particular, this method has the ability to update the noise estimation by using the estimated noise at the previous frame only, without utilizing the statistical information of long past frames and explicit nonspeech frames by voice activity detector. In objective assessments, it was observed that the performance of the proposed method was better than that of the compared (minima controlled recursive averaging, weighted average) methods. Furthermore, the method showed a reliable result even at low SNR.

• Journal of the Korea Institute of Military Science and Technology
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• v.18 no.4
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• pp.423-431
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• 2015

A leased line modem usability was evaluated and investigated in the wireless internet protocol(IP) network. The signal of the modem in the circuit switching network was translated to IP packet by using several voice codecs (PCM, G.711A, $G.711{\mu}$, and etc.) and transmitted through the wireless IP network. The wireless IP network was simulated by the Tactical information and communication network Modeling and simulation Software(TMS). The performance and usability of the leased line modem are simulated using the system-in-the-loop(SITL) function of TMS with respect to packet delay, jitter, packet discard ratio, codecs, and wireless link BER.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2014.05a
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• pp.469-472
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• 2014

S-MAC is hybrids of CSMA and TDMA approaches that use local sleep-wake schedules to coordinate packet exchanges and reduce idle listening. In this method, all the nodes are considered with equal priority which may lead to increased delay during heavy traffic. The method introduced in this paper provides high throughput and small end-to-end delay suitable for applications such as real-time voice streaming and its functionality is independent of underlying synchronization protocol. The novel idea behind our scheme is that it uses the priority concept with (m,k)-firm scheduling in order to achieve its objectives. The performance of our scheme is obtained through simulations for various packet sizes, traffic loads which show significant improvements in packet delivery ratio, and delay compared to existing protocols.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.40 no.10
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• pp.9-16
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• 2003

In this paper, we propose an ATM processor suitable for VoDSL subscriber's equipments. The processor is composed of ATM block, AAL protocol block and ATS scheduler, and provides up to 4 VCC which service data and voice traffics on the ATM network. The proposed ATS scheduler can guarantee QoS of the voice traffic and supports multiple AAL2 packet. The ATM processor is manufactured on the 0.35 micron fabrication line of HYNIX semiconductor and provides the maximum data transfer rate of up to 52 Mbps. We implement the LAD, which is the VoDSL subscriber's equipment. The experimental results on the test bed network shows that the proposed hardware scheme successfully services most of the applications of the VoDSL services.

• IEIE Transactions on Smart Processing and Computing
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• v.5 no.3
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• pp.215-221
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• 2016

This paper describes an automatic vowel sequence reproduction system for a talking robot built to reproduce the human voice based on the working behavior of the human articulatory system. A sound analysis system is developed to record a sentence spoken by a human (mainly vowel sequences in the Japanese language) and to then analyze that sentence to give the correct command packet so the talking robot can repeat it. An algorithm based on a short-time energy method is developed to separate and count sound phonemes. A matching template using partial correlation coefficients (PARCOR) is applied to detect a voice in the talking robot's database similar to the spoken voice. Combining the sound separation and counting the result with the detection of vowels in human speech, the talking robot can reproduce a vowel sequence similar to the one spoken by the human. Two tests to verify the working behavior of the robot are performed. The results of the tests indicate that the robot can repeat a sequence of vowels spoken by a human with an average success rate of more than 60%.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.11 no.3
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• pp.48-54
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• 2012

The quality of real-time voice communication over Internet Protocol networks based on telematics is affected by network impairments such as delays, jitters, and packet loss. To resolve this issue, this paper proposes a receiver-based enhancing method of VoIP speech quality. The proposed method enables users to deliver high-quality voice using playout control and signal reconstruction, which consists of concealment of lost packets, adaptive playout-buffer scheduling using active jitter estimation, and smooth interpolation between two signals in a transition region. The proposed algorithm achieves higher Perceptual Evaluation of Speech Quality (PESQ) values and low buffering delay than the reference algorithm.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.8B
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• pp.739-746
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• 2004

An ATM/AAL2 is standardized to transmit delay sensitive application services, which has small size packet, efficiently. An AAL2 transmission scheme is used to deliver voice and data traffic on the lob interface between base station (Node-B) and Radio Network Controller (RNC) in UMTS network. To predict AAL2 performance, a detailed end-to-end UMTS network performance simulator was developed. We performed detailed simulation(cell packing density and bandwidth gain) for voice and data services in UTRAN. The results indicate that the maximum bandwidth gain in Node-B is about 17% and the bandwidth gain of AAL2 multiplexing in $I_{ub}$ for data services is less than that for voice service. Futhermore, the more offered load increase the more the bandwidth gain decreases in a concentrator.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.39 no.6
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• pp.687-695
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• 2002

Media-specific FEC techniques, suggested to confront with VoIP speech packet loss, improve speech quality at the expense of generating additional one-frame delay. In this paper, we suggest new media-specific FEC, i.e, LSF FEC technique which is able to improve speech quality with much shortened additional delay. In the proposed technique, the LSF parameters of the future frame are utilized to recover a lost packet. To evaluate performance of the proposed technique, we use ITU-T G.723.1 and G.729 Codec and apply Gilbert packet loss model and estimate MOS per every packet loss rate using PESQ speech quality estimation algorithm. The proposed technique has effect of shortening delay over from 6.5ms to 27ms compared with existing media-specific FEC techniques. Simulation results for comparison of reconstructed speech quality show this novel technique improves the MOS over 0.1 in practical lossy environment of 5 % packet loss rate.