• Broadcasting and Media Magazine
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• v.19 no.1
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• pp.48-55
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• 2014

One of the recent noticeable evolutions in mobile communication systems is that wideband-codec is deployed rapidly in VoLTE (Voice over Long Term Evolution) service or HD voice. This paper is concerned with next generation HD voice or VoLTE service that is coined to describe high quality communication with super-wideband voice codec. 3GPP EVS (Enhanced Voice Service) Codec is being standardized to develop the super-wideband voice codec. This paper deals with the codec design constraints, performance requirements, the status of standardization, and finally perspective on VoLTE service in future.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.97-100
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• 1999

Mobile network applications are growing and this requires a fast and efficient transport method between the BS(Base Station) and the MSC(Mobile Switching Center). One possible solution is to use ATM and a voice CODEC standard which compresses 64kbps voice data to less than 8kbps. The low bit tate and small-sized packets made by the voice CODEC imply that significant amount of link bandwidth would be wasted, if this small-sized packet is carried by one ATM cell. The cell assembly delay increases if one ATM cell is fully filled with the small-sized packets. For the bandwidth-efficient transmission of low-rate, short, and variable length packets in delay sensitive applications, AAL-2 was standardized. This paper evaluates performance of AAL-2 by using voice CODEC standard.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.2
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• pp.51-59
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• 2002

In this paper, we studied the voice cryptograph system designed by the SEED algorithm for the safe transmission and receipt on the voice communication. Voice band signal converts to digital signal by the CODEC and DSP that applied the improved SEED algorithm encrypt the digital signal. The CODEC convert Encryption signal into analog voice signal. This voice signal is transmitted safely because of encryption signal even if someone wiretap. Receiver can hear the source voice, because the encryption signal decrypted using the SEED algorithm. In this paper, We designed the 32 round key instead of 16 round key in the SEED algorithm so that we improve the truncated differential probability from $2^{-143.1}$ to $2^{-286.6}$. 

• Journal of Korea Society of Industrial Information Systems
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• v.23 no.6
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• pp.59-66
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• 2018

In this paper, we experimented how long distance voice communication is possible After implemented PTT(Push to talk) Bi-directional radio using open source project Codec 2, which is a low speed voice codec for digital amateur radio and 900MHz FSK transceiver. In case of a general digital radio, the AMBE+2 codec, which is regarded as an industry standard in terms of performance, is expensive and has the monopoly of technology. Using the 400MHz band in terms of frequency, narrow bandwidth of DMR(12.5kHz) and DPMR(6.25kHz) is used, so the data rate is low. In the 900MHz bandwidth can be extended, which is advantageous in terms of data transmission. As a result of the voice quality and distance field test, we could find that the communication takes place within about 500m. In this paper, only voice communication is reviewed. if a review of data transmission such as a simple image is added, this solution can be used in various fields as a low cost IOT radio.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.230-235
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• 2008

This paper addresses issues associated with the real-time implementation of a wideband speech codec such as ITU-T G. 729. 1 on an ARM processor in order to provide an improved voice quality of a VoWiFi service. The real-time implementation features in optimizing the C-source code of G.729. 1 and replacing several parts of the codec algorithm with faster ones. The performance of the implementation is measured by the CPU time spent for G.729.1 on the ARM926EJ processor that is used for a VoWiFi phone. It is shown from the experiments that the G.729.1 codec works in real-time with better voice quality than G 729 codec that is conventionally used for VoIP or VoWiFi phones.

• 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 Korean Institute of Industrial Engineers
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• v.31 no.4
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• pp.349-357
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• 2005

The Adaptive Multi-Rate (AMR) speech codec is the mandatory for voice service in WCDMA systems. The AMR codec can be used efficiently to provide a balanced trade-off between the capacity and quality of voice by adjusting various service rates. In this paper, three ways of AMR mode allocation schemes on the downlink in WCDMA system are evaluated. To evaluate users satisfaction efficiently, new system performance measure and analytic models are proposed. The proposed analytic models can be applied to obtain optimal mode allocation ways while considering the system capacity and quality of voice. In numerical examples, the ways of finding optimal parameters are illustrated for the given traffic loads and the performances of three mode allocation schemes are compared.

• Journal of the Korea Institute of Military Science and Technology
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• v.20 no.3
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• pp.431-440
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• 2017

HF communications are used as a last means of long distance communications without any relay node in NLOS (Non Line-Of-Sight) environment. Conventional analog voice communication in the HF band is vulnerable to security as well as severe background noise. To overcome these shortcoming, digital voice was introduced into HF radios in the early 1980s. In this paper, we analyze avaliable digital voice communication ground wave range of HF radios that applied MELPe CODEC and MIL-STD-188-110B physical layer standard using GRWAVE program. And we evaluate usefulness of digital voice communication in HF band.

• Proceedings of the KIEE Conference
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• 1999.07g
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• pp.3246-3248
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• 1999

The APC is a coding algorithm which has the middle property of both Wave Coding(ex ADPCM) and Vocoding(ex CELP) and can decode a proper quality of sound by using scalar quantizer instead of vector quantizer at computation a low calculation. So, the APC required for Voice Codec of INMARSAT-B could be successfully implemented by full duplex using TMS32OC30(DSP).

• 한국정보통신설비학회:학술대회논문집
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• 2008.08a
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• pp.584-587
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• 2008

In this paper, we propose a real time video codec adapter for enabling video communications with terminals having a codec which is different from each other. When multimedia services are playing with an office service phone such as a video phone or software phone which has video capability, each terminal is not being considered to have optimized video or voice codec. So when a video phone with only one type of video codec is used in the video streaming service which requires another type of codec, the streaming service is not successful without codec transformation. The real time codec adapter in this paper provides a real time code transformation which enables communication services such as video conferencing between terminals which have different codec.