• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.2
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• pp.243-248
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• 2012

In this paper we simulate the capacity of VOIP calls through WiFi network by computer simulations using OPNET modeler. The results show that sudden quality degradations occur on all VoIP calls when the number of call of an AP(Access Point) increases beyond a specific value. The reason of the quality degradation was turned out to be the queueing delay at the down link of AP. Under the IEEE 802.11b environments, the maximum number of VoIP calls of an AP maintaining the required voice quality (MOS > 2.5), was evaluated as 5, 12, and 27 when we use G.711, G.729a, and G.729a VAD codec, respectively.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.8
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• pp.1696-1702
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• 2005

Real Time Transport Protocol (RTP) is the Internet standard protocol for transport of real time data audio/video IP Telephony, Multimedia Seivece. In case of 8kbps voice codec, the size of packet per data is 20bytes and become more large to minimal 40bytes with adding each layer's header in RTP/UDP/IP. To solve this problem, various header compression skill were suggested on point-to-point networks. But it compress even IP header and cannot be suitable to apply to end-to-end network Thus, We will renew header compression protocol to apply wired router-based network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.12C
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• pp.273-279
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• 2001

This paper describes a design and implementation of ISDN system on a chip which provides ISDN service and used to develop a low-price multimedia communication terminal. This ISDN SOC is an ISDN system control chip which has 32bit RISC processor, and it includes ISDN S interface transceiver, G.711 voice CODEC, PC interface for data communication, ISDN protocol which includes Q.931 call control protocol and internet protocol. It provides good solution to develope ISDN terminal equipment and ISDN terminal adaptor which connected with basic rate interface, because it minimize external peripheral devices.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.7
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• pp.10-18
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• 2013

With advancing of mobile communication services and commercializing of VoLTE (Voice of LTE), it is getting to pay attention on QoS of VoLTE. This paper proposes an integrated E-model in which some factors influenced to service quality of VoIP and VoLTE based voice communication system are considered in calculating the voice quality of Wideband Codec. The model aims to calculate R value which reflects the situations of access network, network characteristics, terminals' usage and mobility. We mainly deal with the integrated E-model's structure, related algorithms and optimal parameters for VoLTE. Some experiments show that the voice quality difference between VoIP and VoiceChecker, and VoLTE and POLQA, is below 10%. With the proposed model, we can calculate the voice quality by making use of the factors directly affected to service quality and the environment of VoLTE terminal and network. As a result, we can estimate the service quality in advance, without measuring it in real wireless environment.

• The Journal of the Acoustical Society of Korea
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• v.30 no.1
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• pp.1-8
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• 2011

This paper presents a method for improving the performance of ISF coefficients quantizer through compensating the defect of the split structure vector quantization using the ordering property of ISF coefficients. And design the ISF coefficients quantizer for wideband speech codec using proposed method. The wideband speech codec uses split structure vector quantizer which could not use the correlation between ISF coefficients fully to reduce complexity and the size of codebook. The proposed algorithm uses the ordering property of ISF coefficients to overcome the defect. Using the ordering property, the codebook redundancy could be figured out. The codebook redundancy is replaced by the adaptive-extended codebook to improve the performance of the quantizer through using the ordering property, ISF coefficient prediction and interpolation of existing codebook. As a result, the proposed algorithm shows that the adaptive-extended codebook algorithm could get about 2 bit gains in comparison with the existing split structure ISF quantizer of AMR-WB (G.722.2) in the points of spectral distortion.

• The Journal of the Acoustical Society of Korea
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• v.13 no.2E
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• pp.32-40
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• 1994

It is the purpose of this paper to describe the methods for establishing loudness ratings and talker echo out of transmission quality of ISDN telephone connected to fully digital network. In order to design the desirable loudness ratings and talker echo for ISDN telephone, the model system of digital speech communication for subjective tests is developed. Using this model system, opinion tests which decide the optimal CODEC input level, the range of overall loudness rating, sidetone masking rating and talker echo are performed. From the results of tests, we decided that the loudness ratings are 6 to 8dB for sending, 0 to 2dB for receiving, and 8 to 12dB for sidetone masking rating. And, the terminal coupling loss of TCLw of at least 40dB is necessary to provide echo-free telephone communications to telophone users when the overall loudness rating of ISDN telephone is normalized to 10dB.

• The Journal of the Acoustical Society of Korea
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• v.40 no.3
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• pp.247-253
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• 2021

Directional ultrasonic speakers that are used to hear sound only in a specific area have been continuously researched on various improvements in terms of sound quality and cost compared to general speakers. In this paper, we propose a DSP based hybrid-type ultrasonic speaker that can be heard at the same time as a general speaker in order to compensate for the sound in the low-band range, considering that it is difficult to hear the low-band sound below 500 Hz due to the sensor characteristics of the ultrasonic speaker. In the case of the system that is implemented by simply connecting a general speaker and an ultrasonic speaker, there are issues of high cost and difficulties of control as two amplifiers are used to playback ultrasonic and general sound sources. In addition, sound quality deteriorates due to the difference in playback time between ultrasonic and general sound sources. In order to improve issues of cost, control and sound quality, we developed hybrid-type ultrasonic speaker with a DSP based amplifier that can simultaneously playback by synchronizing the general sound source with the regenerated ultrasonic sound source, in addition to implement the existing CODEC functions such as Dynamic Range Control (DRC) and Equalizer (EQ).

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.39 no.3
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• pp.98-109
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• 2002

Bluetooth technology is a publicly available specification proposed for Radio Frequency (RF) communication for short-range :1nd point-to-multipoint voice and data transfer. It operates in the 2.4㎓ ISM(Industrial, Scientific and Medical) band and offers the potential for low-cost, broadband wireless access for various mobile and portable devices at range of about 10 meters. In this paper, we describe the structure and the test results of the bluetooth baseband module with direct memory access method we have developed. This module consists of three blocks; link controller, UART interface, and audio CODEC. This module has a bus interface for data communication between this module and main processor and a RF interface for the transmission of bit-stream between this module and RF module. The bus interface includes DMA interface. Compared with the link controller with FIFOs, The module with DMA has a wide difference in size of module and speed of data processing. The small size module supplies lorr cost and various applications. In addition, this supports a firmware upgrade capability through UART. An FPGA and an ASIC implementation of this module, designed as soft If, are tested for file and bit-stream transfers between PCs.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.12
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• pp.20-29
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• 2014

Recently, LTE is preparing to make a new leap forward LTE-A all over the world. As LTE privides high speed service, the role of mobile phones seems to change from voice to data service. According to Cisco, global mobile data traffic will increase nearly 11-fold between 2013 and 2018. Mobile video traffic will reach 75% by 2018 from 66% in 2013 in Korea. However, voice service is still the most important role of mobile phones. Thus, controllability of throughput and low BLER is indispensable for high-quality VoIP service among various type of traffic. Although the maximum AMR-WB, 23.85 Kbps is sufficient to a VoIP call, it is difficult for the LTE which can provide tens to hundreds of MB/s may not keep the certain level VoIP QoS especially in the cell-edge area. This paper proposes a new scheduling algorithm in order to improve VoIP performance after analyzing various scheduling algorithms. The proposal is the technology which applies more priority processing for VoIP than other applications in cell-edge area based on two-tier scheduling algorithm. The simulation result shows the improvement of VoIP performance in the view point of throughput and BLER.