• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.12
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• pp.5917-5935
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• 2018

Adaptive Multi-Rate Codec is one of the codecs which is used for making voice calls. It helps to connect people who are scattered in various geographical areas. It adjusts its bit-rate according to the user's channel conditions. It plays a vital role in providing an improved speech quality of voice connection in Long Term Evolution (LTE). There are some constraints which need to be addressed in providing this service profitably. Quality of Service (QoS) is the dominant mechanism which determines the quality of the speech in communication. On several occasions, number of users are trying to access the same channel simultaneously by standing in a particular region for a longer period of time. It refers to Multi-user channel sharing problem which leads to resource loss very often. The main aim of this paper is to develop a novel RSA - QoS scheduling algorithm for reducing the Resource Loss Ratio. Eventually, it increases the throughput.The simulation result shows that the RSA - QoS increases the number of users for accessing the resources better than the existing algorithms in terms of resource loss and throughput. Ultimately, it enhances the QoS in Mobile Networks.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2018.10a
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• pp.490-492
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• 2018

This paper presents an analysis of the AMR speech data as a basic work to study the technique of inputting and transmitting AMR voice data which is widely used in the public cell phone. AMR consists of HEADER and Speech Data, and it is transmitted in bit format and has 8 bit-rate modes in total. HEADER contains mode information of Speech Data, and the length of Speech Data differs depending on the mode. We chose the best mode which is best to input into QR code and analyzed that mode. It is a goal to show a higher compression ratio for voice data by the analysis and experiments. This analysis shows improvement in that it can transmit voice data more effectively.

• ETRI Journal
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• v.31 no.4
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• pp.408-418
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• 2009

Adopting an encryption function in voice over Wi-Fi service incurs problems such as additional power consumption and degradation of communication quality. To overcome these problems, a partial encryption (PE) algorithm for compressed speech was recently introduced. However, from the security point of view, the partial encryption sets (PESs) of the conventional PE algorithm still have much room for improvement. This paper proposes a new selection method for finding a smaller PES while maintaining the security level of encrypted speech. The proposed PES selection method employs the perceptual evaluation of the speech quality (PESQ) algorithm to objectively measure the distortion of speech. The proposed method is applied to the ITU-T G.729 speech codec, and content protection capability is verified by a range of tests and a reconstruction attack. The experimental results show that encrypting only 20% of the compressed bitstream is sufficient to effectively hide the entire content of speech.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.147-150
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• 2008

The explanation of Tourism Guide of present whole country main tourist resort helps to understand the tourist resort. However, activity space of Tourism Guide is not state that can be understood all by upbringing by a natural voice. Because action of Tourism Guide is much in case of most sightseeing explanation to use microphone and speaker etc., as sticking that attach and uses to clothing and so on uses, there are much vexatious. Treatise that see hereupon makes use of establishment style fixing microphone and embody inputted obscene sounds by On-board system inflecting MCU (ATmega128), MSM7731-02 Oki-Dual Codec to minimize noise using ecad filter, and embodied a control program by serial communication method with filter codec. The resultant audible direction the maximum 59ms, the line echo maximum 27ms, the echo decrease maximum 35dB, it embodied the system which removes the adaptation elder brother noise of the back.

• The Transactions of the Korea Information Processing Society
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• v.7 no.8
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• pp.2463-2472
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• 2000

In this paper, we consider pits and falls of V oIP service scheme over the air link environment. It results that V oIP over packet switching is a more attractive approach in several points. We point out the ffilIior requirements for successful V oIP service over the air. Also, we propose V oIP CP concept for efficient wireless channel utilization. Additionally, we analyze and evaluate the performance. According to the results, It shows that the long cycle VolP vocoder CODEC such as ITU-T G.723 is better than short cycle V oIP vocoder CODEC. In this case, the increase of the simultaneous user of system is almost 60% larger than conventional circuit switching.

• Journal of Internet Computing and Services
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• v.10 no.6
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• pp.27-36
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• 2009

PSTN (public switched telephone network) provides voice communication service, whereas IP network provides data oriented service, and we can use IP network for multimedia transport service (e.g. voice over IP service) with economic price. In this paper, we propose RoIP (robot on IP) service scenario, signaling call flow, and implementation to provide home automation and monitoring service for remote site users. In our scheme, we used a extended SIP (session initiation protocol) for signaling protocol between remote site users and home robots. For our bearer transport control, we implemented H.263 video codec over RTP (real-time transport protocol) and additionally DTMF (dual tone multi-frequency) transport for robot actuator control. We implemented our scheme on home robots and experimented with KTF operator network, and it shows good communication quality (average MOS = 9.15) and flexible robot controls.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.08a
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• pp.142-147
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• 1998

Recently, 8 kb/s CS-ACELP coder of G.729 is atandardized by ITU-T SG15 and it has been reported that the speech quality of G729 is better than or equal to that of 32kb/s ADPCM. However G.729 is the fixed rate speech coder, and it does not consider the property of voice activity in mutual conversation. If we use the voice activity, we can reduce the average bit rate in half without any degradations of the speech quality. In this paper, we propose an efficient variable rate algorithm for G.729. The variable rate algorithm consists of two main subjects, the rate determination algorithm and algorithm, we combine the energy-thresholding method, the phonetic segmentation method by integration of various feature parameters obtained through the analysis procedure, and the variable hangover period method. Through the analysis of noise features, the 1 kb/s sub rate coder is designed for coding the background noise signal. So, we design the 4 kb/s sub rate coder for the unvoiced parts. The performance of the variable rate algorithm is evaluated by the comparison of speed quality and average bit rate with G.729. Subjective quality test is also done by MOS test. Conclusively, it is verified that the proposed variable rate CS-ACELP coder produced the same speech quality as G.729, at the average bit rate of 4.4 kb/s.

• 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 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.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.12 no.12
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• pp.5843-5848
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• 2011

To improve the efficiency of the channel spectrum and to reduce the power consumption of the system in EVRC, the voice signal is compressed and transmitted only when the user speaks to. In addition to this, voice frames are divided into three rates 1, 1/2 and 1/8 and each frame is handled differently. For example, we assumed that the input is silence region if the 1/8 rate is used. In this paper, the sections are firstly separated into the voiced speech signal region, unvoiced speech signal region, and silence region by using distribution characteristics of LSP parameters. Then the paper suggested to encode 1 rate for the voiced speech signal, 1/2 rate for the unvoiced speech signal region, 1/8 rate for the silence region. In other words, traditional way of transmission is used when sending full rate in the EVRC. However, when sending half rate, the voice is firstly distinguished between voiced and unvoiced. If the voice is distinguished as voiced, voice is converted into full rate before the transmission. If it is distinguished as silence, EVRC's basic rate is applied. In the experimental results with SNR, ASDM, transmission bit rate measurement, we have demonstrated that voice quality was improved by using the proposed algorithm.