• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.2A
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• pp.187-194
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• 2004

We propose priority-based reservation code multiple access (P-RCMA) which can enhance voice traffic quality of the previous RCMA. The proposed protocol maintains two power levels and consider traffic characteristics in contending shared available codes to transmit packets. P-RCMA gives priority to the voice request packets rather than data packets by capture effect at the receiver part of base station. We show numerical results from EPA (equilibrium point analysis) analysis and simulation study in terms of voice packet dropping probability and average data packet transmission delay.

• The Journal of the Acoustical Society of Korea
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• v.18 no.6
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• pp.71-76
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• 1999

In digital cellular communication systems, tandem coding occurs in communications between mobile phones with different speech codecs, resulting in poor voice quality, high computational load, and long transmission delay. In this paper, voice packet conversion technique is proposed to solve the tandem coding problems, and packet conversion algorithm from 13kbps QCELP to 8kbps QCELP is developed. Simulations using various speech data show that the proposed packet conversion method produces voice quality which is equivalent to that by the conventional tandem coding method with shorter transmission delay using about 33% computational load.

• Journal of KIISE:Information Networking
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• v.29 no.2
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• pp.141-155
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• 2002

A medium access control protocol is proposed for integrated voice and data services in the packet CDMA network with a small coverage. Uplink channels are composed of time slots and multiple spreading codes for each slot. This protocol gives higher access priority to the delay-sensitive voice traffic than to the data traffic. During a talkspurt, voice terminals reserve a spreading code to transmit multiple voice packets. On the other hand, whenever generating a data packet, data terminals transmit a packet based on the status information of spreading codes in the current slot, which is received from base station. In this protocol, voice packet does not come into collision with data packet. Therefore, this protocol can increase the maximum number of voice terminals.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1999.05a
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• pp.126-130
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• 1999

In packet CDMA networks, it is important to design a MAC protocol that meets the QoS requirements , of the different traffic types and allocates the radio channels efficiently. In this paper, a RRS$^2$-CDMA MAC protocol is proposed for integrating voice and data services in the microcellular packet CDMA networks. In PRS$^2$-CDMA, a voice terminal can resolve a spreading code to transmit voice packets during a talkspurt while a data terminal has to contend for a code for each packet transmission. The numerical results show that the proposed protocol can improve the system capacity, while guaranteeing the QoS of voice and data services.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.11 no.3
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• pp.179-196
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• 1986

In this paper, the peformance of a statistical packet voice/data multiplexer is studied. In ths study we assume that in the packet voice/data multiplexer two separate finite queues are used for voice and data traffics, and that voice traffic gets priority over data. For the performance analysis we divide the output link of the multiplexer into a sequence of time slots. The voice signal is modeled as an (M+1) - state Markov process, M being the packet generation period in slots. As for the data traffic, it is modeled by a simple Poisson process. In our discrete time domain analysis, the queueing behavior of voice traffic is little affected by the data traffic since voice signal has priority over data. Therefore, we first analyze the queueing behavior of voice traffic, and then using the result, we study the queueing behavior of data traffic. For the packet voice multiplexer, both inpur state and voice buffer occupancy are formulated by a two-dimensional Markov chain. For the integrated voice/data multiplexer we use a three-dimensional Markov chain that represents the input voice state and the buffer occupancies of voice and data. With these models, the numerical results for the performance have been obtained by the Gauss-Seidel iteration method. The analytical results have been verified by computer simylation. From the results we have found that there exist tradeoffs among the number of voice users, output link capacity, voic queue size and overflow probability for the voice traffic, and also exist tradeoffs among traffic load, data queue size and oveflow probability for the data traffic. Also, there exists a tradeoff between the performance of voice and data traffics for given inpur traffics and link capacity. In addition, it has been found that the average queueing delay of data traffic is longer than the maximum buffer size, when the gain of time assignment speech interpolation(TASI) is more than two and the number of voice users is small.

• MALSORI
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• no.43
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• pp.101-112
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• 2002

In real time packetized voice application, missing packets are major source of voice quality degradation. Thus packet loss concealment (PLC) algorithms are needed to guarantee QoS of VoIP. In this paper, we describe packet loss concealment scheme utilizing the next good frame which follows loss packets. When this scheme is combined with other PLC algorithms, such as G.711 pitch waveform replication recommended by ITU-T LP based PLC algorithm, additional voice quality improvement is obtained for consecutive packet loss larger than 60 msec.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.18 no.1
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• pp.110-122
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• 1993

When voice is transmitted through packet switching network, there needs a overload control, that is, a control for the congestion which lasts short periods and occurrs in local extents. In this thesis, we analyzed the performance of the statistical packet voice multiplexer using the overload control strategy by bit dropping. We assume that the voice is coded accordng to (4,2) embedded ADPCM and that the voice packet is generated and transmitted according to the procedures in the CCITT recomendation G. 764. For the performance analysis, we must model the superposed packet arrival process to the multiplexer as exactly as possible. It is well known that interarrival times of the packets are highly correlated and for this reason MMPP is more suited for the modelling in the viewpoint of accuracy. Hence the packet arrival process in modeled as MMPP and the matrix geometric method is used for the performance analysis. Performance analysis is similar to the MMPP IG II queueing system. But the overload control makes the service time distribution G dependent on system status or queue length in the multiplexer. Through the performance analysis we derived the probability generating function for the queue length and using this we derived the mean and standard deviation of the queue length and waiting time. The numerical results are verified through the simulation and the results show that the values embedded in the departure times and that in the arbitrary times are almost the same. Results also show bit dropping reduces the mean and the variation of the queue length and those of the waiting time.

• Journal of KIISE:Information Networking
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• v.27 no.1
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• pp.68-75
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• 2000

In this paper, a media access control protocol is proposed for voice/data integrated services in the packet CDMA network, and the performance of the proposed protocol is analyzed. The proposed protocol uses the spreading code sensing and the reservation schemes. This protocol gives higher priority to the delay-sensitive voice traffic than to the data traffic. A voice terminal can reserve an available spreading code during a talkspurt to transmit multiple voice packets. On the other hand, whenever a data packet is generated, the data terminal transmits the packet through one of the available spreading codes that are not used by the voice terminals. In this protocol, the voice packets do not come into collision with the data packets. The numerical results show that this protocol can increase the maximum number of voice terminals. The performance for the data traffic degrades by increasing the voice traffic load because of the low priority. But it shows that the data traffic performance can be increased in proportion to the number of spreading codes.

• The Journal of the Acoustical Society of Korea
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• v.28 no.2E
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• pp.35-42
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• 2009

In this paper, we present criteria of objective measurement of speech quality to provide the mobile-VoIP services efficiently over wireless mobile internet. The mobile-VoIP service, which is based on mobility and is error-prone compared to conventional VoIP over wired network, is about to be launched, but there have not been adequate quality indexes and the Quality of Service (QoS) standards for evaluating speech quality of Mobile-VoIP. In addition, there are many factors influencing on the speech quality in packet network of which packet loss contribute directly to the overall voice communication quality. For this reason, we adopt the Gilbert-Elliot Channel Model for modeling packet network based on IP and assess the voice quality through the objective speech method of ITU-T P. 862 PESQ and ITU-T P. 862.1 MOS-LQO under various packet loss rates in the transmission channel environments. Our simulation results address the specific criteria and QoS for the mobile-VoIP services in terms of the various packet loss environments.