• The KIPS Transactions:PartC
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• v.19C no.2
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• pp.119-126
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• 2012

This paper covers the service quality of packet-based voice service that is provided over wireless broadband (WiBro) and high speed downlink packet access (HSDPA) systems. Using a measurement software that has been developed in the course of preparing a advanced service quality management scheme for the packet-based voice service over wireless networks, a huge scale of experiment is conducted to measure the real quality of the voice service. Based on our analysis of the measurement results, the service quality of the voice service is supposed to be quite good over both wireless systems. In addition, another experiment to investigate the effect of degradation of wireless transmission conditions on the service quality of the voice service shows the values of wireless service metrics in which mean opinion score (MOS) starts to decrease.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.2
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• pp.249-255
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• 2012

This paper covers the service quality of packet-based voice service that is provided over a high speed downlink packet access (HSDPA) system. Using the measurement software that has been developed in the course of preparing a advanced service quality management scheme for the packet-based voice service over a wireless network[2][3], a huge scale of experiment is conducted to measure the real quality of the voice service. Based on our analysis of the measurement result, the service quality of the voice service is supposed to be quite good over HSDPA system. In addition, another experiment to investigate the effect of degradation of wireless transmission conditions on the service quality of the voice service shows the values of wireless service metrics in which mean opinion score (MOS) starts to decrease.

• 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 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.12B
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• pp.2109-2119
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• 2000

In this paper, the Packet CDMA Reservation ALOHA protocol is proposed to support the multi-traffic services such as voice and videophone services with handoff calls, high-rate data and low-rate data services efficiently on the multi-rate transmission in uplink cellular systems. The frame structure, composed of the access slot and the transmission slot, and the proposed access permission probability based on the estimated number of contending users for each service are presented to reduce MAI. The assured priority to the voice and the videophone handoff calls is given through higher access permission probability. And through the proposed code assignment scheme, the voice service can be provided without the voice packet dropping probability in the CDMA/PRMA protocols. The code reservation is allowed to the voice and the videophone services. The low-rate data service uses the available codes during the silent periods of voice calls and the remaining codes in the codes assigned to the voice service to utilize codes efficiently. The high-rate data service uses the assigned codes to the high-rate data service and the remaining codes in the codes assigned to the videophone service. Using the Markov-chain subsystem model for each service including the handoff calls in uplink cellular systems, the steady-state performances are simulated and analyzed. After a round of tests for the examples, through the proposed code assignment scheme and the access permission probability, the Packet CDMA Reservation ALOHA protocol can guarantee the priority and the constant QoS for the handoff calls even at large number of contending users. Also, the data services are integrated efficiently on the multi-rate transmission.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.4
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• pp.1006-1025
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• 2012

Improving the quality of service in IP networks is a major challenge for real-time voice communications. In particular, packet arrival-delay variation, so-called "jitter," is one of the main factors that degrade the quality of voice in mobile devices with the voice-over Internet protocol (VoIP). To resolve this issue, a receiver-based enhanced timing recovery algorithm combined with active jitter estimation is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. Extensive simulations have shown that the proposed algorithm delivers high voice quality by pursuing an optimal trade-off between average buffering delay and packet loss rate.

• The Journal of the Korea institute of electronic communication sciences
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• v.6 no.6
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• pp.823-829
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• 2011

This paper covers the service quality of packet-based voice service that is provided over a wireless broadband (WiBro) network. Using a measurement software that has been developed in the course of preparing a advanced service quality management scheme for the packet-based voice service over a wireless network[2][3], a huge scale of experiment is conducted to measure the real quality of the voice service. Based on our analysis of the measurement result, the service quality of the voice service is supposed to be quite good over WiBro networks. In addition, another experiment to investigate the effect of degradation of wireless transmission conditions on the service quality of the voice service shows the values of wireless service metris in which mean opinion score (MOS) starts to decrease.

• 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.

• Journal of Korea Multimedia Society
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• v.6 no.7
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• pp.1219-1230
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• 2003

We analyze an integrated voice/data CDMA system, where the whole channels are divided into voice prioritized channels and voice non-prioritized channels. For real-time voice service, a preemptivc priority is granted in the voice prioritized channels. And, for delay-tolerant data service, the employment of buffer is considered. On the other hand, the transmission permission probability in best-effort packet-data service is controlled by estimating the residual capacity available for users. We build a 2-dimensional markov chain about prioritized-voice and stream-data services and accomplish numerical analysis in combination with packet-data traffic based on residual capacity equation.

• Journal of Communications and Networks
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• v.11 no.1
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• pp.36-41
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• 2009

The IEEE 802.16 wireless metropolitan area network (WMAN) standard is designed to correct expensive communication costs in CDMA-based mobile communication systems and limited coverage problems in wireless LAN systems. Thus, the IEEE 802.16e standard can provide mobile high-speed packet access between mobile stations and the Internet service provider through the base station with cheap communication fees. To efficiently accommodate voice over IP (VoIP) services in IEEE 802.16 systems, an uplink quality of service packet-scheduling scheme is proposed, and its performance is evaluated with an NS-2 network simulator in this paper. Numerical results show that this proposed scheme can increase the system capacity by 100% more than in the unsolicited rand service (UGS) scheme and 30% more than the extended real-time polling service (ertPS) scheme, respectively.