A Study of the Relationship of the Packet Loss and the Speech Quality in VoIP Networks
碩士 === 大同大學 === 資訊工程學系(所) === 102 === With the rapid development of network and digital signal processing technology, we can see an obvious coverage expansion trend of 3G phones and IP networks. However, when using a 3G mobile phone to make an VoIP calls, the call quality is not very stable, and how...
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| Format: | Others |
| Language: | zh-TW |
| Published: |
2014
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| Online Access: | http://ndltd.ncl.edu.tw/handle/85499661865659413709 |
| Summary: | 碩士 === 大同大學 === 資訊工程學系(所) === 102 === With the rapid development of network and digital signal processing technology, we can see an obvious coverage expansion trend of 3G phones and IP networks. However, when using a 3G mobile phone to make an VoIP calls, the call quality is not very stable, and how to overcome this problem has become a major challenge for network communication industries.
In this research, we try to find the relationship between the packet loss rate and the voice quality of the VoIP call. The experiment is setup with Asterisk IP-PBX and network simulation tool netem by varying the packet loss rate. The voice quality MOS is measured by using the industrial standard method. We also invited 12 annotators to subjectively justify the call quality by listen test according to the guideline of ITU-T standard for audio signal. Then we compared the results from the annotators and the results from the algorithm we used. The rule proposed in this study is based on the ITU-T standard for audio signal. The score is used as a reference score for the audio quality measurement. From the experimental results, the two scores are close when the packet loss rate is under 10%.
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