What are the benefits of audio quality testing?
Decide on codecs and technologies used for your VoIP solution,
Fine-tune the codec settings to serve your application needs,
Make sure that voice service performs well, even at the worst conditions (bad network, old devices),
Continuously check for any regressions that may appear,
See how you compare with other competitors when it comes to quality metrics of voice call.
These team's logic is as below
you feed original and degraded audio sample into our tool and set audio sample range to narrowband (sample rate up to 8 kHz, used by PSTN calls and older services) or wideband (sample rate >8 kHz, used by modern HD audio codecs). The tool gives out MOS score of the audio – between 1 and 5 (1 for unacceptable to 5 for excellent). The tool will also return delay if it can be calculated – the samples have the delay included between them (the samples are not synced).
Other better environments for testing the software quality are
LTE network – as mobile network is guaranteed, use P2P connections for calls against Wi-Fi network.
50 ms second jitter and 100 ms delay – while extensive for any normal real life conditions, jitter handling is required for any voice application regardless of jitter amount. Delay for the network is added to be able imitate the jitter.
5% and 10% packet loss – ability to recover from lost data is important part of any modern codec. 5% packet loss should result in negligible quality loss for call, while 10% loss normally is much more noticable and a good landmark for checking codec resistance to a bad network. It’s possible to increase the values for deeper investigation.
50 Kbps bandwidth limitation – from our measurements, the data consumption for wideband VoIP call is around 8 kilobytes per second which translates to 40 Kbps bandwidth. 50 kbps bandwidth is only barely in the limits of unaffected voice quality and is a challenge for voice application using unnecessary background data. Possible to reduce to see the threshold at which service still functions.
References
https://www.testdevlab.com/blog/2017/12/how-we-test-audio-quality-in-voip-applications/
Decide on codecs and technologies used for your VoIP solution,
Fine-tune the codec settings to serve your application needs,
Make sure that voice service performs well, even at the worst conditions (bad network, old devices),
Continuously check for any regressions that may appear,
See how you compare with other competitors when it comes to quality metrics of voice call.
These team's logic is as below
you feed original and degraded audio sample into our tool and set audio sample range to narrowband (sample rate up to 8 kHz, used by PSTN calls and older services) or wideband (sample rate >8 kHz, used by modern HD audio codecs). The tool gives out MOS score of the audio – between 1 and 5 (1 for unacceptable to 5 for excellent). The tool will also return delay if it can be calculated – the samples have the delay included between them (the samples are not synced).
Other better environments for testing the software quality are
LTE network – as mobile network is guaranteed, use P2P connections for calls against Wi-Fi network.
50 ms second jitter and 100 ms delay – while extensive for any normal real life conditions, jitter handling is required for any voice application regardless of jitter amount. Delay for the network is added to be able imitate the jitter.
5% and 10% packet loss – ability to recover from lost data is important part of any modern codec. 5% packet loss should result in negligible quality loss for call, while 10% loss normally is much more noticable and a good landmark for checking codec resistance to a bad network. It’s possible to increase the values for deeper investigation.
50 Kbps bandwidth limitation – from our measurements, the data consumption for wideband VoIP call is around 8 kilobytes per second which translates to 40 Kbps bandwidth. 50 kbps bandwidth is only barely in the limits of unaffected voice quality and is a challenge for voice application using unnecessary background data. Possible to reduce to see the threshold at which service still functions.
References
https://www.testdevlab.com/blog/2017/12/how-we-test-audio-quality-in-voip-applications/
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