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What tools for testing sound quality in Asterisk ?
Hi,
please tell me if you know professional or basic tools for testing sound quality in Asterisk.
What I mean is set of sample audio (standard wave samples) files, audio spectrum analyzer, FFT, synchro 2+ audio channels mixer / player
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Just recorded “what I hear” from Asterisk + celliax, playing
congrats GSM file + MusicOnHold wav file
and uploaded to sendspace
audio file format is .wav
please play it and let me know your opinion, what is behind poor quality of audio played by Asterisk + celliax ,
via sound card at my laptop
I welcome your comments, as I am trying to develop a number of sample (wave sample) audio files to be played by Asterisk MusicOnHold ( 8 bit, mono, 8000Hz wav format) to watch sound distortion in graphical interface ( audio spectrum analyzer).
However, you don’t need any tools other than ears to tell you that there is severe packet loss, and that you are using a near linear codec, like G.711. You can also see the large number of flat line segments in the Windows Media Player time domain display.
The packet loss will be because of:
processor overload;
network overload;
network traffic shaping limiting the amount of RTP traffic.
If you were trying to run this on a virtual machine, my money would be on processor overload.
[quote=“david55”]I’ve already responded on another thread.
However, you don’t need any tools other than ears to tell you that there is severe packet loss, and that you are using a near linear codec, like G.711. You can also see the large number of flat line segments in the Windows Media Player time domain display.
The packet loss will be because of:
processor overload;
network overload;
network traffic shaping limiting the amount of RTP traffic.
If you were trying to run this on a virtual machine, my money would be on processor overload.
Your attachments site was painful to use.[/quote]
thanks David,
I am trying to learn about more professional tools than my ears, as there is still a chance to get some code corrected
in Celliax audio channel.
As my testing environment is network -off, having Asterisk playing sound directly to a sound card (via Celliax audio channel) to headphone, the last 2 cases can be eliminated
Processor load can be monitored , just tested having Asterisk to play standard MusicOnHold file
and CPU load is in 0-10% range
So what is to my concern is algorithm’s code executed, tx, rx buffer sizes , as Celliax is audio channel for Asterisk,
a middleware between Asterisk and audio card.
Firstly I missed one possible cause, which is jitter buffer/clock frequency discrepancies. That will typically cause regular packet drops.
You really do not need anything better than ears here, although you could use the normal time domain display in Audacity, to see what proportion is missing.
[quote=“david55”]Firstly I missed one possible cause, which is jitter buffer/clock frequency discrepancies. That will typically cause regular packet drops.
You really do not need anything better than ears here, although you could use the normal time domain display in Audacity, to see what proportion is missing.[/quote]
Thanks David,
thanks for your excellent support.
The issue is I am just learning Audacity how to use it.
My Asterisk configuration is desktop only, laptop running Asterisk, no network, no sip client, no packets over the ethernet.
Asterisk is playing directly to Celliax audio channel module and I record sound from sound card’s line out -> headphone.
In another configuration, X-Lite softphone SIP client , registered to Asterisk’s server is running on the same laptop (that’s all, no packets in network).
And quality of audio files played by Asterisk via SIP softphone X-Lite client is fine, sound is clear, not distorted ( no jitter or alike).
So I just exactly need to play audio samples as MusicOnHold and record played audio in Audacity, looking for jitter , echo and other distortions in audio samples.
Although there may be bugs in Asterisk transcoding, sound quality issues are most likely to be due to fundamental characteristics of the codecs, rather than anything to do with asterisk.
Most VoIP codecs, and all mobile phone ones, are specifically designed for voice, so a chirp is unlikely to be a valid test.
The passband for telephony is 300 to 3,400 Hz.
What is your exact chain of transducers, codecs and transcoders. I’m not familiar with “Celiax” I don’t really know what to expect from it.
Also, please describe the technical results in words, as downloading files, especially when one hast to negotiate Captcha, and then analyzing them, is a hassle.
Asterisk is ok, the problem looks like to be with Celliax - audio channel for Asterisk, redirecting audio from Asterisk, playing MOH files to a sound card.
Just uploaded wav files to another web / ftp server.
No more captcha, images, download problems.
I plan to make more tests today, generating more audio files by Audacity -
sine, square, sawtooth tones, silence, noise, DTMF .
Once again.
Problem is not with Asterisk itself, problem is with audio channel module for Asterisk, Celliax, intended to connect GSM Gateway to Asterisk server.
On other hand, I am planning to develop a library of sound samples for testing sound quality in SIP VoIP environments.