[Freeswitch-users] FS SIP audio quality?

[Paul D.]

Well, I tried several call scenarios:
1. Call from X-Lite or Linksys to VM.
2. Call from X-Lite or Linksys to a conference.
3. Call from X-Lite or Linksys to a PSTN number via Gafachi and CallWithUs.

I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise 
grade Intel server. So just comparing audio in the call scenarios above 
* somehow does noticeably better job, sounds clearer and volume is at 
the right level. I am not changing any phone settings of course when 
switching between * and FS.
I am not biased towards FS or * at the moment, though FS seems to have a 
better designed configuration options and community.
Just wanted to share my experience, and hear some opinions. 
Unfortunately I cannot spend whole amount of time investigating this 
case now, capturing packets etc., but I will try to do that once I have 
time. Meanwhile I will have to stick to * for prod.


Anthony Minessale wrote:
> it's digital audio.  The only thing doing sampling and reconstruction 
> of the signal are the phones.  The audio files have been captured long 
> ago from the microphone in the studio.
> We do nothing to alter the volume of the audio signal or manipulate it 
> in any way unless you are transcoding between sample rates or codecs 
> which you are not because you mentioned it was PCMU.
>
> If you are making a call from x-lite to a linksys using just PCMU 
> there is no transcoding going on at all and it would not be any more 
> or less loud than if the
> devices were exchanging media directly because all we would be doing 
> is passing the digital packets across.
>
> I believe you are somehow mistaken in your explanation.  There is a 
> good chance that your x-lite has the gain set lower when you are 
> testing FS since that's the only device
> in your whole scenario that is capable of adjusting the gain.
>
> If you wish, please get a complete packet capture of a completed call 
> in both situations.
>
>
> On Sat, Feb 14, 2009 at 8:37 PM, Paul D. <pauld at versafon.com 
> <mailto:pauld at versafon.com>> wrote:
>
>     Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip
>     call, or
>     call to VM prompt, or call via  gateway to PSTN - FS audio volume
>     level
>     (should I say gain?) seems noticeably lower than on *, this may be a
>     reason that FS audio seems to be subpar, more noise less clear. Test
>     calls made using PCMU codec from X-Lite and Linksys 2002.
>     Is there anything can be tweaked in FS to correct that? Same issue was
>     with 1.0.2.
>
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> -- 
> Anthony Minessale II
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