another thing to try here... <br>is to put FS in RTP proxy and bypass mode.<br><br><a href="http://wiki.freeswitch.org/wiki/Bypass_Media">http://wiki.freeswitch.org/wiki/Bypass_Media</a><br><br>it would be interesting to see if your still experiencing this problem in either of those 2 modes.<br>
<br>Jay<br><br><div class="gmail_quote">On Mon, Feb 16, 2009 at 12:04 PM, Paul D. <span dir="ltr"><<a href="mailto:pauld@versafon.com">pauld@versafon.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Well, I tried several call scenarios:<br>
1. Call from X-Lite or Linksys to VM.<br>
2. Call from X-Lite or Linksys to a conference.<br>
3. Call from X-Lite or Linksys to a PSTN number via Gafachi and CallWithUs.<br>
<br>
I have now * 1.6.5 and FS 1.0.3RC1 installed on the same enterprise<br>
grade Intel server. So just comparing audio in the call scenarios above<br>
* somehow does noticeably better job, sounds clearer and volume is at<br>
the right level. I am not changing any phone settings of course when<br>
switching between * and FS.<br>
I am not biased towards FS or * at the moment, though FS seems to have a<br>
better designed configuration options and community.<br>
Just wanted to share my experience, and hear some opinions.<br>
Unfortunately I cannot spend whole amount of time investigating this<br>
case now, capturing packets etc., but I will try to do that once I have<br>
time. Meanwhile I will have to stick to * for prod.<br>
<div class="Ih2E3d"><br>
<br>
Anthony Minessale wrote:<br>
> it's digital audio. The only thing doing sampling and reconstruction<br>
> of the signal are the phones. The audio files have been captured long<br>
> ago from the microphone in the studio.<br>
> We do nothing to alter the volume of the audio signal or manipulate it<br>
> in any way unless you are transcoding between sample rates or codecs<br>
> which you are not because you mentioned it was PCMU.<br>
><br>
> If you are making a call from x-lite to a linksys using just PCMU<br>
> there is no transcoding going on at all and it would not be any more<br>
> or less loud than if the<br>
> devices were exchanging media directly because all we would be doing<br>
> is passing the digital packets across.<br>
><br>
> I believe you are somehow mistaken in your explanation. There is a<br>
> good chance that your x-lite has the gain set lower when you are<br>
> testing FS since that's the only device<br>
> in your whole scenario that is capable of adjusting the gain.<br>
><br>
> If you wish, please get a complete packet capture of a completed call<br>
> in both situations.<br>
><br>
><br>
> On Sat, Feb 14, 2009 at 8:37 PM, Paul D. <<a href="mailto:pauld@versafon.com">pauld@versafon.com</a><br>
</div><div class="Ih2E3d">> <mailto:<a href="mailto:pauld@versafon.com">pauld@versafon.com</a>>> wrote:<br>
><br>
> Comparing FS 1.0.3 audio quality vs * 1.4.2, simple Sip-to-Sip<br>
> call, or<br>
> call to VM prompt, or call via gateway to PSTN - FS audio volume<br>
> level<br>
> (should I say gain?) seems noticeably lower than on *, this may be a<br>
> reason that FS audio seems to be subpar, more noise less clear. Test<br>
> calls made using PCMU codec from X-Lite and Linksys 2002.<br>
> Is there anything can be tweaked in FS to correct that? Same issue was<br>
> with 1.0.2.<br>
><br>
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> --<br>
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><br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Sincerely<br><br>Jay<br>