[Freeswitch-users] FS SIP audio quality?

[Paul D.]

I am not sure what else I can add to that, I would love to elaborate 
more if you ask anything specific.
I haven't tried the latest trunk, but since there's no difference 
between 1.0.2 and 1.0.3RC1 in audio quality I don't think
it make sense trying. From what I see in FS logs there's no resampling 
involved, and that looks like true since I specifically restricted 
codecs in my test SIP equipment.
But the fact is I tried different boxes, same OS centos 5.2 x64, and I 
had to bring audio volume and mic level all the way up in X-Lite to 
compensate for the difference to * audio,
and in * such volume level sounds like way too high.
FS installed cleanly from scratch, mostly default settings, except some 
dialplan/directory additions.

Brian West wrote:
> I haven't ever experienced this issue can you maybe elaborate on the  
> issue a little more?  We usually hear that the audio quality is much  
> better... have you tried latest SVN trunk?  If resampling was involved  
> it might cause some audio issues but those were usually gain issue and  
> that has since been fixed in SVN trunk as of yesterday.
>
> /b
>
> On Feb 14, 2009, at 8:37 PM, Paul D. 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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