[Freeswitch-users] FS SIP audio quality?

[Paul D.]

I re-tested calls to VM replacing some of FS prompts with * ones, and it 
appears that * sounds were recorded with a better quality/higher volume, 
so FS itself has nothing to do with that. That's solved. :-)
I am going to double check all  the equipment we used for tests, like 
headphones, telephone sets, cables since I am almost convinced that 
there's nothing in FS which can produce effects I observe.
I will post back if I find anything wrong, appreciate everybody's help 
with this.

Brian West wrote:
> I'm not able to reproduce this issue.. can you verify the codecs are  
> what you think they are on both Asterisk and FreeSWITCH.
>
> /b
>
> On Feb 15, 2009, at 8:04 PM, Paul D. wrote:
>
>   
>> 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.
>>     
>
>
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