[Freeswitch-users] FS SIP audio quality?

[Brian West]

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.