[Freeswitch-users] FS SIP audio quality?

[Anthony Minessale]

The typing it takes to start a pcap of each call and email them is less than
you have typed thusfar.
Please just take the captures and send them to us to examine. That's all. If
you have a real issue we would like to address it.

On Feb 15, 2009 8:06 PM, "Paul D." <pauld at versafon.com> 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.

Anthony Minessale wrote: > it's digital audio. The only thing doing sampling
and reconstruction ...

> <mailto:pauld at versafon.com>> wrote: > > Comparing FS 1.0.3 audio quality
vs * 1.4.2, simple Si...
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