[Freeswitch-users] FS SIP audio quality?

[jay binks]

another thing to try here...
is to put FS in RTP proxy and bypass mode.

http://wiki.freeswitch.org/wiki/Bypass_Media

it would be interesting to see if your still experiencing this problem in
either of those 2 modes.

Jay

On Mon, Feb 16, 2009 at 12:04 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
> > of the signal are the phones.  The audio files have been captured long
> > ago from the microphone in the studio.
> > We do nothing to alter the volume of the audio signal or manipulate it
> > in any way unless you are transcoding between sample rates or codecs
> > which you are not because you mentioned it was PCMU.
> >
> > If you are making a call from x-lite to a linksys using just PCMU
> > there is no transcoding going on at all and it would not be any more
> > or less loud than if the
> > devices were exchanging media directly because all we would be doing
> > is passing the digital packets across.
> >
> > I believe you are somehow mistaken in your explanation.  There is a
> > good chance that your x-lite has the gain set lower when you are
> > testing FS since that's the only device
> > in your whole scenario that is capable of adjusting the gain.
> >
> > If you wish, please get a complete packet capture of a completed call
> > in both situations.
> >
> >
> > On Sat, Feb 14, 2009 at 8:37 PM, Paul D. <pauld at versafon.com
> > <mailto:pauld at versafon.com>> 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.
> >
> >     _______________________________________________
> >     Freeswitch-users mailing list
> >     Freeswitch-users at lists.freeswitch.org
> >     <mailto:Freeswitch-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest@conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:213-799-1400
> > ------------------------------------------------------------------------
> >
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> >
>
>
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-- 
Sincerely

Jay
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