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. <pa...@versafon.com > <mailto:pa...@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@lists.freeswitch.org > <mailto:Freeswitch-users@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_miness...@hotmail.com > <mailto:msn%3aanthony_miness...@hotmail.com> > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com > <mailto:paypal%3aanthony.miness...@gmail.com> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > sip:8...@conference.freeswitch.org > <mailto:sip%3a...@conference.freeswitch.org> > iax:gu...@conference.freeswitch.org/888 > <http://iax:gu...@conference.freeswitch.org/888> > googletalk:conf+...@conference.freeswitch.org > <mailto:googletalk%3aconf%2b...@conference.freeswitch.org> > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org