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. <pa...@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. <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 <msn%3aanthony_miness...@hotmail.com> > > <mailto:msn%3aanthony_miness...@hotmail.com<msn%253aanthony_miness...@hotmail.com> > > > > GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com> > > <mailto:paypal%3aanthony.miness...@gmail.com<paypal%253aanthony.miness...@gmail.com> > > > > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch > > > > FreeSWITCH Developer Conference > > sip:8...@conference.freeswitch.org <sip%3a...@conference.freeswitch.org> > > <mailto:sip%3a...@conference.freeswitch.org<sip%253a...@conference.freeswitch.org> > > > > iax:gu...@conference.freeswitch.org/888 > > <http://iax:gu...@conference.freeswitch.org/888> > > googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org> > > <mailto:googletalk%3aconf%2b...@conference.freeswitch.org<googletalk%253aconf%252b...@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 > -- Sincerely Jay
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