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> 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 > 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> GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<paypal%3aanthony.miness...@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org <sip%3a...@conference.freeswitch.org> iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org<googletalk%3aconf%2b...@conference.freeswitch.org> pstn:213-799-1400
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