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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>You can turn off G726 AAC bit-packing in spandsp.conf.xml.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>By the way, there are other codecs out there you can try. SPEEX comes to mind if all your endpoints don’t deal with the PSTN.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div style='border:none;border-left:solid blue 1.5pt;padding:0in 0in 0in 4.0pt'><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] <b>On Behalf Of </b>Brian Foster<br><b>Sent:</b> Saturday, August 3, 2013 2:37 PM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] garbled audio with G726-32, other codecs are fine<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p>AAC bitpacking by any chance? I thought I had a similar issue, happened so long ago I cant remember what I did.<o:p></o:p></p><p>Thank you,<o:p></o:p></p><p>Brian Foster<br>Project Manager/Owner's Rep.<br>Davri Investments, Inc.<br>O: 317-787-2686 x2102<br>M: 317-600-9753<br>E: <a href="mailto:bdfoster@davri.com">bdfoster@davri.com</a><br>Indianapolis, Indiana<o:p></o:p></p><p>Sent from a mobile device.<o:p></o:p></p><div><p class=MsoNormal>On Aug 3, 2013 5:20 PM, "Ivan Mitev" <<a href="mailto:imitev@c3i.bg">imitev@c3i.bg</a>> wrote:<o:p></o:p></p><p class=MsoNormal>Hello<br><br>I'm migrating an office setup from asterisk to FS and in the process I<br>was considering using G726-32 for some bandwidth starved remote<br>endpoints. However I only get metallic/garbled audio with that codec<br>even when simply playing moh to the endpoint, while other codecs work<br>fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds<br>marginally better but still garbled and really worse than G711.<br><br>The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest<br>(centos6 64bit host). But please don't shoot ! :) - I know about virtual<br>environment limitations but for these tests the host is only lightly<br>loaded, there aren't any calls to the FS instance except my tests, and<br>the fact that it works with other codecs makes me think that<br>virtualization is not the issue here. I may be wrong though.<br><br>Is there any guide for debugging that kind of problem before reverting<br>to a fresh install on bare-metal with the latest HEAD ? Until now I've<br>tried:<br><br>- improving timers ; but the default soft timer (which I guess uses<br>timerd) works best. The time interval between sent packets on a tcpdump<br>trace looks identical to the output of "timer_test", so that doesn't<br>seem to be a network/jitter problem. And there's no problem with other<br>codecs, but maybe G726-XX is specific. For info the guest's clocksource<br>is kvm_clock, while the host uses tsc.<br><br>- using different endpoints: the production ones are Linksys PAP2<br>("fixed" for 20ms psize, and G726-32 SDP type indentification), but the<br>same thing happens with linphone on a fedora 19 laptop.<br><br>A call with rtp media going through FS without transcoding - G726-32 to<br>G726-32 - works perfectly (I can't hear the difference with G711). The<br>problem is only when there's transcoding to G726 (from wav for moh, or<br>from any other codec when bridging). I've looked at the wiki, posts,<br>changelogs, jira, ..., but am a bit at a loss now.<br><br>Any pointers ?<br><br>Except that little problem, FS rocks, and I'm happy I can finally ditch<br>asterisk. Kudos to the core devs and contributors.<br><br>Ivan<br><br><br>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services:<br><a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br><a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br><br>FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br><a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br><br>Official FreeSWITCH Sites<br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br><a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br><a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br><br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><o:p></o:p></p></div></div></div></body></html>