<p dir="ltr">AAC bitpacking by any chance? I thought I had a similar issue, happened so long ago I cant remember what I did.</p>
<p dir="ltr">Thank you,</p>
<p dir="ltr">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</p>
<p dir="ltr">Sent from a mobile device.</p>
<div class="gmail_quote">On Aug 3, 2013 5:20 PM, "Ivan Mitev" <<a href="mailto:imitev@c3i.bg">imitev@c3i.bg</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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>
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