[Freeswitch-users] garbled audio with G726-32, other codecs are fine

[Jeff Leung]

You can turn off G726 AAC bit-packing in spandsp.conf.xml.

 

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.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
Foster
Sent: Saturday, August 3, 2013 2:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] garbled audio with G726-32, other codecs are
fine

 

AAC bitpacking by any chance? I thought I had a similar issue, happened so
long ago I cant remember what I did.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.

On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev at c3i.bg> wrote:

Hello

I'm migrating an office setup from asterisk to FS and in the process I
was considering using G726-32 for some bandwidth starved remote
endpoints. However I only get metallic/garbled audio with that codec
even when simply playing moh to the endpoint, while other codecs work
fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds
marginally better but still garbled and really worse than G711.

The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest
(centos6 64bit host). But please don't shoot ! :) - I know about virtual
environment limitations but for these tests the host is only lightly
loaded, there aren't any calls to the FS instance except my tests, and
the fact that it works with other codecs makes me think that
virtualization is not the issue here. I may be wrong though.

Is there any guide for debugging that kind of problem before reverting
to a fresh install on bare-metal with the latest HEAD ? Until now I've
tried:

- improving timers ; but the default soft timer (which I guess uses
timerd) works best. The time interval between sent packets on a tcpdump
trace looks identical to the output of "timer_test", so that doesn't
seem to be a network/jitter problem. And there's no problem with other
codecs, but maybe G726-XX is specific. For info the guest's clocksource
is kvm_clock, while the host uses tsc.

- using different endpoints: the production ones are Linksys PAP2
("fixed" for 20ms psize, and G726-32 SDP type indentification), but the
same thing happens with linphone on a fedora 19 laptop.

A call with rtp media going through FS without transcoding - G726-32 to
G726-32 - works perfectly (I can't hear the difference with G711). The
problem is only when there's transcoding to G726 (from wav for moh, or
from any other codec when bridging). I've looked at the wiki, posts,
changelogs, jira, ..., but am a bit at a loss now.

Any pointers ?

Except that little problem, FS rocks, and I'm happy I can finally ditch
asterisk. Kudos to the core devs and contributors.

Ivan


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