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

[Ivan Mitev]

Thanks for the suggestion but I'm testing with G726-32, not AAL2-G726-32 
; so bitpacking shouldn't be used. By the way when I tested with AAL2 to 
the linphone client I only got cracks and whitenoise, I've forgot to 
mention that in my post.

That said I've tried to uncomment and set <param name="bitpacking" 
value="none"/> in internal.xml ("none" is a wild guess - I couldn't find 
any doc on values accepted by this parameter), but that doesn't help.

Speex: the ATAs don't support it. And being stubborn I'd like to 
understand what's wrong with G726 :)


On 08/04/2013 05:22 AM, Jeff Leung wrote:
>
> 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 <mailto:bdfoster at davri.com>
> Indianapolis, Indiana
>
> Sent from a mobile device.
>
> On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev at c3i.bg 
> <mailto: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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