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

[Ivan]

Just found some time to do more tests: in the end it was simply a matter 
of enabling aal2 bitpacking in sofia. I really didn't think that would 
solve the problem since both endpoints had the same problem. (BTW, 
looking back at the interop page with the right set of keywords, I now 
see that someone had the same problem with a linksys SPA2000).

Anyway, thanks to everybody who replied with pointers to bitpacking - I 
assumed wrongly there was no bitpacking at all on G726.

Note: as I'm not knowledgeable enough on G726/bitpacking to update the 
wiki, I've set up a bounty - see the wiki page.



On 08/13/2013 09:57 AM, Ivan wrote:
> Ah, I didn't know there was always bit packing - I guess I should read
> how G726 works, thanks for pointing that.
>
> The thing is, both linksys PAPs and linphone have the same problem, and
> they are not "exotic" endpoints so that makes me think the cause is my
> freeswitch setup. I'll try to test with Xlite when I have a chance, and
> also see if wireshark shows enough codec detail to find out if the
> endpoints get it wrong. I'm of course interested if you have other ideas
> on how to debug that.
>
> ivan
>
>
> On 08/12/2013 05:14 PM, Michael Jerris wrote:
>> There is always bit packing, but there are 2 different ways to do the bit packing.  A lot of devices get it wrong so its worth looking at that.
>>
>> Mike
>>
>> On Aug 4, 2013, at 1:53 AM, Ivan Mitev <imitev at c3i.bg> wrote:
>>
>>> 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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>
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> http://www.freeswitchsolutions.com
>
> 
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