[asterisk-users] Bad audio quality in 1.4-SVN when encoding to alaw/ulaw

[Bernhard Schmidt]

Hi,

after some years of abstinence from bigger Asterisk installations I was
trying to give it a start again with the asteriskv6.org project. I ran
into some issues regarding the audio quality and tried the vanilla 1.4
code, with the same results.

I have asterisk (latest 1.4 branch checkout from SVN) running on my home
server (Linux 2.6.22.11, two HFC BRI cards with latest mISDN from git).
I have had very bad call quality as soon as Asterisk needs to encode
something into alaw/ulaw (choppy sound). I tried two different clients
(Snom 360 hardphone and ekiga on my workstation), the network in between
is fine. 

Snom360 ---- SIP/alaw ---- Asterisk voiceprompts (demo application)
           <---------------- bad
	   ----------------> unknown

Snom360 ---- SIP/gsm  ---- Asterisk voiceprompts (demo application)
           <---------------- good
           ----------------> unknown

Snom360 ---- SIP/alaw ---- Asterisk ---- IAX/gsm ---- Digium (demo)
           <---------------- bad
	   ----------------> unknown

Snom360 ---- SIP/gsm  ---- Asterisk ---- IAX/gsm ---- Digium (demo)
           <---------------- bad
	   ----------------> unknown

Snom360 ---- SIP/alaw ---- Asterisk ---- SIP/alaw ---- ekiga
           <---------------> good (both with reinvite and without)

Snom360 ---- SIP/alaw ---- Asterisk ---- SIP/ulaw ---- ekiga
           <---------------> bad
 

Snom360 ---- SIP/gsm  ---- Asterisk ---- mISDN/(alaw?)
	   <---------------- good
           ----------------> bad

Snom360 ---- SIP/alaw ---- Asterisk ---- mISDN/(alaw?)
           <---------------> good

for all tests where Snom360 was the only client on my side you can use
ekiga and get the exact same results, so this rules out a problem with
Snom. You can use ulaw instead of alaw without changes.

The pattern here is quite clear, as soon as Asterisk needs to encode
alaw/ulaw, sound gets bad.

Does anyone have any clue what might be causing this?

Thanks,
Bernhard