[Asterisk-Users] voice choppy

[Luke Catranis]

Nearly everyone of my clients is between 600 and 700 ms and there is no
choppyness to the calls. They are on VSAT uplinks with SCPC return channels,
but I would say that latency is certainly not your problem. 




-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Dana Nowell
Sent: Friday, August 13, 2004 2:22 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] voice choppy

OK, background/config.

running * (show version reports 0.9.0) on Mandrake 9.2 (kernel:
2.4.22-32mdk) with a dual 800mhz PIII with 256M Ram 4port FXO digium card,
no IRQ sharing I can find (cat /proc/pci & cat /proc/interrupts), vmstat
reports a minimum of 80+% CPU idle when problem occurs.

connect to a Grandstream 101 (GS) via vpn (no nat).  Link has 100ms - 150ms
ROUND TRIP latency (constant 'ping' during test).  Codec is alaw or ulaw.
TDM card is plugged to an NEC PBX (old NEAX 2000 IVS) via the analog
station card in the PBX.

PROBLEM:
Establish a call from the GS to an NEC phone (Dterm III) connected to the
PBX.  The voice quality on the GS sounds good.  The voice quality on the
NEC gets very choppy (random).  A call from the same GS to an internal GS
here (same latency, same IP path) is much better (some chop).

SOLUTIONS tried to date:
Tried a local network test, sound excellent.

Tried a 'low latency vpn' test bed to reduce latency (is latency the
issue?).  Latency down to ~70ms round trip, better but still noticeably
choppy.  

Tried different codecs (ulaw, alaw, iLBC), no impact.

Tried various echo cancel/train on/off values, cancel and train on seem
best.

Tried modifing Samples/TX packet changes, 2 and 64 tried as 'ends of
spectrum' values.  little noticable impact (2 seems marginally better,
still clips, MAY be less often).

Tried changing TXgain and RXgain under the assumption audio level clipping
occurs between tdm and NEC, some improvement.

Substituted an SJPhone on a PC instead of the GS, better, still choppy.

Tried the 'demo speech' on an extension, called from Dterm, sound is
excellent, tried GS over low latency vpn, choppy, tried SJ over low latency
vpn, choppy.  (hmmmm, does that eliminate the tdm to NEC link as an issue?)

Tried internal GS over low latency vpn to SJ (out thru vpn to * back thru
vpn to sj), choppy.

Current state:
I've improved from 'crappy cell call about to disconnect' to 'average to
crappy cell call' quality.  Not exactly ready for customer use.

ASSUMPTIONS:
Original assumption was latency, but latency was still below the '150MS one
way' values I see quoted for good sound quality.  In fact over the low
latency vpn the values were less than a third the quoted value.  Sometimes
I get a high latency and get a drop out, OK expected.  Sometimes I get chop
at a lower latency than I was getting with 'good' sound.

Second assumption was interface between NEC and TDM was clipping, after
changes to tx/rx gain the 'demo' speech sounds fine on the Dterm. (Does
this remove the interface from the possible bad guys list?)

HELP Needed:
So given that latency is in the 70 - 100 ms round trip. Given that I've
diddled rx/tx gain.  Given that I've tried basic echo on/off settings.
What's next?  

Is the 'common' 150 ms one way (150+ms round trip) value a bunch of crap?
Do I need some other magic latency goal?  

Would a 729 codec help?  

Is there a test I missed?  Some other values to twiddle?  

Is this a 'known issue' I don't know about, fixed in some more recent
version?  (Yeah, I can just try the CVS-HEAD and hope to get lucky and I
probably will as I'm out of options/ideas but KNOWING it is fixed is better
than hoping it is fixed)

I'm stuck and my forehead is getting flat from pounding it on the wall.
Anyone handing out clues?



-- 
Dana Nowell     Cornerstone Software Inc.
Voice: 603-595-7480 Fax: 603-882-7313
email: DanaNowell_at_CornerstoneSoftware.com
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