[Linphone-users] Sync problem with Asterisk??

[John A. Sullivan III]

First of all, thanks to all who produce this nice product.  I am
absolutely brand new to not only IP Telephony but Telephony in general. 
This new asterisk box I have stayed away all day Friday and Saturday to
get running so our scattered open source teams
(http://iscs.sourceforge.net) can freely communicate is my very first
PBX.

In my ignorance, I do not know what is related so I will recount all my
grief in the hope that you gurus will see the common thread and point me
in the right direction :-)

I first started with KPhone.  All seemed fine although the received
sound was a little poor.  However, the transmitted channel is completely
full of noise - so much so that it makes conference calls and voice
mails impossible.  Loud, loud static and buzz - almost like a bad wire.

That's when I scoured the Internet to find all kinds of broken
implementations until I came across Linphone!!!! At first, other than
the interface shortcomings of such a young product, I was astounded. 
The sound was so clear and once I got the DTMF setting right, all seemed
paradise.

Then I decided to try to integrate Speex since Linphone spoke so highly
of it and we have some developing world developers who are on limited
bandwidth.  The initial sound was wonderful but then we got streams of
buffer errors in Asterisk.  We assumed that is not a good thing :-)  So
we disabled speex.

We went back to ulaw, alaw, ilbc and gsm.  I thought all was fine until
I started retrieving voice mails.  It almost seems like the asterisk PBX
and Linphone go out of sync.  The voice starts to break up and then
there is no sound at all.  I can see asterisk playing the messages.  I
can send commands from Linphone via DTMF which are honored by Asterisk
but there is no sound.  If I hang up the call and dial again, the sound
is perfect until I start retrieving voice mail.  I must state that I
have not had anyone to test on the other end yet so it could be that,
given enough voice traffic, we would have the same problem.

I then thought I would play with the jitter buffer and changed it from
60 to 100 ms.  To my horror, I had the same noisy, broken sound as I had
in KPhone.

What do I do?
1) Is there a way to use speex, Linphone and Asterisk without the buffer
problem?
2) How do I fix this suspected synchronization/overrun problem?
3) Why does changing the jitter buffer create such awful noise and
broken voice?

By the way, at your suggestion, I replaced all my Intel i810 drivers
with Alsa drivers.  Thanks - John

PS - here are some console message from Linphone.  I do not know if they
are significant but they look unhealthy:

** Message: Sending dtmf 1
MediaStreamer-Message: Sending DTMF.
** Message: Sending dtmf 1
MediaStreamer-Message: Sending DTMF.
** Message: Sending dtmf 1
MediaStreamer-Message: Sending DTMF.
| INFO1 | <nict_callbacks.c: 30> Transaction 11 killed.

| INFO1 | <osipdialog.c: 1918> Dialog is removed. It remains 1 dialog(s)
in the ua list.

** Message: Sending dtmf 2
MediaStreamer-Message: Sending DTMF.
** Message: Sending dtmf 1
MediaStreamer-Message: Sending DTMF.
** Message: Sending dtmf 1
MediaStreamer-Message: Sending DTMF.
** Message: Sending dtmf 1
MediaStreamer-Message: Sending DTMF.
** Message: Sending dtmf 2
MediaStreamer-Message: Sending DTMF.
MediaStreamer-Message: Mediastreamer processing thread is exiting.
oRTP-stats-Message:
   Global statistics :
 packet_sent=10328
 sent=1510937 bytes
 packet_recv=6564
 hw_recv=1052173 bytes
 recv=520005 bytes
 unavaillable=10235 bytes
 outoftime=532168
 bad=0
 discarded=0

-- 
John A. Sullivan III
Open Source Development Corporation
Financially sustainable open source development
http://www.opensourcedevel.com