[Asterisk-Users] Re: Meetme and Sipura SPA-941-badjitter/distortion

[Ryan Booz]

Dan, thank you for the pointer.  I read through the whole thing and will
potentially try this next week.  I'll post back with any thoughts.

Thanks!

Ryan Booz
Director of IT
Good Steward Software, LLC
111 Sowers Street, Suite 400
State College, PA 16801
Phone: 877-327-3702 x.26 (814-237-3744 x.26)
Fax: 719-623-0577
Visit us at www.energycap.com
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dan Austin
Sent: Thursday, December 08, 2005 6:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Re: Meetme and Sipura
SPA-941-badjitter/distortion

> It might be.  I'm going to work with one of the remote users again
tomorrow
> to see if we can get it working better.  You're also right that the
PSTN
> calls don't hear the echo, INSTEAD I hear a faint "static/waves on a
beach"
> sound whenever I talk though a PSTN set through the system to this
user.
> Pushing the packet size back to .03 makes direct calls better, but
then
> MeetMe goes screwy again.  ARG!  :-)


> Anyone have experience with the mentioned fix at:
> http://bugs.digium.com/view.php?id=5374 and Asterisk 1.2?  Does it
make call
> quality difference with SIP?  I read the whole thing thinking it was
going
> to end up saying this was a 1.2 feature, but looks like it got pushed
to
> 1.3.  Thoughts?

That patch and bug does help quite a few scenarios, but they won't help
with this problem.  MeetMe strictly assumes 20ms audio in 1.2.0.
Earlier
releases would and could process larger payloads, but the method used
was identified as a source of increasing delay.  The buffering used in
1.2.0 to send and receive audio packets from the zaptel mixing engine
now drops anything past the initial 20ms.

Check out http://bugs.digium.com/view.php?id=5697 for one possible
fix.

Dan
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