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

[Ryan Booz]

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?

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 Wolfgang S.
Rupprecht
Sent: Thursday, December 08, 2005 4:27 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Re: Meetme and Sipura SPA-941 -
badjitter/distortion


"Ryan Booz" <ryanb at energycap.com> writes:
> Now, however, there is a (very) slight echo introduced into any calls made
> to this extension.  So obviously the way that the phone sends packets is
> causing some issues.  Anyone have a resource or guide to point me to on
best
> way to debug packet transmission for good calls?

Are you sure the echo isn't acoustic echo from the handset itself?

Its older sibling, the SPA-841 was really bad in this regard.  On a
purely sip call between two SPA-841's, if you bumped the earphone gain
past halfway on the display the other side would invariably complain
about the echo.  I always wanted to fill the Sipura handset with
modeling clay and see if that helped things any.

(The echo was only a problem on direct sip-to-sip calls.  Any calls
going into the PSTN seemed to always be processed by an echo-can, so
it wasn't noticed there.)

-wolfgang
-- 
Wolfgang S. Rupprecht                http://www.wsrcc.com/wolfgang/
Direct SIP URL Dialing: http://www.wsrcc.com/wolfgang/phonedirectory.html
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