[Asterisk-Users] Meetme and Sipura SPA-941 - bad jitter/distortion

[Ryan Booz]

The RTP packet size defaults to .03 (packet size in seconds).  Changing this
to .02 or .01 fixed the issue with Meetme.  Anything .03 or above introduces
the "doppler" effect in a Meetme conference.  Thanks.  Codec is uLaw and
silence suppression was off already.

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?

Thanks so much for the quick help!  Most Excellent!

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 Senad
Jordanovic
Sent: Thursday, December 08, 2005 11:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Meetme and Sipura SPA-941 - bad
jitter/distortion


> 
> I'd greatly appreciate any help or thoughts!

try: RTP Packet size on "SIP" tab


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