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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
