I'd greatly appreciate any help or thoughts!
try: RTP Packet size on SIP tab
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Any ideas what could be going on and how to fix it. I thought it could
be a timing thing. The documentation on the Sipura phones is
non-existent at the moment, so I have no idea what might be able to be
changed.
I’d greatly appreciate any help or thoughts!
How about disabling silence
Are any of the phones setup to use a codec payload of more
than 20ms? Bugid 5697 on the
bug tracker has a patch to deal with very poor MeetMe
performance when any of the participants
are using audio packetization greater than
20ms.
Beta1 and beta2 did not have this problem, and I am not
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