[Asterisk-Users] Re: Jittery meetme conference using Linksys 942 phones

[LJ]

I have not used the Linksys 942 phones yet, but I have a couple of Sipura 
841's.  Check to see what your RTP payload encoding frame length is, ie. 
20ms or 30ms.  Also check to see if there is a setting to surpress or 
transmit silence.  If so you want to transmit silence.


"Rana Dutt" <astuserlist at gmail.com> wrote in message 
news:dd0d3b9c0603181116y26db80e2r4fd33a970c7a2340 at mail.gmail.com...
We have two Linksys 942 phones which sound great when they call each other 
directly through Asterisk. But when they both dial in to a meetme conference 
room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 
sound fine when using meetme.

Both Linksys phones are set to use the default g711u (ulaw) codecs. 
Adjusting the jitter buffer and jitter level settings to various values did 
not help.

We are running Asterisk 1.2.1 on Centos 4.2 (Linux 2.6x kernel) on a 
dual-processor Dell Poweredge 2850 server with 1 Gb RAM. This machine has a 
TE-210 Dual-T1 card plugged in. The meetme.conf file has no general 
settings, just a list of two conference rooms.

Has anyone else experienced sound quality issues with meetme conferences 
using Linksys phones? Any idea what could fix this? Thanks.

Ron



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