asterisk users - Mar 2006 - Jittery meetme conference using Linksys 942 phones

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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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@gmail.com> wrote in message 
news:dd0d3b9c0603181116y26db80e2r4fd33a970c7a2340@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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What are your zttest results?  zttest can be run from /usr/src/zaptel/
directory (run ./zttest from there).  Do you have Digium hardware or
ztdummy?

Pedro
http://www.TRACI.net

On 3/18/06, Rana Dutt <astuserlist@gmail.com>
wrote:
>
> 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
>
> _______________________________________________
> --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
>
>
>
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