[Asterisk-Users] Audio Quality over LAN very bad

[Nic le Roux]

Thanks for the reply,
 
It was on GSM, Ive changed to ulaw last night,
It did make a differance but I'd say its still not as good in quality as the
recorded messages being played back.
 
What is the suggested or should I say, "Best Practise" when it comes to
audio codecs used on asterisk ?
 


  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
Chamberland-Larose, Guillaume
Sent: 01 February 2005 03:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Audio Quality over LAN very bad


Maybe you're transcoding on the server with cpu intensive codecs? That would
be the first thing I'd look at. Try using G.711 (ulaw) on both SIP phones
and remove reinvite=no and canreinvite=no from your phone declarations in
sip.conf.
 
Hope that helps.
 
Guills


  _____  

From: Nic le Roux [mailto:nicl at i-procc.za.net] 
Sent: Monday, January 31, 2005 7:01 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Audio Quality over LAN very bad


Hi All,
 
I'm running Asterisk on the following
 
vendor_id       : GenuineIntel
model name      : Celeron (Coppermine)
cpu MHz         : 668.202
cache size      : 128 KB

 
with 192 MB Ram
 
Audio coming from Asterisk (the demo ) is excellent when using a SIP phone
on the LAN to Asterisk,
and when dialling in from outside via ISDN to Asterisk.
 
However, when connecting from SIP phone to SIP phone (across LAN) and
dialling from externally to SIP which is on the local LAN
it is very choppy and one can barely make out the other party.
I'm using an Eicon Diva 2-m card and 100mb network all round.
 
What could be the cause as I believe bandwidth is ruled out.
 
 
Thanks and regards
Nic
 
 


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