[asterisk-users] Choppy sound, SIP calls within LAN

[John A. Sullivan III]

On Fri, 2009-09-25 at 13:01 +0300, andreil1 wrote:
> Hi!
> 
> I have installed Asterisk 1.6.1 on SuSE Linux (from OpenSuSE  
> repository). As a clients I use XLite on Mac, all on the same LAN.  
> Server where asterisk is is barely loaded at 5% CPU, have a lot of RAM  
> and plenty of disk space on LEVEL 5 RAID.
> 
> Calls to another SIP server (also asterisk) hosted by another company  
> are 100% OK, so it is clearly problem with my server setup.
> 
> Background music (before pickup) runs fine, but transmitted voice  
> sound is very choppy, no matter of which codec I use.
> 
> I have searched over net, and implemented one by one every reasonable  
> receipt found, including.
> 
> highpriority = yes
> internal_timing = yes
> 
> transmit_silence = no
> 
> nat = yes
> localnet=192.168.0.0/255.255.0.0
> externip = xx.xx.xx.xx
> 
> dtmfmode=rfc2833
> 
> Downgrading asterisk did not solved problem, too.
> 
> Anyone please help if possible..
> 
> Many thanks in advance for any suggestion(s).
> 
<snip>
My first guess would be a network problem.  Is there something different
in the network path between the users and the hosted Asterisk server
versus the users and the internal Asterisk server? Have you implement
some form of CoS / QoS internally (one should)? If you run a continuous
ping from a user to the internal Asterisk server, is there any packet
loss or congestion (indicated by widely varying response times)? Just a
few thoughts - John
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsullivan at opensourcedevel.com

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