[Asterisk-Users] BudgeTone 102 flakey sound

[WipeOut]

cg at cdegroot.com wrote:

>I have experienced lots of apparently dropped packets (in other words,
>lots of short interruptions of what the other party tries to tell me)
>with a GS102 and chan_capi. The GS102 is connected through a lightly-loaded
>switch directly connected to the * server, so bandwidth/latency
>shouldn't pose a problem. Funny thing is that the switch indicates
>10mbit on the GS102 port - is that correct? (still, 10mbit should be
>plenty).
>
Yes it is 10Mbps..

>
>I'm a bit at a loss where to start debugging - I've tried to play with
>the voice frames per TX parameter (but as I'm experiencing drops on
>RX...), the order of codecs (lowest bandwdith first), but it doesn't 
>seem to make a difference.
>
IIRC there was a discussion a while back and the outcome was that 
changing the frames per packet does not make any difference bacause its 
statically set in Asterisk..

The ONLY codec that works between a GS phone and Asterisk (unless you 
buy a G.729 licence) is the G.711 codec, in otherwords ulaw or alaw.. 
Fortunately these require the least amout of processing power, 
Unfortunately these codecs also use the most bandwidth..

>
>A PII/400 should be enough to do codec translation on one call in any
>case, so I feel that whatever I tune, it's just fiddling and doesn't get
>to the real problem. However, I don't have the faintest idea about where
>to start looking for the real problem.
>  
>
A PII/400 *should* be enough for a single call at least.. of course is 
you are using a passive ISDN card this could be putting extra load in 
your system..

>What could cause drops on such a LAN connection?
>
>  
>
Don't know.. maybe your switch has some form of traffic shaping or 
somthing similar..

Later..