Yes it is 10Mbps..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).
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..
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.
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 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..
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.
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..
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