Dear all
I have small lan and i have configure hardphone with my
asterisk with one E1 PSTN line now i have configue to use canreinvite=yes in
sip.conf
If i user conreinvite=no then my RTP goes throgh asterisk means asterisk come in
media path
and if i user conreinvite=yes then RTP path would be sip phone to sip phone ???
My all phone in LAN not behind the NAT so guessest me what option would be best
for my setup
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Satish Patel
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On Sat, 27 Oct 2007, satish patel wrote:> My all phone in LAN not behind the NAT so guessest me what option would > be best for my setupThat depends on what you think the best option for your setup is. :) No, seriously. On the one hand, in a LAN environment, it's probably easier for your phones to pass media peer-to-peer than to bog down your Asterisk box with it. On the other hand, if it's just a handful of phones, it really couldn't possibly make any less of a difference. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671