Hello,
I'm currently using OpenSER as REGISTER server and Asterisk for the call
routing. Do i need the OpenSER nathelper module if i want to offer
(mostly) automatic NAT traversal to my users or does Asterisk have the
same functionality?
It seems that the nathelper module should be able to automatically
traverse any NAT as long as the User-Agents use symmetric RTP. Further
it is possible (in the ser.cfg) to automatically detect if the use of
nathelper is needed for a specific call.
Is this also possible with Asterisk? I found the options 'canreinvite'
and 'nat' in the sip.conf, but I can't find any information about
what
behaviour the 'nat' option does change. Further I don't want to set
'canreinvite' globally to 'no' as I don't want to proxy the
RTP stream
if this isn't needed.
Should I use Asterisk for this task - or is nathelper the better option?
/gst
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