a) to what extent Asterisk can manage everything necessary to allow
machines A and B to communicate if they were SIP phones. Is it
possible to go for a setup with the firewalls/NAT devices as shown
If the asterisk machine isn't NATed you shouldn't have a problem at all.
If you're using SIP clients just make sure nat=yes
is set in each of the client definitions in sip.conf
b) if I go with IAX softphones, does communication between A and B have
to go through S, or can Asterisk "hand-off" the IAX conversation so
that A and B talk directly.
I'm not sure in this case since both clients are going to be NATed. I'm
pretty sure that this wouldn't work with SIP clients.
Since IAX has less problems with NAT traversal it might work fine - try
setting canreinvite=yes in your iax.conf and monitor
rtp traffic at the asterisk CLI
c) the example documentation shows seperate entries in iax.conf for
incoming and outgoing calls. In my case (assuming IAX softphones)
would I just have entries for A and B of type friend?
Can someone give me some advice about how to proceed.
type=friend works for me...
If you decide to use iax check out moziax - a firefox plugin iax client
that's simple to set up.
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