On 01/11/2012 05:29 AM, Steve Davies wrote:
Hi,
Since the recent update to the NAT configuration options and defaults
in chan_sip.so, I am interested in any SIP/NAT best practices advice.
What I've always done in the past is:
Global: nat=no
SIP handsets that are local: nat=no
SIP handsets that are remote: nat=yes
ITSP SIP trunks: nat=yes
I will then set externip and localnet to reflect the local setup,
UNLESS there is a functional SIP ALG doing the work in the gateway
device. I make this statement because I've found one or two firewalls
where it is best to disable the SIP ALG, and one or two where it is
best to leave it enabled.
The above always worked very well, but I now find my asterisk logs
being spammed with warnings containing lots of "!!" and I'd like to
know the best way to operate to achieve what I've always had while
following the new rules in order to be as secure as possible with
"clean" logs. I should add that we do not accept unsolicited
connections, and 99% of attempts to connect will be stopped at the
firewall.
The simplest answer is to always use 'nat=yes' (or at least
'nat=force_rport' in recent versions of Asterisk that support it), until
you come across a SIP endpoint that fails to work properly with that
setting. If you do come across such an endpoint, try hard to get it to
work with that setting; if you can't, then set 'nat=no' for that
endpoint, and understand that the endpoint's name could be discoverable
using the attack methods previously disclosed. If the endpoint's
configuration is suitably locked down (permit/deny, for example) this
may not be a concern for you. If it's not locked down (for example, if
it has to register to your Asterisk server from random locations), then
the next step would be to seriously consider requesting that the user of
that endpoint consider switching to some other SIP endpoint.
To date, the only endpoints that have been identified that do *not* work
with Asterisk's 'rport' handling forced upon them are Cisco phones.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
--
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