[asterisk-users] Should you "ever" use nat=no?

Bruce B bruceb444 at gmail.com
Fri Feb 17 12:01:52 CST 2012


On Thu, Feb 16, 2012 at 12:30 PM, Kevin P. Fleming <kpfleming at digium.com>wrote:

> On 02/11/2012 06:59 PM, Bruce B wrote:
>
>> If your server is open to the internet and in SIP general section you
>> have nat=no and in peers you have nat=yes or vice versa then it's
>> possible to enumerate your extension. Because Asterisk responds with
>> different messages if the extension exists or not based on that
>> difference in the nat setting then it's possible to tell if an extension
>> 100 exists or not. Over the past few years, Digium has come to
>> realization to respond to all unauthenticated calls the same way in
>> order to thwart any attack attempts or guesses on the extension but it's
>> still not perfect yet as these improvements are done at a really slow
>> pace. Regardless, they are being made and there truely is a security risk.
>>
>
> "really slow pace"? Please point out any one of these issues that took an
> unnecessarily long time to resolve once it was identified and brought to
> the development team's attention.
>
> Was referring to general state and mindset of logging and standardizing it
for security tools.  I was not referring to any Jira
issues particularly though sometimes it takes a lot to convince the dev
team something is a bug or missing. Security should be taken more
importantly and I don't feel it is. A good example is that of "core set
verbose 0" effecting all the logging and rendering all security tools
useless. There is another thread going on about this right now...


>
>> I always use nat=yes. I don't even know why nat=no exists as there is
>> nothing that can't be done with nat=yes. Plus nat=yes will take care of
>> some of the surprise one-way audio scenarios as well so why use nat=no
>> at all?! I vote to totally get rid of the nat setting all together and
>> hard code it and set it to yes but again there are others who may not
>> agree.
>>
>
> As was already pointed out in the discussions that lead up to the 'nat'
> default being changed, there are SIP endpoints out there that do not work
> properly with 'nat=yes' (or 'nat=force_rport'). They behave improperly when
> Asterisk adds an 'rport' parameter to the top-level Via header in its
> responses. Setting 'nat=no' is the only way to keep this from happening.


I agree. Wish they followed a standard to make everyone's life easier.


>
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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>
>
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