[VoiceOps] Preventing random SIP connections to handsets

Alex Balashov abalashov at evaristesys.com
Fri Nov 20 15:27:49 EST 2015

On 11/20/2015 03:23 PM, Carlos Alvarez wrote:

> That's the default for all the handsets, I believe.  There are various
> options such as "accept only from proxy" or "only from registrar," but
> like I said it varies so it could be more challenging to employ that.
> Also in our limited testing it seems like it may not have had the
> intended effect.  Possibly because NAT hides the original IP, but I
> don't know that for sure.

Any properly standards-compliant registrar will send a Request URI on 
incoming INVITEs that is equivalent to the Contact binding provided by 
the phone originally. It can choose to send that INVITE to a network and 
transport-layer destination that is different to the network and 
transport-reachability in the contact provided by the handset, i.e. for 
far-end NAT traversal, but the integrity of the RURI should not be 

> Most phones also have an option to force auth for incoming invites,
> which we have not tested yet.

I don't think you want that. SIP servers and registrars will certainly 
definitely expect the registrant to trust them. You can certainly 
configure Asterisk per se to answer 401/407 challenges from the phone 
with digest credentials, but that's not a very simple or interchangeable 

-- Alex

Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States

Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

More information about the VoiceOps mailing list