[SR-Users] sip invite proxy-authorization
David Villasmil
david.villasmil.work at gmail.com
Sun Jan 21 23:45:10 CET 2018
I doubt the relay does any authentication.
Any invite seems to be straight to that "tobridge" route which sets the
destination and relays it... there's no auth there whatsoever... you might
want to do ip auth before sending it to the "tobridge" route?
On Sat, Jan 20, 2018, 15:14 Dave & Hazel <dogbark at indigo.ie> wrote:
> Nope, I’m not calling a registered user. Actually, there are no registered
> users. The function of the Kamailio service in this case is to relay WebRTC
> calls to a conferencing bridge. The problem is that whoever set it up did
> so without considering the security implications. And, in fact, it was
> malevolent SIP traffic hitting the bridge that lead me to looking more
> closely at the Kamailio solution.
>
>
>
> There appears to be some custom configuration around routing. I’m not sure
> if route(RELAY) on its own is meant to challenge for authorisation.
>
>
>
> #!ifdef WITH_CONFERENCEBRIDGE
>
> if(is_method("INVITE") && (!route(FROMBRIDGE))) {
>
> # if new call from out there -
>
> # - non-INVITE request are routed directly by Kamailio
>
> # - traffic from is routed also directy by Kamailio
>
>
>
> route(TOBRIDGE);
>
> exit;
>
> }
>
> #!endif
>
>
>
> #!ifdef WITH_CONFERENCEBRIDGE
>
> # Send to bridge
>
> route[TOBRIDGE] {
>
> $du = "sip:" + $sel(cfg_get.bridge.bindip) + ":"
>
> + $sel(cfg_get.bridge.bindport);
>
> route(RELAY);
>
> exit;
>
> }
>
>
>
> #!endif
>
>
>
> As I short term solution to filter unwanted traffic, I’ve updated the
> configuration to only relay calls using a definitive DDI.
>
>
>
> #!ifdef WITH_CONFERENCEBRIDGE
>
> # Send to bridge
>
> route[TOBRIDGE] {
>
>
>
> * if($rU != "8835100xxxxx")*
>
> * return -1;*
>
>
>
> $du = "sip:" + $sel(cfg_get.bridge.bindip) + ":"
>
> + $sel(cfg_get.bridge.bindport);
>
> route(RELAY);
>
> exit;
>
> }
>
>
>
> #!endif
>
>
>
> Thanks guys for your help.
>
>
>
> *From:* sr-users [mailto:sr-users-bounces at lists.kamailio.org] *On Behalf
> Of *David Villasmil
> *Sent:* Saturday, January 20, 2018 1:06 AM
>
>
> *To:* Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>
> *Subject:* Re: [SR-Users] sip invite proxy-authorization
>
>
>
> Again, are you calling a local registeted user?
>
>
>
> On Fri, Jan 19, 2018, 17:28 Dave & Hazel <dogbark at indigo.ie> wrote:
>
> Cheers Alex for your insight.
>
> My problem still remains in that my invites are not being challenged even
> though AUTH is defined.
>
> On 19 Jan 2018, at 15:57, Alex Balashov <abalashov at evaristesys.com> wrote:
>
> Hi,
>
> > On Fri, Jan 19, 2018 at 03:38:24PM -0000, Dave & Hazel wrote:
> >
> > preforming a SIP INVITE without first registering with the Kamailio
> > service (SJ-Phone speak). I am dialling in remotely via NAT and my
> > call is being relayed successfully.
>
> Although many people are led to believe that there is some intrinsic
> connection between registration and outbound calling by the way phone
> UIs present these concepts, there is in fact no such connection
> whatsoever.
>
> Registration is an inbound concept, not an outbound concept. You can
> make calls without being registered. You can make outbound calls using
> different AAA mechanisms. Being registered in no way implies being able
> to make outbound calls. They're just completely unrelated.
>
> There is a common authentication mechanism used in both scenarios:
> digest challenge authentication. As a practical matter, Kamailio sends a
> 407 proxy challenge for requests it is meant to relay (e.g. INVITEs) and
> a 401 Unauthorized challenge for requests of which it is the logical
> destination (e.g. REGISTER), and both draw on the same set of
> authentication credentials and otherwise work the same way. The AUTH
> route covers both of these cases.
>
> -- Alex
>
> --
> Alex Balashov | Principal | Evariste Systems LLC
>
> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>
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