18 Jun
2007
18 Jun
'07
9:21 a.m.
I do not think that this is a bad configuration. If the WRONG_IP_ADDRESS
is the IP address of a gateway this would be a simple approach to bypass
the billing logic in the SIP proxy to achieve free calls.
The INVITE is perfectly valid. For example, if a sip:alice@atlanta calls
sip:bob@biloxy the Atlanta SIP proxy sees a similar request. (although
usually the To URI would be the same as the RURI).
You could: screen the RURI for bad IP addresses (manually or by using
the permission module).
regards
klaus
Andreas Sikkema wrote:
Klaus,
Maybe someone sent a SIP message with the local
IP address in
the R-URI
to your proxy. According to your config this should not matter as you
rewrite the domain for every incoming SIP request - except
for loose-route.
The IP address in the Request-URI is one that for some reason
(incorrect configuration?) is being sent by the SIP UA. This IP
address is NOT the same machine as OpenSER...
The loose_route block could indeed be the culprit here. Is there a
way I can see in an INVITE being sent to OpenSER that loose_route() will
return true?
This is the (sanitized) INVITE that is causing the problems:
INVITE sip:[NUMBER]@[WRONG_IP_ADDRES] SIP/2.0
Call-ID: 6181509477cc0df82db6313a276492d5@[SIP_DOMAIN]
From: <sip:[USERNAME]@[IPADDRESS_PROXY]:[PROXYPORT]>;tag=45fa8
To: "[NUMBER]" <sip:[NUMBER]@[SIP_DOMAIN]>;tag=as52bb63c2
CSeq: 101 INVITE
Via: SIP/2.0/UDP
10.1.2.71:5060;branch=z9hG4bK-45a33f186083645f5232ca79048af818
Contact: "[USERNAME]" <sip:[USERNAME]@10.1.2.71:5060>
Max-Forwards: 70
Route: <sip:[IPADDRESS_PROXY]:[PROXYPORT];lr>
User-Agent: [USER_AGENT]
Supported: timer
Content-Type: application/sdp
Content-Length: 151
(I've removed the body of the message; it's not very interesting here)
The problem is that the Request-URI contains [WRONG_IP_ADDRESS]. When
this
OpenSER receives an INVITE it should never forward the INVITE to an IP
address a SIP UA has added to the Request-URI. OpenSER should be the one
to decide this.
So far we've never had any issues in this area, so I don't want to
change
too much in the config (like removing the whole loose_route() stuff)
unless that is the only way I can solve this problem.
If the Request-URI would have been correct, [NUMBER] would have been
rewritten to something usefull and the call would have succeeded.
For loose-route I would suggest to check for a
to-tag too. Then, the
called SIP UA should reject the call. You could also use
dialog module
to see if a dialog exists before forwarding in loose_route.
How would checking for a to_tag help? There is no existing Ddialog; this
is handling the initial INVITE message where it already goes wrong...