SIPVicious and similar tools are claimed to be used to audit SIP based VoIP systems.
Whilst this is a legitimate usage most of the time they are used to brute force SIP servers or phones and then utilize the found information for toll fraud.
Polycom phones from factory can simply receive invites to their IP address and therefore ring once such INVITE is progressed.
Below Example shows such call to the IP of the Phone in Wireshark:
Or via the Phone logs (log.level.change.sip="0" log.render.level="0"):
1024152724|sip |0|03|<<<Packet Received 1024152724|sip |0|03| INVITE sip:1414160845@10.252.149.102 SIP/2.0 1024152724|sip |0|03|Via: SIP/2.0/UDP 10.20.30.40;branch=z9hG4bK13579926;rport 1024152724|sip |0|03|From: "Call 1" <sip:1414160845@10.20.30.40>;tag=as370e68ea 1024152724|sip |0|03|To: <sip:3100@10.252.149.102> 1024152724|sip |0|03|CSeq: 1 INVITE 1024152724|sip |0|03|Call-ID: 1414160845@10.20.30.40:5060 1024152724|sip |0|03|Contact: <sip:1414160845@10.20.30.40> 1024152724|sip |0|03|Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER 1024152724|sip |0|03|User-Agent: Invite and Cancel Perl Script 1024152724|sip |0|03|Accept-Language: en 1024152724|sip |0|03|Supported: 100rel,replaces 1024152724|sip |0|03|Allow-Events: talk,hold,conference 1024152724|sip |0|03|Max- 1024152724|sip |0|03| Forwards: 70 1024152724|sip |0|03|Content-Type: application/sdp
In the above example, a fictitious IP address of 10.20.30.40 was used.
Option 1:
A simple below configuration parameter should stop most anonymous calls and require that the INVITE is sent to the specific user in the registration.
<web voIpProt.SIP.strictUserValidation="1"/>A phone is registered as 555@example.com
Only calls sent to 555@example.com in the request URI will ring.
Calls to unregistered lines like anonymous@example.com, 1000@example.com, or example.com will all be dropped
Option 2:
In order to add additional security the Admin Guides matching the currently used Software Version document the Incoming Signaling Validation.
You can choose from three optional levels of security for validating incoming network signaling:
- Source IP address validation
- Digest authentication
- Source IP address validation and digest authentication
In the below configuration we choose to utilize the Source IP address validation:
<voIpProt.SIP.requestValidation voIpProt.SIP.requestValidation.1.method="source" voIpProt.SIP.requestValidation.1.request="INVITE" />
An incoming call that does not match the reg.1.server.1.address="" Value will receive a 400 Bad Request SIP Message from the phone.
Wireshark:
Or via the Phone logs (log.level.change.sip="0" log.render.level="0"):
1024153153|sip |0|03|>>> Data Send to 10.20.30.40:5060 1024153153|sip |0|03| SIP/2.0 400 Bad Request 1024153153|sip |0|03| Via: SIP/2.0/UDP 10.20.30.40;branch=z9hG4bK13579926;rport 1024153153|sip |0|03| From: "Call 1" <sip:1414161114@10.20.30.40>;tag=as370e68ea 1024153153|sip |0|03| To: <sip:3100@10.252.149.102>;tag=A9A4BF00-FD048511 1024153153|sip |0|03| CSeq: 1 INVITE 1024153153|sip |0|03| Call-ID: 1414161114@10.20.30.40:5060 1024153153|sip |0|03| User-Agent: PolycomSoundPointIP-SPIP_550-UA/4.0.4.2906 1024153153|sip |0|03| Accept-Language: en 1024153153|sip |0|03| Content-Length: 0
An example configuration file is attached which can be imported via the Web Interface as explained => here <= if running UC Software 4.0.0 or later or use a provisioning server as explained => here <=