Monday, December 10, 2012

If SIPVicious gives you a ring...

Note: SIPVicious version 0.28 is out, go get it.

I like to keep an eye on the social media and Google alerts for SIPVicious and in the last few months I noticed a rise in mentions of the tools. Specifically, a number of Korean twitter users (who have their service with KT, a VoIP service provider) complaining about receiving a call from a caller-id showing ‘SIPVicious’.

After contacting a Korean friend, this led to an interview by a reporter for an article that was published on a Korean tech news site Boan News. The good news is that, according to the reporter, KT decided to release security update Nov. 7th for the vulnerable ip phones following the article. 

Since most of this blog’s audience is not Korean-speaking, I thought I’d publish the answers I gave to the reporter.

Questions and Answers

  1. Reporter: Did (sic: does) anybody try to commit a crime using your tool? How do they do?

    Me: Normally (i.e. by default) SIPVicious does not call any phones and does all the security tests silently. Since the tools have been abused by cyber-criminals, the PBX software vendors have updated their software to disallow such scans. What seems to have happened in this case is that someone discovered that if you configure SIPVicious to scan by sending a "call" command (an INVITE scan), then you can still attack some PBX software. Therefore this bypasses the protection implemented by the PBX software vendors. Someone (not me!) even made videos showing this off on Facebook.

    The reason why they would target PBX systems is to find phone lines on that PBX system which have a weak password. Then they would make "free" phonecalls through it to international numbers or premium numbers at the expense of the victim. Therefore that would make them money.

    In this case, however, things are a bit different. While launching an INVITE scan on a vulnerable PBX system can be useful for the hacker, doing the same thing on an IP Phone (or VoIP Phone) just makes it ring. While some phones will only ring when the correct number is called, others ring when any number (or rather, any SIP address) is specified in the INVITE message. So the attackers/hackers/cyber-criminals ended up getting phones to ring. I think this is a mistake that they were making, possibly because they are not differentiating between a phone and a PBX system.
  2. Reporter: Is it a kind of call fraud? I mean, should the person who receive the call pay the money? Is that what they want?

    Me: No I think the person receiving the call does not pay any money in this case as long as it is their IP Phone which is ringing.
  3. Reporter: How should we protect ourselves from it?

    Me: The service provider (i.e. the ISP or VoIP company) could probably prevent the attackers from reaching the phones. The phone vendor could also make the phone ring only when the correct number is called.

One Canadian person posted a picture of her phone showing this:

I wanted to expand a bit on possible mitigation for this. Security precautions that could be taken by different persons or organizations:

  • End users: put their VoIP devices behind a firewall (not necessarily NAT) and only allow it to be contacted by the provider.
  • Vendors (who control firmware): could configure the phones to only ring when the right number is called. Sometimes this has to be done through a firmware upgrade while sometimes this is a configurable option.
  • ITSP/VoIP provider: could configure the network to only allow SIP destined to the VoIP devices from their own gateways. Unfortunately this might prevent end users from making use of third-party SIP providers if not properly configured, so it may cause undesired side-effects in some cases. The VoIP devices might also be configurable to either disallow the phones from ringing on any number, or to only allow SIP messages from the provider itself (through firewall or matching of the SIP registrar IP address).
I also had other friends who reported receiving calls from SIPVicious when they had configured their Asterisk box to route all (unauthenticated) calls to their mobiles. This is probably the same sort of attack. This could lead to extra phone bill charges and the Asterisk installation should probably be configured to only allow authenticated / trusted calls to be routed through.

A reminder of the SIP digest leakage 

This attack is also known as "SIP digest authentication relay attack". Keep in mind that this same behavior (peer to peer calling) can often be abused to leak out the digest authentication response. With VOIPPACK I provide a tool that reproduces this security issue in a number of VoIP phones and ATA devices.

And a SIPVicious update

SIPVicious has supported the OPTIONS method for enumeration for while, which is often similar to sending an INVITE but without getting any phones to ring. This method can often be used to bypass Asterisk's protection against enumeration. I have now updated SIPVicious to officially support INVITE enumeration more gracefully (i.e. sending ACK to responses and BYE). And of course, I fixed yet a few more bugs.

Wednesday, February 22, 2012

SIPVicious 0.2.7 released and rewrite coming up, looking for testers!

Get it now! This is the last release in the 0.2 series which fixes a number of stability issues and bugs before moving on to a total rewrite.

Are you a SIPVicious user? Get in contact if you have a VoIP lab or simply want to test the rewrite of SIPVicious. The internal version already includes support for TCP, TLS and IPv6 ;-)

The changelog for this one:
  • Feature: has a new option -b which bruteforces the attacker's port 
  • Feature: now tries the extension as password by default, automatically 
  • Feature: and now support setting of source port 
  • Feature: new parameter --domain can be passed to all tools which specifies a custom domain in the SIP uri instead of the destination IP 
  • Feature: new --debug switch which shows the messages recieved 
  • Bug fix: Sometimes nonces could not be extracted due to an incorrect regex 
  • Bug fix: Fixed an unhandled exception when decoding tags 
  • Bug fix: now using hashlib when available instead of md5 
  • Bug fix: removed the space after the SIP address in the From header which led to newer version of Asterisk to ignore the SIP messages 
  • Bug fix: dictionaries with new lines made stop without this fix 
  • Change: renamed everything to start with sv 
  • Bug fix: changed the way shelved files are opened by the fingerprinting module 
  • Change: fingerprinting disabled by default since it was giving too many problems and very little benefits
Download SIPVicious from

Monday, January 2, 2012

Asterisk forensics: the logs vs the attackers

Recently I had the opportunity to present on VoIP insecurity around various conferences this year, on my own and also with Joffrey Czarny. 

At Secure 2011 we had one day a workshop and one of the things we showed was the effect of a typical SIPVicious attack on an Asterisk box. The following videos (best seen in full screen and high quality) illustrate what happens.

1. When we run, nothing usually shows up on the asterisk logs. 

2. Running floods the logs with attempts for registrations for various extensions.

3. Running on a valid extension shows a large number of "Wrong password" errors. 

Enumeration and password cracking are not the only attacks being performed on target PBX systems on the Internet. Honeypots and victims are able to pick up a number of INVITE scans looking  for "open sip relays". This is a vulnerability that may affect SIP gateways without proper access control or badly configured dialplans that allow calls to pass through without authentication. 

The following video shows how this looks when done using X-lite (which is what some of the attackers are using) on an Asterisk box. You can see the log entries filling up. 

Hope someone finds this useful when looking at log files or studying attacks on SIP. Feedback is welcome as always

Tuesday, January 25, 2011

VOIPPACK updated to v1.4

Quick note, VOIPPACK now includes support for Cisco Call Manager and more tools to break that Asterisk PBX (FreePBX / Trixbox focus). The blog post on EnableSecurity includes more details.

Tuesday, December 14, 2010

11 million Euro loss in VoIP fraud .. and my VoIP logs

And the attackers made over 1 million in profits.
This just emerged from a raid (and hearing apparently) in Romania and other countries. The two main persons being fingered are Catalin Zlate and Cristian Ciuvat. It seems that they were scanning for PBX servers with phone extensions that have weak passwords. Then they abused these accounts to make phone calls for "free", except that free has the price of 11 million EUR for the victims!

Apparently, originally they used these accounts for their own personal phone calls. However they got greedy and between October 2009 to February 2010, they made 23500 calls / 315000 minutes to premium numbers. Then (from what I understood), they got even more greedy and used Shadow Communication Company Ltd. This site is still available right now - whois shows the name of Cristian Ciuvat (thanks for someone on pointing this out). This site contains lists of prices for premium numbers, linking to another site Using this they recruited other people to make  1,541,187 fraudulent calls or 11,094,167 minutes of talk time.

So right now there are 42 people in court with raids happening in London, Neamt, Brasov, Cluj and Maramures. This is all according to various sites in Romanian, translated automatically using Google translate.

One of the original articles on this can be found here.

Here's a screenshots from their site (in case it goes down):

Some thoughts

On our honeypots we have seen Zoiper messages from Romania and this could possibly be very related. I decided to check out my logs and sure enough found the sort of behavior described in the articles describing the illegal activities. The following IP addresses had Zoiper in their user-agent header when connecting to my simple VoIP honeypot:

  • - Romania
  • - US (San Jose)
  • - US - Brooklyn
  • - US (San Jose)
  • - Romania
  • - Romania
  • - Romania
  • - US (Austin)
  • - Romania 
  • - Romania
  • - Romania
  • - Spain
Here's an example SIP message that was sent from the 1st IP:

INVITE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP;branch=z9hG4bK-d8754z-07cf25937cf90e2a-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:5060;transport=udp>
To: <sip:[email protected];transport=udp>
From: "Unknown"<sip:[email protected];transport=udp>;tag=ce2e1a65
Content-Type: application/sdp
User-Agent: Zoiper rev.6751
Content-Length: 329

o=Zoiper_user 0 0 IN IP4
c=IN IP4
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 98 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

The phone number that the Zoiper user tried calling was "0040767091012". Romanian numbers start with +40, so one can assume that this is some phone that the attacker was using to see if the call is terminated or not. This number is not a premium number. I have looked at other logs and some were probably premium numbers.

Note regarding Catalin Slate: is this the same person who was in 2005 caught with credit card fraud (reference here)?

I would be interested in hearing more about this case as it sheds some light on what's actually happening in the background.

Thanks goes to Stefan Tanase for pointing out the original article in Romanian. The also added the following:
As far as I understand, they were using other premium numbers affiliate networks at first and then, when they realized the potential, they set up a company in the UK - Shadow Communications Inc. - through which they were able to sign a  contract on their own with a premium rate number provider and offer their own affiliates service, basically taking their "business" to a whole new level.


All in all, it looks like these attackers were not so technically advanced after all, yet managed to hit the million euro mark. I wonder what more skilled and stealthy criminals are able to do!

Thursday, November 4, 2010

Distributed SIP scanning during Halloween weekend

Over last weekend there were a number of reports of VoIP (especially Asterisk) servers that were "under heavy attack". I have looked at some packet traces and noticed how the SIP messages look very similar to the ones generated by SIPVicious especially In fact, I think this is a modified version of SIPVicious that is being distributed on a botnet.

Take a look at the following message generated by these new scans:
REGISTER sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=88y8n2p4U3;rport
Content-Length: 0
From: "311"
Accept: application/sdp
User-Agent: Asterisk PBX
To: "311"
Contact: sip:[email protected]
Call-ID: 1728224566
Max-Forwards: 70

The following message would be generated by for the same extension number (username):

REGISTER sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK-4059165492;rport
Content-Length: 0
From: "311"; tag=3331310133333331373232343735
Accept: application/sdp
User-Agent: friendly-scanner
To: "311"
Contact: sip:[email protected]
Call-ID: 3259315903
Max-Forwards: 70

Notice how the order of the headers is exactly the same and so on. However there are some obvious differences:
  • the user agent is now "Asterisk PBX" instead of "friendly-scanner"
  • the branch is totally random instead of starting with "z9hG4bK" as is standard in SIP/2.0
Additionally the To tag is apparently generated differently. In SIPVicious the code to generate the "To tag" is the following:

def createTag(data):
    from binascii import b2a_hex
    from random import getrandbits
    rnd = getrandbits(32)
    return b2a_hex(str(data)+'\x01'+str(rnd))

The data would typically contain the extension which is then concatenated to a random number and encoded. This would later be decoded when the target responds, which helps make the scanner stateless.

In the case of this new scanner, we cannot decode the extension number in the same way.
So clearly, these guys want to bypass intrusion detection/prevention systems such as fail2ban, snort-inline etc by doing the following:
  • distributing the scans across different IP addresses
  • changing the obvious SIPVicious signatures to something more common, such as "Asterisk PBX" since no one in their right mind would block that
Some more interesting things:
  • the source ports for these scans appear to be high ports, indicating that the same host in the botnet is scanning many other hosts at the same time (if the port allocation code is the same as the original)
  • each host scans for consecutive numbers for the extension

If I am correct, then the goals are obvious - enumeration of extensions and then password cracking to be able to make fraudulent phone calls.

Unfortunately, this scanning has been causing bandwidth problems for VoIP providers out there who need to expose their SIP servers on the 'net.

So what do we do about this?
It appears that botnets are now being used for distributed scanning. Therefore it might make sense to look at what others have done before when it comes to botnet borne attacks:
  • Improved their IPS systems to protect against attack (I have some ideas on how to do this)
  • Hid behind VPNs or just shifted to another protocol that is not yet being attacked as much (think SIP over TLS or IAX2 etc)
  • In the case of DDoS attacks, financial companies and online-gambling companies just use expensive 3rd parties that should absorb all the network traffic
  • Usage of blacklists for IP addresses that are known to be part of a botnet (check out the VoIP Blacklist Project)
Unfortunately none of these are real solutions and are rather simply mitigation techniques. In many cases, VoIP providers cannot make use of VPNs or other similar solutions.

The mailing lists had some interesting conversations about this:
And Stuart Sheldon and others blogged about this too.

Thanks to anyone (you know who you are :)) who sent me info already. If anyone has any more insight into this, mitigation, solutions, code or packet traces, please email me.

Friday, October 29, 2010

AstriCon roundup and vendors adding security features

So I've finally been to AstriCon and I noticed a great increased interest amongst the attendees with regards to security, fraud and "hacking". The slides for my presentation titled "Just how vulnerable is your phone system" can be downloaded from this location.

So what are the changes and additions from the software developer's side?
  1. Asterisk 1.8 has been released touting TLS support for SIP and SRTP support too, plus a framework to make auditing easier
  2. 3CX have released a major security update with features to make it easier to set proper passwords
  3. I just received an email from Brekeke highlighting their security page on their wiki which was originally published on March 11, 2009
What accounts for these changes? From talking with the people at AstriCon I started understanding why the increased interest in security: organizations are really getting hurt with call fraud and this seems to be on the increase.

Plus the advise I heard again and again from developers for FreePBX-based systems was:
"Do not put your FreePBX / configuration available on the Internet, it is not designed for that!"

But if you do a simple scan for Asterisk boxes (using for example), you'll notice many systems out there that do not heed this advice. Apart from that, as Blake Cornell showed in his presentation, there are many attacks on FreePBX-based systems that can be abused without direct access to the HTTP configuration interface.