Sjur Usken

Views on new technologies and business opportunities from Sjur Usken

Monthly Archives: February 2010

Open letter to the people behind the SIP scans….


I have written a letter to the people in charge of the SIP scanning and owner of the network. I have now sent it to several IP responsible for those IP addresses.

——-

Dear support personnel

We have repeatedly been scanned for open SIP (VoIP) access from IP
adresses: 113.105.152.101, 113.105.152.102 and 113.105.152.104.
The last time was February 27th 2010, but this scanning has been going
on for a long time, first reported back in December 2009.

http://www.usken.no/2010/02/and-the-scanning-just-keeps-on-coming/

http://pbxinaflash.com/forum/showthread.php?t=6223

http://www.freepbx.org/forum/freepbx/users/call-log-with-asterisk-in-both-source-and-clid-columns

We believe this is fraudulent activity to scan for open SIP gateways to
route traffic towards the telephony network.

Please inform the owners of these servers that this activity is not
tolerated and seen as an attack on our servers.

regards

sjur

And the scanning just keeps on coming


A Chinese based server has been very active latest days, and googling the IP addresses ( 113.105.152.102 and 113.105.152.104 ) tells me they have been scanning a long time.

One guy with an Asterisk got hit December 2009 and others back in November. Others starts debugging and asks what it is in public support forums. There will be even more of this scanning coming next months!

Some has added firewall rules like:

deny 113.105.152.102/255.255.255.255
deny 66.117.50.225/255.255.255.255
deny 204.57.122.6/255.255.255.255

But this will not last long until some new IP addresses show up.

What to do about it?

Secure your IP PBX and don’t let port 5060 be open for everybody.
If you must, have very long and strong passwords on all extensions. (or use port knocking..)
Make sure that callers into your PBX is not allowed onto any Outbond context making you pay their calls…

VoIP hackers getting their sentence….


Old news for us in the VoIP, but a reminder for you who think you can abuse VoIP systems and get away with it..

Dan York has a good overview of it on his blog:

Updating a story we have literally been following for years ever since it broke back in July 2006, the FBI recently issued a news release indicating that Edwin Pena pled guilty in what we have been calling the “Pena/Moore VoIP fraud case”. From the news release:

Edwin Pena, 27, a Venezuelan citizen, pleaded guilty before U.S. District Judge Susan D. Wigenton to one count of conspiracy to commit computer hacking and wire fraud and one count of wire fraud. Judge Wigenton continued Pena’s detention without bond pending his sentencing, which is scheduled for May 14.

SIP scanning causes DDoS on IP 1.1.1.1


RIPE said a long time ago that IPv4 is running out of addresses. Now they are also allocating the 1.x.x.x network for production traffic. But this is a bit problematic, since people have been using IP addresses like 1.1.1.1 and 1.2.3.4 as examples in scripts, tools and manuals. People who don’t know any better, they try contact these. When they routes to these networks where alive, a LOT of traffic started coming in.

What made it interesting for Sandro in EnableSecurity was that most traffic was UDP (60 %) and almost 90 % to IP addresss 1.1.1.1. This is a text from RIPEs article about it:

We found that almost 60% of the UDP packets are sent towards the IP address 1.1.1.1 on port 15206 which makes up the largest amount of packets seen by our RRC. Most of these packets start their data section with 0x80, continue with seemingly random data and are padded to 172 bytes with an (again seemingly random) 2 byte value.

This can actually be RTP traffic (VoIP audio traffic) generated from hosts that are vulnerable to SIP INVITE attacks, as Sandro points out in his comment and on his blog.

This is also alarming! This scanning with default RTP audio to IP 1.1.1.1 and port 15206 seems to be doing REALLY well on the Internet. There are a lot of VoIP unsecure platforms accepting and responding to ANY SIP INVITE they get. The software doing it is NOT SIPVicious, but another. It normally uses port 3058 to send the SIP INVITES from. If anybody knows something about this software, please contact me.

I have had a slide in my VoIP presentations about this scenario. If you do a SIP INVITE sweep, you should NOT have a valid IP address for the audio. Every successful INVITE would then generate at least 20 seconds of 0,1Mbit per second stream (g711 audio) to your IP address. Your SIP INVITE sweep with your IP as receiver for RTP traffic will not take long before it backfires on you and you get a DDoS on yourself (well earned though IMHO).

So what is next?

I would love to have a honeypot or get access to the traffic going to port 1.1.1.1. All hosts that would send RTP traffic to this address, should be contacted and asked to secure their servers!

Status now from RIPE:

Since the traffic patterns seemed to be stable we decided to withdraw the announcement of 1.1.1.0/24 and 1.2.3.0/24 on 2 February 2010.