[asterisk-users] asterisk security....again

Rizwan Hisham rizwanhasham at gmail.com
Mon Feb 28 07:27:43 CST 2011


Any suggestions on encrypting the sip and rtp. I have done some googling on
it. looks like it is not supported by most end point devices or service
providers. But still your thoughts will be appreciated on this subject.

On Mon, Feb 28, 2011 at 6:13 PM, Rizwan Hisham <rizwanhasham at gmail.com>wrote:

> You are right Terry. Sorry i did not describe full scenario before. Yes the
> users are remote with atas on port 5060. Attacks on the remote customers was
> my second guess. My network/system admin has already ruled out the
> implementation of VPN. In summary, we dont want to do anything on remote
> customer side. All kind of security and attck prevention techniques have to
> be implemented on the server.
>
> Its comforting to hear someone say "they are harmless". But still i would
> like to know their next step of attack after "guessing/scanning". Or is it
> the only step?
>
> On Mon, Feb 28, 2011 at 5:32 PM, Terry Brummell <terry at brummell.net>wrote:
>
>> When he says “customers” I am assuming he means remote customers.  It
>> sounds like he is a reseller of telecom facilities to me.  Which means his
>> customers most likely have ATA’s with port 5060 forwarded to the ATA, or
>> they are direct on the I’net.
>>
>> He has already set the ATA to only allow calls from the proxy server, so
>> sounds like he has plugged the hole.
>>
>>
>>
>> They are not ‘sniffing’ your traffic, they are guessing/scanning.  That’s
>> it, that’s all, no great conspiracy going on.  They look for open 5060, then
>> send SIP requests to it hopefully finding a badly implemented SIP solution
>> to which they can dial through.  Once they determine they cannot get
>> through, the script will move on to the next sucker.
>>
>>
>>
>> You have a couple of options, which you could implement at **each** of
>> your customers if you wanted.  Set up a VPN, tunnel the SIP/RTP traffic
>> through it.  Set up IPTables at the customer to only allow SIP from your
>> IP.  Or, do what you have already done and forget about these idiots doing
>> the scan, they are harmless at this point.
>>
>>
>>
>> Vlans and DMZ for the server do no good as the attacks are being directed
>> at the remote client side, not the server.
>>
>>
>>
>>
>>
>> *From:* asterisk-users-bounces at lists.digium.com [mailto:
>> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Ricardo Carvalho
>> *Sent:* Monday, February 28, 2011 6:31 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] asterisk security....again
>>
>>
>>
>> Probably, you are receiving INVITE attacks from some tool like sipvicious.
>> You should rearange your network to cover some inportant security issues.
>>
>>
>>
>> The IP address of you server can be revealed in some unincrypted SIP
>> signaling of some call through the Internet to/from your server's client, or
>> simply by your client SRV record in the DNS, if you added it to his DNS.
>>
>>
>>
>> Probably your network is exposed to the Internet. To address those
>> situations, you can use a distinct VLAN to address SIP phones and you also
>> can use port security at the switching ports where you connect your ATAs and
>> phones. You should also deliver with tagging (802.1Q) that VLAN to those
>> ATAs and phones. This should protect you from inside sniffers.
>>
>> This VLAN should just communicate with the DMZ where you should have your
>> asterisk server and between those two networks you should only open the
>> needed ports - for a common SIP infrastructure you should open UDP 5060 and
>> the specified UDP range shown in rtp.conf file for the media to pass. Phones
>> VLAN should not communicate directlly with the world, just in the outbound
>> direction if you like.
>>
>>
>>
>> Regards,
>>
>> Ricardo Carvalho.
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> On Mon, Feb 28, 2011 at 10:33 AM, Rizwan Hisham <rizwanhasham at gmail.com>
>> wrote:
>>
>> Hi all,
>> The problem I have been experiencing since last month is that some of my
>> customers are getting calls with "Asterisk <Unknown>" caller id. Most of
>> them in the middle of the night. And my asterisk server has no record of
>> these calls. The customers were getting irritated as you can imagine. I
>> guessed the only way to receive incoming calls by by-passing the
>> registration server is thru sip-uri calls directly to customers. I have
>> updated the customers atas to not accept any calls from sources other than
>> the registration server. Thats all fine now. But the question is how can
>> anyone know the direct sip uri addresses of our customers.
>>
>> My guess is that someone has been sniffing my server's sip traffic. In
>> that case what should i do to get rid of the sniffers?
>>
>> If you think there is another reason for that then please tell me even if
>> you dont have the solution.
>>
>> Thanks
>>
>> --
>>
>> Best Ragards
>>
>> Rizwan Qureshi
>>
>> VoIP/Asterisk Engineer
>>
>> Axvoice Inc.
>>
>> V: +92 (0) 3333 6767 26
>>
>> E: rizwanhasham at gmail.com
>>
>> W: www.axvoice.com
>>
>>
>>
>>
>>
>> --
>> _____________________________________________________________________
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>
>
>
> --
> Best Ragards
> Rizwan Qureshi
> VoIP/Asterisk Engineer
> Axvoice Inc.
> V: +92 (0) 3333 6767 26
> E: rizwanhasham at gmail.com
> W: www.axvoice.com
>
>


-- 
Best Ragards
Rizwan Qureshi
VoIP/Asterisk Engineer
Axvoice Inc.
V: +92 (0) 3333 6767 26
E: rizwanhasham at gmail.com
W: www.axvoice.com
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