[SR-Users] Eavesdropping SRTP sessions

[Mino Haluz]

Ok so what I understand from the document - there are in fact only these
possibilities, how to be sure there is not Mitm.

1) To use ZRTP for media encryption with SIP TLS (in case proxy is
compromised, attacker can not still decrypt ZRTP even though it goes
through the proxy)
2) To use IPSec for media between the clients (can be SIP or SIPS, does not
matter) if media goes directly between clients
3) To use SRTP with other key management (MIKEY, SDES) ?

When using these ways, audio could be decrypted

1) SRTP with SIP (keys are exchanged in SDP, so if they are not encrypted,
SRTP loses its sense)
2) SRTP with SIPs (if the proxy is hacked, SIPs packet are decrypted on
proxy and SDP payload can be seen, and SRTP packets can be decrypted)

Right?


On Tue, Nov 27, 2012 at 11:21 PM, Jesús Pérez Rubio
<jesus.perez at quobis.com>wrote:

> I forgot something, with Kamailio default configuration media goes always
> directly between clients. Moreover, if you want to be sure that any
> endpoint is who it says to be you should use client side autentication for
> SIP protocol. TLS module documentation clears how to do it.
>
> http://kamailio.org/docs/modules/devel/modules/tls.html
>
>
>
> 2012/11/27 Jesús Pérez Rubio <jesus.perez at quobis.com>
>
>> Hi, If you are using SRTP your conversations will be encrypted, so nobody
>> could eavesdrop it. Only if  your Kamailio was compromised they could be
>> eavesdropped.
>>
>> I think you are confusing SRTP (media) with signaling (SIP). You should
>> implement SIP over TLS too, it makes no sense to use SRTP without encrypt
>> signaling. If not, it could be possible to sniff conversations with a MiTM
>> but, anyway, I don't know any tool which supports it.
>>
>> Here I speak a bit about VoIP encryption, I think it could help you:
>>
>> http://nicerosniunos.blogspot.com.es/2011/08/voip-eavesdropping-counter-measurements.html
>>
>> Best regards.
>>
>>
>>
>> 2012/11/27 Mino Haluz <mino.haluz at gmail.com>
>>
>>>  Hi,
>>>
>>> maybe it is not that kamailio related question, but I dont know any
>>> other place with such good voip professionals ;) I have kamailio and
>>> mediaproxy. Clients are BudgetTone 200 (Grandstream) and CSipSimple. I am
>>> forcing clients to use SRTP but it does not support adding any certificate
>>> on both sides. SRTP call is working fine.
>>>
>>> The question is, in this case, is man-in-the-middle attack possible?
>>> Maybe I should study SRTP more, but basically, if there are no
>>> certificates, there is no method how to be 100% sure that the media goes
>>> directly between clients. Is it true?
>>>
>>> Thanks for response,
>>> Mino
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
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>>>
>>>
>>
>>
>> --
>> Jesús Pérez
>> VoIP Engineer at Quobis
>>
>> Fixed: +34 902 999 465
>> Site: http://www.quobis.com
>>
>>
>
>
> --
> Jesús Pérez
> VoIP Engineer at Quobis
>
> Fixed: +34 902 999 465
> Site: http://www.quobis.com
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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