30 Jul
2014
30 Jul
'14
12:52 p.m.
Thank you so much for this very useful information. I am working on first
approach for the moment since its much simpler and easier to implement with
only difference being that instead of per header or per sdp line, i plan to
do it in one go, i.e. get entire sip message in $mb (sip message buffer),
encrypt it and put it back in $mb.
- i guess randomizing registration time is already provided by kamailio.
- yes packet sizes are a concern, so i already have planned for random
padding as you mentioned.
For client app, i have a developed a basic prototype based on doubango
framework. I am hopping to release a free and open source implementation
using idoubs within next couple of months on Apple app store.
Thank you.
On Wed, Jul 30, 2014 at 12:22 PM, Daniel-Constantin Mierla <
miconda(a)gmail.com> wrote:
On 30/07/14 06:37, Muhammad Shahzad wrote:
Humm, no reply so far, may be because my email was very long and no body
bothered to read it all. Anyways, here is the shorter more direct version
of it. (including kamailio dev list, since question is rather technical).
Is it possible to implement a custom SIP transport in Kamailio script
file i.e. kamailio.cfg. The purpose is to allow experimentation with custom
encryption algorithms such as this,
https://github.com/mshary/itv
What we need is a couple of functions, one to receive incoming raw /
encrypted data received on SIP socket, which then can be parsed / decrypted
in kamailio.cfg (using e.g. LUA or PERL language modules etc.) and
afterwords feed to kamailio for usual processing (as if it was normal /
plain-text sip data received on sip socket). The second function to do the
opposite, it receives the normal / plain-text sip data that is ready to be
sent out from kamailio's core, encrypts it and then send it out to actual
destination.
In case above is not possible. Can i do it in kamailio's native code?
Any hooks / example code for reference?
If you look at encrypting sip messages, look at topoh module. You can
write a replacement for its hooks. Topoh is practically decoding the
headers and then lets the pure SIP message go through config file
execution. Before sending, it encodes the headers and then let it go to the
network.
This is something that should be rather straightforward to do if you are
familiar with C code.
You mentioned that using TLS can still reveal patters of being sip. You
have to think here of ways to obfuscate even in your case of a new
encryption method. What can be matched here:
- periodical registrations - you can have the client (or even the server)
to use different expires times for each registration
- size of packages, specially if user IDs are the same or similar length
(e.g., say everyone uses a 10 digit id), practically no matter who is
calling who, the size will be pretty much the same because most of the
phones I have seen so far use same set of headers. Here you can add random
custom headers for each packet. I haven't checked the proposed encryption
algorithm (some use random blocks implicitly to pad the data), but
eventually you can add random data before and after the packet that you
strip (and re-add) in topoh-replacement module
The other option of having a totally different protocol than SIP should be
possible as well. But you need to re-implement a lot (like location,
authentication, ...). Look at msrp module for an example. This may need to
touch core code a bit.
Of course, in both cases, the client application has to be developed as
well. Perhaps still easier if going for first option, by reusing some open
source sip client and adding the encapsulation/decapsulation layer when
receiving/sending to network.
Cheers,
Daniel
Many thanks and kind regards for your help.
On Mon, Jul 28, 2014 at 2:38 AM, Muhammad Shahzad <shaheryarkh(a)gmail.com>
wrote:
Hi,
As the mobile voip is getting more and more popular these days, there
has been a strong opposition from GSM operators against mobile voip apps.
They often use tactics like blocking voip ports, or detect and block voip
traffic and in some cases restricting udp traffic altogether to very low
upload and download speeds. See below link for some details,
http://www.linphone.org/eng/blog/linphone-over-3g.html
While not all the problems can be solved right now (especially the
limiting udp traffic, since RTP always uses udp transport) I was wondering
if we can at least handle the sip related problems. The most important of
them is SIP traffic detection. While some forks would suggest using TCP/TLS
to encrypt SIP traffic, it has a few problems, e.g.
1. It requires somewhat high resources on mobile devices, so many
low-end android phones simply can't use it.
2. There is possibility that encryption signature may identify it as SIP
traffic. There exists firewalls (often deployed in middle eastern
countries) which have huge database of encryption signatures and patterns
which although may not decrypt the sip packet but at least identify it as
sip packet and block it.
Also with rough agencies of evil empires spying over millions of users
worldwide makes the current encryption standards pretty much pointless, at
least in terms of user privacy and network security. So there is a strong
need to experiment with new ideas and concepts to regain internet freedom.
Some of such ideas are,
1. Convert sip traffic which is plain text to binary format just before
transmitting it and revert it to plain text upon reception.
2. XOR the sip traffic (pretty much same as binary sip).
3. Use some very lightweight but effective / non-standard encryption
algorithm, e.g.
https://github.com/mshary/itv
All these ideas require that SIP server such as Kamailio is able to
adopt to these, preferably with minimal or no change in native code. The
NoSIP module seems an interesting module in this regard. It provides all
traffic it thinks is not the SIP traffic to configuration script, where we
can do our own parsing and do whatever we want with it. I have two
questions about this,
1. If parsed message is SIP, we can we send it back to kamailio core to
get it processed as if it is a normal SIP message received by kamailio?
2. Can this module or any other module available in kamailio, that can
provide us full sip packet that is about to be transmitted over sip socket,
so we can "encode" it just before it is sent to next hop?
I know this would be like writing a SIP transport in kamailio script
which would be very tough if not impossible to implement in native core.
But it will really help in winning the modern mobile voip challenges.
Thank you.
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda