Overview
So you'd like to make some secure calls.Here's how to do it, using Blink, a SIP soft client for Mac OS X, Windows, and Linux.
These instructions assume that you're running as the root user (sudo su -).
Part 1 (TLS)
Transport Layer Security (TLS) provides encryption for call signaling. It's a practical way to prevent people who aren't Asterisk from knowing who you're calling. Setting up TLS between Asterisk and a SIP client involves creating key files, modifying Asterisk's SIP configuration to enable TLS, creating a SIP peer that's capable of TLS, and modifying the SIP client to connect to Asterisk over TLS.Keys
First, let's make a place for our keys.mkdir /etc/asterisk/keys
./ast_tls_cert -C pbx.mycompany.com -O "My Super Company" -d /etc/asterisk/keys
- The "-C" option is used to define our host - DNS name or our IP address.
- The "-O" option defines our organizational name.
- The "-d" option is the output directory of the keys.
- You'll be asked to enter a pass phrase for /etc/asterisk/keys/ca.key, put in something that you'll remember for later.
- This will create the /etc/asterisk/keys/ca.crt file.
- You'll be asked to enter the pass phrase again, and then the /etc/asterisk/keys/asterisk.key file will be created.
- The /etc/asterisk/keys/asterisk.crt file will be automatically generated.
- You'll be asked to enter the pass phrase a third time, and the /etc/asterisk/keys/asterisk.pem will be created, a combination of the asterisk.key and asterisk.crt files.
./ast_tls_cert -m client -c /etc/asterisk/keys/ca.crt -k /etc/asterisk/keys/ca.key -C phone1.mycompany.com -O "My Super Company" -d /etc/asterisk/keys -o malcolm
- The "-m client" option tells the script that we want a client certificate, not a server certificate.
- The "-c /etc/asterisk/keys/ca.crt" option specifies which Certificate Authority (ourselves) that we're using.
- The "-k /etc/asterisk/keys/ca.key" provides the key for the above-defined Certificate Authority.
- The "-C" option, since we're defining a client this time, is used to define the hostname or IP address of our SIP phone
- The "-O" option defines our organizational name.
- The "-d" option is the output directory of the keys."
- The "-o" option is the name of the key we're outputting.
- You'll be asked to enter the pass phrase from before to unlock /etc/asterisk/keys/ca.key.
asterisk.crt asterisk.csr asterisk.key asterisk.pem malcolm.crt malcolm.csr malcolm.key malcolm.pem ca.cfg ca.crt ca.key tmp.cfg
The Asterisk SIP configuration
Now, let's configure Asterisk to use TLS.In the sip.conf configuration file, set the following:
tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/asterisk.pem tlscafile=/etc/asterisk/keys/ca.crt tlscipher=ALL tlsclientmethod=tlsv1 ;none of the others seem to work with Blink as the client
We're binding it to our local IPv4 wildcard (the port defaults to 5061 for TLS).
We've set the TLS certificate file to the one we created above.
We've set the Certificate Authority to the one we created above.
TLS Ciphers have been set to ALL, since it's the most permissive.
And we've set the TLS client method to TLSv1, since that's the preferred one for RFCs and for most clients.
Configuring a TLS-enabled SIP peer within Asterisk
Next, you'll need to configure a SIP peer within Asterisk to use TLS as a transport type. Here's an example:[malcolm] type=peer secret=malcolm ;note that this is NOT a secure password host=dynamic context=local dtmfmode=rfc2833 disallow=all allow=g722 transport=tls context=local
Configuring a TLS-enabled SIP client to talk to Asterisk
Next, we'll configure Blink.First, let's add a new account.
Then, we need to modify the Account Preferences, and under the SIP Settings, we need to set the outbound proxy to connect to the TLS port and transport type on our Asterisk server. In this case, there's an Asterisk server running on port 5061 on host 10.24.13.233.
Now, we need to point the TLS account settings to the client certificate (malcolm.pem) that we copied to our computer.
Then, we'll point the TLS server settings to the ca.crt file that we copied to our computer.
Press "close," and you should see Blink having successfully registered to Asterisk.
Depending on your Asterisk CLI logging levels, you should see something like:
-- Registered SIP 'malcolm' at 10.24.250.178:5061 > Saved useragent "Blink 0.22.2 (MacOSX)" for peer malcolm
Now, make a call. You should see a small secure lockbox in your Blink calling window to indicate that the call was made using secure (TLS) signaling:
Part 2 (SRTP)
Now that we've got TLS enabled, our signaling is secure - so no one knows what extensions on the PBX we're dialing. But, our media is still not secure - so someone can snoop our RTP conversations from the wire. Let's fix that.SRTP support is provided by libsrtp. libsrtp has to be installed on the machine before Asterisk is compiled, otherwise you're going to see something like:
[Jan 24 09:29:16] ERROR[10167]: chan_sip.c:27987 setup_srtp: No SRTP module loaded, can't setup SRTP session.
With that complete, let's first go back into our peer definition in sip.conf. We're going to add a new encryption line, like:
[malcolm] type=peer secret=malcolm ;note that this is NOT a secure password host=dynamic context=local dtmfmode=rfc2833 disallow=all allow=g722 transport=tls encryption=yes context=local
Reload Asterisk's SIP configuration (sip reload), make a call, and voilà:
We're making secure calls with TLS (signaling) and SRTP (media).
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
No hay comentarios:
Publicar un comentario