This security checklist is a good start to harden your VoIP setup:
Principle of minimal privilege: Try to restrict your setup as much as possible to do exactly what you intended it to do, not more. This principle implicitly applies to all of the following points.
- Use virtual environments, such as Xen, VirtualBox, OpenVZ, …
- Use a chroot environment.
- Run Yate with a dedicated system user and group.
- Set ulimits to prevent resource exhaustion.
- Use application security systems, e.g. AppArmor.
- Don’t run any other server software on the system.
- Don’t let many users access the system.
- Log admin access.
Filesystem: Restrict access to files:
Yate files, scripts and modules should be owned by a different system user than the user that runs Yate. E.g.
chown -R root:yate /usr/local/etc/yate /usr/local/share/yate
Files should be set read-only for the user that runs Yate. E.g.
chmod -R go-w /usr/local/etc/yate /usr/local/share/yate
Files containing passwords or other sensitive information should be set unreadable for others:
cd /usr/local/etc/yate chmod 640 accfile.conf regfile.conf mysqldb.conf
Consider using encrypted filesystems to protect sensitive data, e.g. voicemail messages or remote VoIP account credentials.
- Set up a firewall to restrict access to SIP, rmanager, extmodule, … and don’t forget IPv6.
- Set up flood protection, e.g. fail2ban.
- Use a VPN to restrict access to access all or parts of Yate.
- Configure management services like rmanager and extmodule to listen on localhost only.
- Configure a dedicated VLAN for VoIP traffic.
- Protect switch ports with IEEE 802.1x if possible.
- Set switch ports to be disabled after link is down.
- Write your SQL statements with caution: Only use appropriately escaped or whitelisted values in dynamic queries in order to prevent SQL injection attacks. Keep in mind, that variables may contain user provided values, such as user agent, caller ID or custom SIP headers.
- Restrict Yate database user to DELETE, INSERT, SELECT, USAGE, UPDATE. There is no reason for the database to be dropped or altered by a phone call.
- Think about rejecting suspicious database queries by whitelisting or blacklisting queries before execution using the regexroute module.
- Only allow SIP methods actually needed, e.g. disable OPTIONS.
- Don’t enable subscribe/notify features to unauthenticated users.
Don’t leak information about server software versions to the outside. Change the default SIP header Server: (or User-Agent: for SIP clients) to omit version numbers:
- Filter traffic to other networks, e.g. with a Session Border Controller (SBC).
VoIP routing and dialplan considerations:
- Avoid routing loops. Yate has an internal loop detection. But bouncing calls from one VoIP server to another and back several times will exhaust resources and provide attackers with a deny-of-service attack surface.
- Restrict internal numbers to authenticated clients.
- Categorise clients by source IP, if possible. E.g. internal clients may always have an internal IP.
- Protect your dialout. Anonymous users or SIP scanners should not be able to generate charges on your telephone bill.
- Never trust an incoming caller ID. Caller IDs can be faked, is PSTN as well as in VoIP. Also: Obscure caller IDs should be rejected or rewritten at an early routing stage, e.g. allow only digits 0-9, A-D and maybe allow the international `+’ character in some cases.
- Do not allow users to change their caller ID, e.g. set caller ID based on the authenticated username.
- Explain your dialplan. Draw diagrams. Write tables. Fill Wikis. Anything. Please.
- Test your configuration. In particular, regular expressions as used to create a dialplan with the regexroute module can be tricky.
- Generate strong and random user passwords, e.g. with APG.
- If possible, avoid passwords at all, but use certificates or hardware tokens instead.
- Protect phone applications, e.g. voicemail, with passcodes longer than four digits. If possible, add additional checks for valid caller-IDs, user authentication credentials, IPs, time of day or other criteria.
- Users must be able to change their passwords and PINs on their own. They should also be made aware of this feature.
- Regularly check for new versions.
- Know how to easily update Yate. Take notes on how to compile, deploy, install, upgrade Yate to make life easier for the future you or possibly for other administrators. Also: Store notes where they can be found, e.g. in a file
../YATE_NOTES.txtor a documentation wiki (or even an offline notebook).
Transport Encryption: Consider setting up encryption if possible:
- Enable SIP over TLS (SIPS).
- Enable SRTP.
- As a client, validate certificates in order to prevent man-in-the-middle attacks.
- Consider enforcing encrypted calls - SIPS + SRTP - for some numbers, e.g. confidential conference rooms.
- For performance reasons it may be better to use VPN solutions - e.g. IPSec or OpenVPN - for point-to-point links in some cases.
- Log nothing unless absolutely required. For personal use, this may be unnecessary. For business use it may even be against privacy laws to store connection data.
- Think about logging only statistics - e.g. usage counters - without associated names/numbers.
- A cronjob should be in place to delete old data.
Monitoring: Set up monitoring software in order to know when something went wrong.
Security Checks: Implement as many security features as possible and check them on a regular basis.
Disaster Recovery: Keep your VoIP setup well documented and create automated backups on a regular basis. It should be well known what to do after discovering a security incident - for example:
- Disconnect from the internet.
- Restore VoIP setup to a defined state.
- Find and fix vulnerability, e.g. upgrade software.
- Change all passwords, PINs, SSH keys, … and revoke certificates.
- Inform users.