This is what we suggest firewall-wise for voip customers:

Firewall Requirements on Voiceless Platform
	Ports	Source
SIP (IPv4)	UDP 5060	81.187.30.110 - 81.187.30.119
SIP (IPv6)	UDP 5060	2001:8b0:0:30::5060:0/112
RTP (IPv4)	UDP 1024-65535	81.187.30.110 - 81.187.30.119
RTP (IPv6)	UDP 1024-65535	2001:8b0:0:30::5060:0/112

Firewall Requirements on Legacy 'C' Platform
	Ports	Source
SIP	UDP 5060	81.187.30.110 - 81.187.30.119
RTP	UDP 1024-65535	Everywhere

SIP is the call routing information that creates and manages calls. in practice if you allow port 5060 from the outside world you'll see attacks and possibly receive spam phone calls. We do not recommend leaving 5060 open unless you really know what you are doing. Phones rarely use ports as low as 5060 for media.

RTP is the actual media (eg, the audio). On the older call servers it will be as direct as possible the media can be sent from anywhere on the internet. Using the ne call servers it is only from the same call server as the SIP control messages. On most phones you can configure which ports to use for RTP, so you can restrict this range further. For example, on a SNOM phone the default range for RTP is 49152 to 65534.

Other things to Firewall

Don't allow access to your phone or servers web configuration pages from the Internet.
If you run your own server and allow phones to use it from your WAN/Internet, then lock this down as much as possible - perhaps only allow access to your PBX from the Internet via a VPN.

NAT

Avoid using NAT where possible. However, some NAT gateways provide an adequate SIP ALG (e.g. Technicolor TG582), and some devices provide NAT that works with the new call server (e.g. FireBrick 2500/2700 and many simple NAT routers). If NAT works, then well done, but if not we cannot guarantee to be able to make it work.

Further VoIP Security

Please see our VoIP Security page for further information on securing your phones, accounts and VoIP systems.