This is what we suggest firewall-wise for voip customers:
|Firewall Requirements on Voiceless Platform|
|SIP (IPv4)||UDP 5060||184.108.40.206 - 119|
|SIP (IPv6)||UDP 5060||2001:8b0:30::5060:0/112|
|RTP (IPv4)||UDP 1024-65535||220.127.116.11 - 119|
|RTP (IPv6)||UDP 1024-65535||2001:8b0:30::5060:0/112|
|Firewall Requirements on Legacy 'C' Platform|
|SIP||UDP 5060||18.104.22.168 - 119|
SIP is the call routing information that creates and manages calls
RTP is the actual audio. On the older call servers it will be as direct as possible the audio 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.
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.