VoIP Firewall

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Revision as of 10:08, 30 July 2013 by AA-Andrew (talk | contribs)

Go to the VoIP Category

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

Firewall Requirements on Voiceless Platform
Ports Source
SIP (IPv4) UDP 5060 - 119
SIP (IPv6) UDP 5060 2001:8b0:30::5060:0/112
RTP (IPv4) UDP 1024-65535 - 119
RTP (IPv6) UDP 1024-65535 2001:8b0:30::5060:0/112

Firewall Requirements on Legacy 'C' Platform
Ports Source
SIP UDP 5060 - 119
RTP UDP 1024-65535 Everywhere

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.