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VoIP Firewall

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Revision as of 18:59, 13 August 2014 by CrazyTeeka (talk | contribs) (Example Config)
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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 -, (NEW)

SIP (IPv6) UDP 5060 2001:8b0:0:30::5060:0/112

2001:8b0:5060::/48 (NEW)

RTP (IPv4) UDP 1024-65535 -, (NEW)

RTP (IPv6) UDP 1024-65535 2001:8b0:0:30::5060:0/112

2001:8b0:5060::/48 (NEW)

The IPs marked as NEW were added in August 2014 for room for a planned expansion. Customers should add these additional IPs to their firewall rules as we expect to be using them soon.

Firewall Requirements on Legacy 'C' Platform
Ports Source
SIP UDP 5060 -
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 new 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.

Example Config

Snom Phone:

<rule-set name="VoIP" target-interface="LAN" no-match-action="reject">
<rule name="SIP" source-ip=" 2001:8b0:0:30::5060:0/112 2001:8b0:5060::/48" target-ip="" target-port="5060" action="accept"/>
<rule name="RTP" source-ip=" 2001:8b0:0:30::5060:0/112 2001:8b0:5060::/48" target-ip="" target-port="49152-65534" protocol="17" action="accept"/>

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.


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. See: VoIP NAT

Further VoIP Security

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