sip.audio:
Depending on your SIP client's configuration, TCP or UDP port 5060, and UDP ports 16834 to 32768. We recommend to use TCP or TLS for SIP signalling.  When we talk about 'opening' ports, we really just mean 'not filtered', as Standard NAT configuration will allow the inbound stream through those ports, once the outbound stream has established the route.

Additionally, you should generally ensure that any 'ALG SIP Helper' is disabled on your network, as these tend to be unhelpful in reality.

The sip.audio website's SIP Usernames configuration page has a setting for Relay mode.  'Smart' is the default setting, and is the safest option for navigating through NAT, but it only works with our preferred codecs (Opus, G722 & G711).

ipDTL & hybrIP:
As above, for ipDTL's SIP functionality. In addition, ipDTL can negotiate with other ipDTL instances directly over TCP port 443, to overcome difficult firewalls. This is only the case when ipDTL's native calling functionality is used (e.g. Send a Link) and not when a SIP address is dialed.

Whitelisting
If your corporate firewall requires whitelisting, then the information below may be helpful, but a simpler approach would be to use wildcard rules for *.ipdtl.com and *.sip.audio

*1) If UDP audio data cannot traverse the firewall, a TURN relay server will be used.