Our devices and services are designed to be connected to any standard internet connection, to work 'out of the box'.

Basic Network Requirements:

• Unfiltered ethernet connection to the public internet. A private address with NAT is fine.
  
• DHCP with DNS is preferred for simplicity, as manual configuration can be impractical.
  
• Consistent uncontended bandwidth. 200kbit/s (up & down per call) absolute minimum.
• Disable any 'ALG SIP Helper', as these tend to be unhelpful in reality.

Open ports for sip.audio:
Depending on your SIP client's configuration, TCP or UDP port 5060 (or 5061 if using TLS), and UDP ports 16834 to 32768. 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.

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).

We recommend to use TCP or TLS for SIP signalling, in your device configuration, although our network also supports UDP.

Open ports for 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.

Allow Listing
If your corporate firewall requires allowlisting (aka whitelisting), then the information below may be helpful, but a simpler approach if required 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.