ipDTL's native calling will always try to establish a ‘point-to-point’ connection between two computers, allowing the most efficient path for optimum latency. If the firewall and/or NAT configuration prevents this, then your audio streams are automatically routed via one of our cloud based relay servers.

With sip.audio, the Relay mode which is set for each Username, determines if calls will be routed via a relay server...

Smart
This is the default option and the most reliable, as calls will always be routed via one of our Relay Servers, improving the chances of NAT traversal. Call codecs will be transcoded where required. G711, G722 & Opus only.

Pass-Thru
This option has the same relaying benefits as Smart, and can be used with any codec (encoding algorithm), so long as both clients offer a common codec.

None
This mode has the potential to give the lowest latency but both clients must offer a common codec, and have STUN enabled.

SIP calls to ipDTL are always relayed, as are phone calls and conferences.