• Saved one until DNS TTL (Stay on responding IP until DNS
timeout):
On this mode, the phone will resolve DNS SRV records and tries to send
the request to the server having lowest priority and if it doesn’t respond, it
will move on to the next IP until one of the servers responds, once this
happen the phone will keep contacting this responding IP until DNS timeout
(30 minutes) before starting over.
• Saved one until no response (Stay on responding IP until its
failure):
On this mode, the phone will send SIP requests to the last responding IP,
and it doesn’t failover/switchover to the next one until this responding
server is down.
Specifies which NAT traversal mechanism will be enabled on the phone
system. It can be selected from the dropdown list:
• NAT NO
• STUN
• Keep-alive
• UPnP
• Auto
• VPN
If the outbound proxy is configured and used, it can be set to “NAT NO”.
If set to "STUN" and STUN server is configured, the phone system will
periodically send STUN message to the SUTN server to get the public IP
address of its NAT environment and keep the NAT port open. STUN will
not work if the NAT is symmetric type.
If set to “Keep-alive”, the phone system will send the STUN packets to
maintain the connection that is first established during registration of the
phone. The “Keep-alive” packets will fool the NAT device into keeping the
connection open and this allows the host server to send SIP requests
directly to the registered phone.
If it needs to use OpenVPN to connect host server, it needs to set it to
“VPN”. If the firewall and the SIP device behind the firewall are both able
to use UPNP, it can be set to “UPNP”. The both parties will negotiate to
use which port to allow SIP through. The default setting is "Keep-alive".
Adds the Proxy-Required header in the SIP message. It is used to indicate
proxy-sensitive features that must be supported by the proxy. Do not
configure this parameter unless this feature is supported on the SIP server.
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