Yeastar Technology TG200 User Manual page 70

Tg series gsm/3g voip gateway
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Configure NAT Settings
If you want to register the TG VoIP account remotely, you need to configure NAT setting on TG
gateway. Navigate to Gateway > VoIP Settings > SIP Settings > NAT, to configure the SIP NAT
settings.
Check the description of NAT settings below.
Option
Enable STUN
STUN Address
External IP Address
External Host
External Refresh
Interval
NAT Mode
Allow RTP Re-invite
Description
STUN (Simple Traversal of UDP through NATs) is a protocol for assisting
devices behind a NAT firewall or router with their packet routing.
The STUN server allows clients to find out their public address, the type of
NAT they are behind and the internet side port associated by the NAT with
a particular local port. This information is used to set up UDP
communication between the client and the VOIP provider and so establish
a call.
The IP address that will be associated with outbound SIP messages if the
system is in a NAT environment.
Alternatively
you can specify an external host, and the system will perform
,
DNS queries periodically.
This setting is only required when your public IP address is not static. It is
recommended that a static public IP address is used with this system.
Please contact your ISP for more information.
Used to identify the local network using a network number/subnet mask
pair when the system is behind a NAT or firewall.
Some examples of this are as follows:
"192.168.0.0/255.255.0.0": All RFC 1918 addresses are local networks;
"10.0.0.0/255.0.0.0": Also RFC1918;
"172.16.0.0/12": Another RFC1918 with CIDR notation;
"169.254.0.0/255.255.0.0": not get the IP address from DHCP server
Please refer to RFC1918 for more information.
Global NAT configuration for the system; the options for this setting are as
follows:
Yes = Use NAT. Ignore address information in the SIP/SDP headers and
reply to the sender's IP address/port.
No = Use NAT mode only according to RFC3581.
Never = Never attempt NAT mode or RFC3581 support.
Route = Use NAT but do not include rport in headers.
By default, the system will route media steams from SIP endpoints through
itself. Enabling this option causes the system to attempt to negotiate the
endpoints to route packets to each other directly, bypassing the system. It
is not always possible for the system to negotiate endpoint-to-endpoint
media routing.
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