Grandstream Networks GXP1610 Administration Manual page 23

NAT Traversal
Proxy-Require
Account x -> SIP Settings -> Basic Settings
TEL URI
SIP Registration
Unregister Before New
Registration
Register Expiration
Reregister Before
Expiration
Local SIP Port
SIP Registration Failure
Retry Wait Time
SIP T1 Timeout
SIP T2 Timeout
SIP Transport
SIP URI Scheme when
using TLS
Use Actual Ephemeral
Port in Contact with
TCP/TLS
Remove OBP from

Backup IP 2.
If SIP server is configured as domain name, phone will not send DNS query,
but use "Primary IP" or "Backup IP x" to send SIP message if at least one of
them are not empty. Phone will try to use "Primary IP" first. After 3 tries
without any response, it will switch to "Backup IP x", and then it will switch
back to "Primary IP" after 3 re-tries.
If SIP server is already an IP address, phone will use it directly even "User
Configured IP" is selected.
This parameter configures whether the NAT traversal mechanism is activated.
Users could select the mechanism from No, STUN, Keep-Alive, UPnP, Auto
or VPN. If set to "STUN" and STUN server is configured, the phone will route
according to the STUN server. If NAT type is Full Cone, Restricted Cone or
Port-Restricted Cone, the phone will try to use public IP addresses and port
number in all the SIP&SDP messages. The phone will send empty SDP
packet to the SIP server periodically to keep the NAT port open if it is
configured to be "Keep-Alive". Configure this to be "No" if an outbound proxy
is used. "STUN" cannot be used if the detected NAT is symmetric NAT.
A SIP Extension to notify the SIP server that the phone is behind a
NAT/Firewall. Do not configure this parameter unless this feature is supported
on the SIP server.
If the phone has an assigned PSTN telephone number, this field should be
set to "User=Phone". Then a "User=Phone" parameter will be attached to the
Request-Line and "TO" header in the SIP request to indicate the E.164
number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP
request. The default setting is "Disable".
Selects whether or not the phone will send SIP Register messages to the
proxy/server. The default setting is "Yes".
If set to "Yes", the SIP user's registration information will be cleared when the
phone reboots. The SIP Contact header will contain "*" to notify the server to
unbind the connection. The default setting is "No".
Specifies the frequency (in minutes) in which the phone refreshes its
registration with the specified registrar. The default value is 60 minutes. The
maximum value is 64800 minutes (about 45 days).
Specifies the time frequency (in seconds) that the phone sends re-registration
request before the Register Expiration. The default value is 0.
Defines the local SIP port used to listen and transmit. The default value is
5060 for Account 1 and 5062 for Account 2.
Specifies the interval to retry registration if the process is failed. The default
value is 20 seconds.
SIP T1 Timeout. The default setting is 0.5 seconds.
SIP T2 Timeout. The default setting is 4 seconds.
Determines the network protocol used for the SIP transport. Users can
choose from TCP, UDP and TLS.
Specifies if "sip:" or "sips:" will be used when TLS/TCP is selected for SIP
Transport. The default setting is "sips:".
This option is used to control the port information in the Via header and
Contact header. If set to No, these port numbers will use the permanent
listening port on the phone. Otherwise, they will use the ephemeral port for
the particular connection.
Configures to remove outbound proxy from route. This is used for the SIP
GXP1610/GXP1620/GXP1625/GXP1628
Administration Guide
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