User's Manual
Grandstream Networks, Inc. HT701 User Manual Page 25 of 34
Firmware Version 1.0.4.2 Last Updated: 06/2011
Outbound Proxy IP address or Domain name of Outbound Proxy, or Media Gateway, or Session Border
Controller. Used by HT701 for firewall or NAT penetration in different network
environments. If symmetric NAT is detected, STUN will not work and ONLY outbound
proxy can correct the problem.
SIP transport
User can select UDP or TCP or TLS.
NAT Traversal (STUN)
This parameter defines whether or not the HT701 NAT traversal mechanism is
activated. If activated (by choosing “Yes”) and a STUN server is also specified, then the
HT701 performs according to the STUN client specification. Using this mode, the
embedded STUN client will detect if and what type of firewall/NAT. If the detected NAT
is a Full Cone, Restricted Cone, or a Port-Restricted Cone, the HT701 will use its
mapped public IP address and port in all of its SIP and SDP messages.
If the NAT Traversal field is set to “Yes” with no specified STUN server, the HT701 will
periodically (every 20 seconds or so) send a blank UDP packet (with no payload data) to
the SIP server to keep the “hole” on the NAT open.
SIP User ID User account information, provided by VoIP service provider (ITSP). Usually in the form
of digit similar to phone number or actually a phone number.
Authenticate ID SIP service subscriber’s Authenticate ID used for authentication. Can be identical to or
different from SIP User ID.
Authenticate Password
SIP service subscriber’s account password.
Name
SIP service subscriber’s name for Caller ID display.
DNS Mode
One from the 3 modes are available for “DNS Mode” configuration:
-A Record (for resolving IP Address of target according to domain name)
-SRV (DNS SRV resource records indicates how to find services for various protocols)
-NAPTR/SRV (Naming Authority Pointer according to RFC 2915)
One mode can be chosen for the client to look up server.
The default value is “A Record”
User ID is Phone
Number
If the HT701 has an assigned PSTN telephone number, this field should be set to “Yes”.
Otherwise, set to “No”. If set to “Yes”, a “user=phone” parameter will be appended to
the “From” header in SIP request.
SIP Registration Controls whether the HT701 needs to send REGISTER messages to the proxy server.
The default setting is Yes.
Unregister on Reboot Default is No. If set to Yes, the SIP user’s registration information will be cleared on
reboot.
Outgoing Call without
Registration
Default is No. If set to “Yes,” user can place outgoing calls even when not registered (if
allowed by Internet Telephone Service Provider) but is unable to receive incoming calls.
Register Expiration This parameter allows the user to specify the time frequency (in minutes) the HT701
refreshes its registration with the specified registrar. The default interval is 60 minutes
(or 1 hour). The maximum interval is 65535 minutes (about 45 days).
Registration Retry Wait
Time
Retry registration if the process failed. Default is 30 seconds.
Local SIP port Defines the local SIP port the HT701 will listen and transmit. The default value for FXS
port 1 is 5060. The default value for FXS port 2 is 5062.
Local RTP port Defines the local RTP-RTCP port pair the HT701 will listen and transmit. It is the base
RTP port for channel 0. When configured,
channel 0 uses this port _value for RTP and the port_value+1 for its RTCP; channel 1
uses port_value+2 for RTP and port_value+3 for its RTCP.
The default value for FXS port 1 is 5004. The default value for FXS port 2 is 5012.
Use Random Port This parameter forces the random generation of both the local SIP and RTP ports when
set to Yes. This is usually necessary when multiple HT701 are behind the same NAT.
Refer to Use Target
Default is NO. If set to YES, then for Attended Transfer, the “Refer-To” header uses the