Grandstream Networks Grandstream GXP-2010 User Manual page 32

Outbound Proxy
SIP User ID
Authenticate ID
Authenticate Password
Name
Use DNS SRV:
User ID is Phone
Number
SIP Registration
Un-register on Reboot
Register Expiration
Local SIP Port
SIP Registration Failure
Retry Wait Time
SIP T1 Timeout
SIP T2 Interval
SIP Transport
Use RFC3581
Symmetric Routing
NAT Traversal (STUN)
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IP address or Domain name of Outbound Proxy, Media Gateway, or Session
Border Controller. Used for firewall or NAT penetration in different network
environment. If the system detects symmetric NAT, STUN will not work. ONLY
outbound proxy can provide solution for symmetric NAT.
User account information provided by VoIP service provider (ITSP); either an
actual phone number or formatted like one.
SIP service subscriber's Authenticate ID used for authentication. It can be
identical to or different from SIP User ID.
SIP service subscriber's account password for GXP to register to (SIP) servers of
ITSP.
SIP service subscriber's name that is used for Caller ID display.
Default is No. If set to "Yes", the client will use DNS SRV to look up server.
If the phone has an assigned PSTN telephone number, this field should be set to
"Yes". Otherwise, set it to "No". If "Yes" is set, a "user=phone" parameter will be
attached to the "From" header in SIP request
This parameter controls sending REGISTER messages to the proxy server. The
default setting is "Yes".
Default is No. If set to "Yes", the SIP user's registration information will be
cleared on reboot.
This parameter allows user to specify the time frequency (in minutes) that GXP
refreshes its registration with the specified registrar. The default interval is 60
minutes. The maximum interval is 65,535 minutes (about 45 days).
This parameter defines the local SIP port used to listen and transmit. The default
value for Account 1 is 5060. It is 5062, 5064, 5066 for Account 2, Account 3 and
Account 4 respectively.
Retry registration if the process failed. Default is 20 seconds.
RFC 3261 SIP T1 timer. Default is 1 second.
RFC 3261 SIP T2 timer. Default is 0.5 seconds.
Choose SIP Transport between UDP and TCP. Default is UDP.
This option allows a SIP client to request that the server send the response back
to the source IP address and port from which the request originated.
This parameter activates the NAT traversal mechanism. If activated (by choosing
"Yes") and a STUN server is also specified, the phone performs according to the
STUN client specification. Using this mode, the embedded STUN client detects if
and what type of NAT/Firewall configuration is used. If the detected NAT is a Full
Cone, Restricted Cone, or a Port-Restricted Cone, the phone 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 GXP 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.
GXP User Manual
Firmware 1.1.5.15
Page 32 of 38
Last Updated: 12/2007