Grandstream Networks GXV3380

GXV3380 Administration Guide
Version 1.0.1.19
P a g e | 57
Phone Settings Page Definitions
Phone Settings/General Settings
Basic Settings
Local RTP Port
Defines the local RTP-RTCP port pair used to listen and transmit.
The following rule is applied: N>=0
- Audio RTP port: Port_Value+10*N
- Audio RTCP port: Port_Value+10*N+1
- Video RTP port: Port_Value+10*N+2
- Video RTCP port: Port_Value+10*N+3
- FEC RTP port: Port_Value+10*N+4
- FEC RTCP port: Port_Value+10*N+5
- BFCP Protocol port: Port_Value+10*N+6
- BFCP RTP port: Port_Value+10*N+8
- BFCP RTCP port: Port_Value+10*N+9
The default value is 50040. The valid range is from 50040 to 65000.
Use Random Port
Forces the phone system to use random ports for both SIP and RTP
messages. This is usually necessary when multiple phones are behind the
same full cone NAT. The default setting is “No”.
Note: This parameter must be set to “No” for Direct IP Calling to work.
Enable in-call DTMF
display
Enables/disables the phone system to omit the DTMF digits displaying
from the LCD screen. The default setting is “Yes”.
Enable Enterprise
Contacts Timeout Auto
Search
Configures whether to display the matched content automatically in search
of the LDAP contacts when timeout. If set to “No”, users need to click the
“Search” button to search the matched contacts mentioned above. The
default setting is “Yes”.
Keep-alive Interval (s)
Specifies how the phone system will send a Binding Request packet to the
SIP server in order to keep the “ping hole” on the NAT router to open. The
default setting is 20 seconds. The valid range is from 10 to 160.
STUN Server
Configures the URI of STUN (Simple Traversal of UDP for NAT) server.
The phone system will send STUN Binding Request packet to the STUN
server to learn the public IP address of its network. Only non-symmetric
NAT routers work with STUN. The default setting is “stun.ipvideotalk.com”.
TURN Server Username
Fill in the username to validate TURN server.
TURN Server Password
Fill in the password to validate TURN server.
Use NAT IP
Configures the IP address for the Contact header and Connection
Information in the SIP/SDP message. It should ONLY be used if it’s
required by your ITSP. The default setting is keep the box blank.