Grandstream Networks GHP6 Series - SIP transport; SIP Registration; Register Expiration; Enable SIP OPTIONS;NOTIFY Keep Alive

Basic Settings
Local RTP
Port
Defines the local RTP-RTCP port pair used to listen and transmit.
The following rule is applied:
N>=0, the default value of Port_Value is 5004.
– 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.
Note: Only when the video FEC mode is set to 1, is the FEC RTP port used.
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.
Hide User
Info for
Video Call
Configures whether to display user information in a video call. If set to “Yes”, user information will not be
displayed in the upper left corner of video area during a video call.
Enable in-call
DTMF display
Enables/disables the phone system to omit the DTMF digits displaying from the LCD screen.
The default setting is “No”.
Enable LDAP
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”.
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.
Table 15: Phone Settings/General Settings
Phone Settings/Call Settings
Call Settings
Enable Video Call
Feature
Enables the video call feature on the phone. The default setting is “Yes”.