User's Manual

GXV3240 User Manual
GXV3240 WEB GUI
Grandstream Co., Ltd
Software Version：1.0.0.11 Document Version：0.1
157
Local RTP Port
Defines the local RTP-RTCP port pair used to listen and transmit. It is the base RTP
port for channel 0.
Users should configure 2 ports for one RTPdialogue: one for RTP and the other for
RTCP.
When configured, channel 0 will use this port_value for RTP and the port_value+1 for
its RTCP; channel 1 will use port_value+2 for RTP and port_value+4 for RTCP. The
default value is 5004.
Use Random Port
When set to "Yes", this parameter will force random generation of both the local SIP
and RTP ports. This is usually necessary when multiple phones are behind the same
full cone NAT. The Default setting is "Yes".
Disable in-call
DTMF display
Disable in-call DTMF.
Keep-alive Interval
(s)
Specifies how often the phone sends a blank UDP packet to the SIP server in order
to keep the port on the NAT router to open. The default setting is 20 seconds.
STUN Server
The IP address or Domain name of the STUN server to solve NAT Traversal.
Use NAT IP
The NAT IP address used in SIP/SDP messages. This field is blank at the default
settings.
SIP TLS Certificate
GXV3240 supports SIP over TLS via the built-in private key and SSL certificate. The
custom SSL certificate used for SIP over TLS should be in X.509 format.
SIP TLS Private
Key
Users could custom SSL private key. The custom SSL Private key used for SIP over
TLS should be in X.509 format.
SIP TLS Private
Key Password
Defines the SSL Private key password used for SIP over TLS.
x-PDFDivision