Grandstream Networks GSC3620 User Manual page 35

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Sets the local SIP port. Default setting is 5060 for Account 1, 5062 for Account 2,
Local SIP Port
5064 for Account 3, 5066 for Account 4.
Chooses the SIP transport protocol. UDP, TCP or TCP/TLS.
SIP Transport
Default setting is UDP.
Select the Video stream to be used by the GSC3620 when call is made from this
SIP Account.
Stream
Default is Stream 2.
Specifies the mechanism to transmit DTMF digits. There are 2 supported modes:
Enable DTMF
Configures the payload type for DTMF using RFC2833.
DTMF Payload
Type
Default value is 101. Range: 96~127.
Allows the SIP user's registration information to be cleared when the GSC
Unregister on
reboots. The SIP REGISTER message will contain "Expires: 0" to unbind the
Reboot
connection.
This parameter configures whether the NAT traversal mechanism is
activated. Users could select the mechanism from No, STUN, Keep-alive,
UPnP, Auto. The default setting is "No".
If set to "STUN" and STUN server is configured, the GSC will route according to
the STUN server. If NAT type is Full Cone, Restricted Cone or
Port-Restricted Cone, the unit will try to use public IP addresses and port
NAT Traversal
number in all the SIP&SDP messages.
The GSC will send empty SDP packet to the SIP server periodically to keep the
NAT port open if it is configured to be "Keep-alive". Configure this to be "No" if an
outbound proxy is used. "STUN" cannot be used if the detected NAT is symmetric
NAT. If the firewall and the SIP device behind the firewall are both able to use
UPNP, it can be set to "UPNP". Both parties will negotiate to use which port to
allow SIP through.
Enable SRTP mode based on your selection from the drop-down menu.
Enable SRTP
The default setting is "Disabled", the two other modes are "Enabled but Not
Forced" and "Enabled and Forced".
Configures GSC settings to meet different vendors' server requirements.
Special Feature
Users can choose from Standard, BroadSoft or Telefonica Spain.
The default setting is "Standard".
RFC2833 sends DTMF with RTP packet. Users can check the RTP
packet to see the DTMFs sent as well as the number pressed.
SIP INFO uses SIP INFO to carry DTMF. Default setting is "RFC2833"
GSC362X Series User Manual
Version 1.0.5.3
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