Grandstream Networks GXV3240 Administration Manual page 54

Ignore Alert-Info header
SIP Transport
SIP URI Scheme When
Using TLS
Use Actual Ephemeral
Port in Contact with
TCP/TLS
Symmetric RTP
RTP IP Filter
Support SIP Instance ID
Validate Incoming SIP
Messages
Check SIP User ID for
Incoming INVITE
Authenticate Incoming
INVITE
Allow Unsolicited
REFER
Incoming alert info header from the SIP server will be ignored and default
configured ringtone will be played.
Determines which network protocol will be used to transport the SIP
message. It can be selected from TCP/UDP/TLS. Default setting is "UDP".
Defines which SIP header, "sip" or "sips", will be used if TLS is selected for
SIP Transport. The default setting is "sip".
Determines the port information in the Via header and Contact header of
SIP message when the phone system use TCP or TLS. If set to No, these
port numbers will use the permanent listening port on the phone.
Otherwise, they will use the ephemeral port for the particular connection.
The default setting is "No".
Configures if the phone system enables the symmetric RTP mechanism. If
it is set to "Yes", the phone system will use the same socket/port for
sending and receiving the RTP messages. The default setting is "No".
Receives the RTP packets from the specified IP address and Port by
communication protocol. If it is set to "IP Only", the phone only receives the
RTP packets from the specified IP address based on the communication
protocol;
If it is set to "IP and Port", the phone will receive the RTP packets from the
specified IP address with the specified port based on the communication
protocol. The default setting is "Disable".
Determines if the phone system will send SIP Instance ID. The SIP
instance ID is used to uniquely identify the device. If set to "Yes", the SIP
Register message Contact header will include +sip.instance tag. The
default setting is "Yes".
Specifies if the phone system will check the incoming SIP messages caller
ID and CSeq headers. If the message does not include the headers, it will
be rejected. The default setting is "No".
Configures the phone system to check the SIP User ID in the Request URI
of the SIP INVITE message from the remote party.
If it doesn't match the phone's SIP User ID, the call will be rejected. If set
to "Yes", this feature will be active. The default setting is "No".
Configures the phone system to authenticate the SIP INVITE message
from the remote party. If set to "Yes", the phone will challenge the incoming
INVITE for authentication with SIP 401 Unauthorized response. The
default setting is "No".
Configures whether to dial the number carried by Refer-to after receiving
SIP REFER request actively. If it is set to "Disabled", the phone will send
error warning and stop dialing. If it is set to "Enabled/Force Auth", the
phone will dial the number after sending authentication, if the
GXV3240 Administration Guide
Version 1.0.3.207
P a g e |
54