Call Settings - Grandstream Networks GAC2570 User Manual

SRTP Key Length
Enable SRTP Key Lifetime
RTCP Destination
Symmetric RTP
RTP IP Filter
RTP Timeout (s)

Call Settings

Call Features
Auto-answer
Play Warning Tone for Auto Answer
Intercom
Send Anonymous
Reject Anonymous Calls
"Enabled but not forced". The default
setting is "Disable".
Configures all the AES (Advanced
Encryption Standard) key sizes within
SRTP. It can be selected from the
dropdown list:
AES128&256 bit, AES 128 bit or AES
256 bit
If it is set to "AES 128&256 bit", the
device will provide both AES 128 and 256
cipher suites for SRTP. If set to "AES 128
bit", it only provides 128-bit cipher suite;
if set to "AES 256 bit", it only provides
256-bit cipher suite. The default setting is
"AES128&256 bit".
Defines the SRTP key lifetime. When this
option is set to be enabled, during the
SRTP call, the SRTP key will be valid
within 231 SIP packets, and GAC2570
will renew the SRTP key after this
limitation. The default setting is "Yes".
Configures a remote server URI where
RTCP messages will be sent to during an
active call.
Configures if the device enables the
symmetric RTP mechanism.
If it is set to "Yes", the device 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 the
communication protocol. If it is set to "IP
Only", the device only receives the RTP
packets from the specified IP address
based on the communication protocol; If it
is set to "IP and Port", the device will
receive the RTP packets from the specified
IP address with the specified port based on
the communication protocol.
The default setting is "Disable".
Configures the RTP timeout of the
GAC2570. If the GAC2570 does not
receive the RTP packet within the
specified RTP time, the call will be
automatically disconnected. The default
range is 0-600. If set to 0, the GAC2570
will not hang up the call automatically.
If set to "Yes", the GAC2570 will
automatically answer incoming calls. If set
to "Intercom/Paging Only", it will answer
the call based on the SIP Call-Info or
Alert-Info header sent from the
server/proxy. The Default is No.
When this option is enabled, the device
will play a warning tone When auto-
answering the intercom. The default
setting is "yes".
If enabled, the "From" header in outgoing
INVITE messages will be set to
anonymous, blocking the Caller ID to be
displayed. Default is No.
If set to "Yes", anonymous calls will be
rejected. Default is No.