Call Settings - Grandstream Networks GSC3505 Series User Manual

Configures all the AES (Advanced Encryption Standard) key size within SRTP. It can be selected from
dropdown list:
SRTP Key AES128&256 bit, AES 128 bit or AES 256 bit
Length
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 life time. When this option is set to be enabled, during the SRTP call, the SRTP key will
Enable SRTP
be valid within 2
Key Life
Time
The default setting is "Yes".
RTCP
Configures a remote server URI where RTCP messages will be sent to during an active call.
Destination
Configures if the device enables the symmetric RTP mechanism.
Symmetric
If it is set to "Yes", the device will use the same socket/port for sending and receiving the RTP messages.
RTP
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 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
RTP IP Filter
with the specified port based on the communication protocol.
The default setting is "Disable".

Call Settings

Call
Features
Play
Warning
Tone for When this option is enabled, the device will play a warning tone When auto-answering intercom. The default
Auto setting is "yes".
Answer
Intercom
Send
If set to "Yes", the "From" header in outgoing INVITE messages will be set to anonymous, essentially blocking
Anonym
the Caller ID to be displayed.
ous
Intercept
Anonym If set to "Yes", anonymous calls will be rejected.
ous Calls
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SIP packets, and phone will renew the SRTP key after this limitation.