Account/Call Settings - Grandstream Networks WP820 Administration Manual

Enable SRTP Key Life
Time
RTCP Destination
Symmetric RTP
RTP IP Filter
RTP Timeout (s)

Account/Call Settings

Start Video Automatically
Auto Answer
Play Warning Tone for
Auto Answer Intercom
Intercom Barging
● AES128&256 bit
● AES 128 bit
● AES 256 bit
If it is set to "AES 128&256 bit", the phone will provide both AES 128 and 256 cipher suite 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 be valid within 231 SIP packets, and phone will renew the SRTP key after this
limitation. The default settings is "Yes".
Configures the server address. When there is a call, the RTCP package sent from the phone will
also be sent to this address.
Note: The address should contain port number.
Configures if the phone enables the symmetric RTP mechanism.
If it is set to "Yes", the phone 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".
Configures the RTP timeout setting; If the phone does not receive any RTP packet within the
specified RTP time, the call will be automatically disconnected. Default settings is 30s and the
supported range is between 0 and 600.
The corresponding P-value of this settings for Account 1 and Account 2 are:
P29068 and P29168, respectively.
Table 14: Account/Codec Settings
Call Features
Defines whether to enable video automatically when there is a call from contacts or dialing
interface. If set to "Yes", the video call will be started automatically. If set to "No", dial the call
via SIP audio.
If set to "Yes", the phone will automatically turn on the speaker to answer incoming calls after a
short reminder beep.
If set to "Enable Intercom/Paging", it will answer the call based on the SIP Info header sent from
the server/proxy. The default setting is "No".
When enabled, the phone will play warning tone when auto answer Intercom. By default, it is
enabled.
Configures whether to answer the incoming intercom call when there is already an active call
on the phone. When "Intercom Barging" is enabled, if the current active call is an intercom call,
the incoming intercom call will be automatically rejected;
If the current active call is not an intercom call, the current active call will be put on hold and
the incoming intercom call will be automatically answered.
When "Intercom Barging" is disabled, a prompt will show up indicating the incoming intercom
call without interrupting the current active call.