Grandstream Networks GXV34 0 Series Manual page 41

SRTP Key Length
Enable SRTP Key
Lifetime
RTCP Destination
Symmetric RTP
RTP IP Filter
RTP Timeout Timer (s)
VQ RTCP-XR Collector
Name
VQ RTCP-XR Collector
Address
VQ RTCP-XR Collector
Port
Account/SIP/Call Settings
Enable Video Call
Start Video
Automatically
Configures all the AES (Advanced Encryption Standard) key size within SRTP. It can be selected from
dropdown list:
● AES128 & 256 bit
● AES 128 bit
● AES 256 bit
If it is set to "AES 128 & 256 bit", the phone system 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 lifetime. 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.
Default is "Yes".
Configures a remote server URI where the RTCP messages will be sent to during an active call.
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".
Disconnects the call automatically when there is no RTP stream for a specific timeout. Default is 30
seconds.
Configures the host name of the RTCP server that accepts voice quality reports contained in SIP
PUBLISH messages.
Configures IP address of the RTCP server that accepts voice quality reports contained in SIP PUBLISH
messages.
Configures the port of the RTCP server that accepts voice quality reports contained in SIP PUBLISH
messages.
Table 22: Account/SIP/Codec Settings
Call Settings
Configures the video call function for this account. If set to "Default", it will be configured according
to global video call function.
Permits the phone system to enable the video feature automatically when it makes an outbound
call. If set to "Yes", the video codec attributes will be included in the SIP INVITE message. Or the
attributes will not be included.
The default setting is "Yes".