Account/Rtp Settings - Grandstream Networks GXV3350 Administration Manual

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/RTP Settings

SRTP Mode
SRTP Key Length
"Disable".
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 10: Account/Codec Settings
Sets if the phone system will enable the SRTP (Secured RTP) mode. It can be selected from
dropdown list:
● Disable
● Enabled but not forced
● Enabled and forced
SRTP uses encryption and authentication to minimize the risk of denial of service. (DoS). If the
server allows to use both RTP and SRTP, it should be configured as "Enabled but not forced". The
default setting is "Disable".
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