Grandstream Networks WP820 Administration Manual page 24

Use H.264 Constrained
Profiles
Video Bit Rate
SDP Bandwidth Attribute
SRTP Mode
SRTP Key Length
Enable SRTP Key Life
Time
Note: For some network environment, the default setting "720P" might be too high that causes not
receiving video or video quality issue during video call to WP820. In this case, please change "H.264 Image
Size" to "VGA" or "CIF" and change "Video Bit Rate" to "384kbps" or lower. The default setting is 720P.
Configures that whether to set H.264 constrained profiles to use H.264 CBP to establish video call with
WebRTC. The function takes effect when H.264 profile setting includes BP type. It is recommended to set
to "Yes" when establishing video call with WebRTC
The default setting is "No".
Configures the bit rate for video stream received on the WP820. The default setting is 2048 kbps. The valid
range is from 32 – 2048 kbps.
Note: The video bit rate can be adjusted based on the network environment. Increasing the video bit rate
may improve video quality if the bandwidth is permitted. If the bandwidth is not permitted, the video
quality will decrease due to packet loss. For some network environment, the default setting "720P" might
be too high that causes no video or video quality issue during video call to WP820. In this case, please
change "H.264 Image Size" to "VGA" or "CIF" and change "Video Bit Rate" to "384kbps" or lower.
Sets the SDP bandwidth attribute. It can be selected from the drop-down list. The default setting is "Media
Level".
● Standard: use AS format in session level; use TIAS format in media level
● Media Level: use AS format in media level.
● Session Level: use AS format in session level.
● None: no modifications in the session format.
Note: Please do not modify this setting without knowing the session format supported by the server.
Otherwise, it might cause video decoding failure.
RTP Settings
Sets if the phone will enable the SRTP (Secured RTP) mode. It can be selected from dropdown list:
● Disable
● Enabled but not forced
● Enabled and forced
● Follow SIP transport
The default setting is "Disable".
● 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". It uses SDP
security description to exchange key. Please refer to:
SDES: https://tools.ietf.org/html/rfc4568
SRTP: https://tools.ietf.org/html/rfc3711
● If set to "Follow SIP transport," the device will use RTP/SRTP according to the SIP transport
configured. The SIP transport may be automatically selected when using DNS NAPTR mode. In this
case, the RTP/SRTP will be automatically selected as well.
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 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".