Grandstream Networks GXV3240 Administration Manual page 59

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H.264 Payload Type
H.263 Encoder
Resolution
SRTP Mode
SRTP Key Length
Enable SRTP Key Life
Time
Silence Suppression
Voice Frames Per TX
RTCP Destination
None: no modifications in the session format
The default setting is "Media Level".
Note: Please do not modify this setting without knowing the session format
supported by the server. Otherwise, it might cause video decoding failure.
Specifies the H.264 codec message payload type format.
The default setting is 99. The valid range is from 96 to 127.
Configures H263 encoder resolution to use.
The default setting is CIF.
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; the default setting is
"AES128&256 bit":
AES128&256 bit
AES 128 bit
AES 256 bit
If 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.
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 2
and phone will renew the SRTP key after this limitation. Default is "Yes".
Enables the silence suppression/VAD feature. If it is set to "Yes", when
silence is detected, a small quantity of VAD packets (instead of audio
packets) will be sent during the period of no talking. If set to "No", this
feature is disabled. The default setting is "No".
Configures the number of voice frames transmitted per packet. When
configuring this, it should be noted that the "ptime" value for the SDP will
change with different configurations here. This value is related to the codec
used and the actual frames transmitted during the in payload call. For end
users, it is recommended to use the default setting, as incorrect settings
may influence the audio quality. The default setting is 2.
Configures a remote server URI where RTCP messages will be sent to
during an active call.
GXV3240 Administration Guide
Version 1.0.3.207
31
SIP packets,
P a g e
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