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DTMF Payload Type
Enable Audio RED with FEC
Audio FEC Payload Type
Audio RED Payload Type
Silence Suppression
Jitter Buffer Type
Jitter Buffer Length
Voice Frames Per TX
G.723 Rate
RTP Settings
SRTP Mode
SRTP Key Length
Crypto Life Time
2. RFC2833 sends DTMF with RTP packet. Users can check the RTP packet to see the
DTMFs sent as well as the number pressed.
3. SIP INFO uses SIP INFO to carry DTMF.
Default setting is "RFC2833".
Configures the payload type for DTMF using RFC2833. Cannot be the same as iLBC or
OPUS payload type.
If set to "Yes", FEC will be enabled for audio call.
Configures audio FEC payload type. The valid range is from 96 to 126.
The default value is 121.
Configures audio RED payload type. The valid range is from 96 to 126.
The default value is 124.
If 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. For codec G.723 and G.729 only.
Default setting is "No".
Selects either Fixed or Adaptive for jitter buffer type, based on network conditions. The
default setting is "Adaptive".
Selects jitter buffer length from 100ms to 800ms, based on network conditions. The default
setting is "300ms".
Configures the number of voice frames transmitted per packet. It is recommended that the IS
limit value of Ethernet packet is 1500 bytes or 120 kbps. 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 in the codec table or negotiate the payload type during the
actual call. For example, if set to 2 and the first code is G.729, G.711 or G.726, the "ptime"
value in the SDP datagram of the INVITE request is 20 ms. If the "Voice Frame/TX" setting
exceeds the maximum allowed value, the phone will use and save the maximum allowed value
for the selected first codec. It is recommended to use the default setting provided, and
incorrect setting may affect voice quality.
The default setting is 2.
Selects encoding rate for G723 codec.
Enable SRTP mode based on your selection from the drop-down menu.
● No
● Enabled but Not forced
● Enabled and Forced
● Optional
The default setting is "No".
Allows users to specify the length of the SRTP calls. Available options are:
● AES 128&256 bit
● AES 128 bit
● AES 256 bit
Default setting is AES 128&256 bit
Enable or disable the crypto lifetime when using SRTP. If users set to disable this option,
phone does not add the crypto lifetime to SRTP header. The default setting is "Yes".
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