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Enable checking of SIP trunk device availability. It sends a SIP OPTIONS request regularly to check
that the SIP trunk is still online. If the device does not answer within the configured period GSM
Gateway Switch_VoIP considers the device off-line for future calls. Expiration time is configured in
seconds (default is 2 seconds). This feature may also be used to keep a UDP session open to the
trunk device located behind the NAT.
 DTMF mode
Select mode to transport DTMF digits.
 NAT
Allow communication with SIP proxy server behind the NAT.
 Need INVITE authentication
Enable authentication for incoming calls (incoming INVITE requests).
o Codecs
Select allowed voice compression codecs in order of preference.
 Support clearmode
Enable 64-kbit/s transparent data transfer over the RTP.
o Voice Processing
 Echo cancel
Enable/disable echo cancellation. Click the Advanced button to configure echo cancellation tail
length.
 VAD
Enable/disable Voice Activity Detection to avoid unnecessary coding/transmission of silence packets.
Click the Advanced button to tune VAD threshold.
 AGC
Enable/disable Automatic Gain Control.
 Jitter buffer
Enable/disable jitter buffer. Click the Advanced button to configure jitter buffer size.
 Noise suppression
Enable/disable noise suppression for better voice quality.
o Assignment to incoming/outgoing groups
It is possible to set each SIP trunk its Incoming Groups and Outgoing groups.
5.5.3.3
SI P Account Advanced Voice Options
This pane contains other options of SIP Account. This setting is not recommended to change.