Matrix ETERNITY NE System Manual 345
• SIP - System (Tx-Gain and Rx-Gain): Configure the Gain Settings that you want the system to apply on
the SIP Trunk with respect to the system (for example Call Conference). These will not be applicable for
any other port type. Valid Range for Tx Gain: -31dB to +31dB and Rx Gain -31dB to +31dB. Default: 0dB.
• SIP - Voice Mail (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply on
the SIP Trunk when the incoming calls on the SIP Trunk are being answered by the VMS. Valid Range for
Tx Gain: -31dB to +31dB and Rx Gain -31dB to +31dB. Default: 0dB.
• SIP - Voice Module (Tx-Gain and Rx-Gain): Configure the Gain Settings that you want the system to
apply on the SIP Trunk when incoming calls are answered using Auto Attendant or Trunk Auto Answer.
Valid Range for Tx Gain: -31dB to +31dB and Rx Gain -31dB to +31dB. Default: 0dB.
• SIP - Call Progress Tones (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system
to apply on the SIP Trunk while playing Call Progress Tones. Valid Range for Tx Gain: -31dB to +31dB
and Rx Gain -31dB to +31dB. Default: 0dB.
• SIP - SLT (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply on the
SIP Trunk when the SIP Trunk is connected to any FXS Port of the system during an incoming or outgoing
call. Valid Range for Tx Gain: -31dB to +31dB and Rx Gain -31dB to +31dB. Default: 0dB.
• SIP - CO (Tx-Gain and Rx-Gain): Configure the Gain setting that you want the system to apply on the SIP
Trunk when the SIP Trunk is connected to the CO Port of the system during an incoming or outgoing call.
Valid Range for Tx Gain: -31dB to +31dB and Rx Gain -31dB to +31dB. Default: 0dB.
• SIP - SIP (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply on the
SIP Trunk when the SIP Trunk is connected to any other SIP Trunk or SIP Extension of the system during
an incoming or outgoing call. Valid Range for Tx Gain: -31dB to +31dB and Rx Gain -31dB to +31dB.
Default: 0dB.
• SIP - Mobile (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the system to apply on the
SIP Trunk when the SIP Trunk is connected to any Mobile Port of the system during an incoming or
outgoing call. Valid Range for Tx Gain: -31dB to +31dB and Rx Gain -31dB to +31dB. Default: 0dB.
SRTP
SARVAM UCS supports SRTP (Secure Real Time Protocol) for secure conversations over SIP. The VoIP module
of SARVAM UCS supports the following options:
• Disable: Select Disable if you want SARVAM UCS to use normal RTP for transporting the speech
packets.
• Optional: Select Optional if you want SARVAM UCS to use SRTP for transporting the speech packets. If
the remote user does not support SRTP, SARVAM UCS will use normal RTP for transporting the speech
packets.
• If you select this option, you must configure the SRTP Media Type. You may select AVP or SAVP.
Default: AVP.
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