146 Matrix ETERNITY NE System Manual
• SIP - Call Progress Tones (Tx-Gain and Rx-Gain): Configure the Gain Setting that you want the
system to apply on the SIP Extension 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 Extension when it 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 Extension when it 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 Extension when it is connected to the SIP Trunk or any other 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 Extension when it 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.
• 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.
• Forced: Select Forced if you want SARVAM UCS to use only SRTP (SAVP) for transporting the
speech packets. If the remote user does not support SRTP, SARVAM UCS will reject incoming calls
from and drop outgoing calls made to such users.
By default, SRTP Mode is Disabled.
• To apply Echo Cancellation for SIP to CO trunk calls, SIP to Digital Trunks (Mobile, SIP) and Extensions
(SIP, SIP).
• Keep the Echo Cancellation check box enabled. Default: Enabled.
• Select Echo Cancellation Tail Length (msec) for CO trunks. It may be 32, 64, or 128 milliseconds.
Default: 128. milliseconds.
• Select Echo Cancellation Tail Length (msec) for Extensions and Digital Trunks. It may be 32, 64, or
128 milliseconds. Default: 32 milliseconds.
• Configure Jitter Buffer
32
to cut down on packet delays and improve voice quality.
To Next Page
Loading...
