Overview
reestablished until the remote source goes inactive and again active.
If Forced Reconnect is enabled, PTT Max Duration must be set to “0”.
Call-Session
Verification Time
This monitors the IP connection of a paired link while a source is active.
(Heartbeat is used to monitor the IP path between links while the source is idle).
Valid range 5 to 60 seconds (1 second increments). Default 40 seconds.
Audio
Line Type The line type of the radio interface, either Unbalanced, Balanced 2-Wire, or
Balanced 4-wire (default). 4-wire audio is required if the IP Gateway is used with
a Zetron iRIM.
RX Pair Impedance This parameter is only available if Line Type is set to Balanced 4-wire.
Sets the RX Termination impedance of a 4-wire line to one of the following:
• Lo-Z (7.5K Ohms, default)
• Hi-Z (200K Ohms)
TX Pair Impedance This parameter is only available if Line Type is set to Balanced 2-Wire or
Balanced 4-Wire.
Sets the TX Termination impedance of a 4-wire line to one of the following:
• Lo-Z (7.5K Ohms, default)
• Hi-Z RX/Lo-Z TX (200K Ohms RX/7.5K Ohms TX).
Full Duplex Enable This parameter is only available if Line Type is set to Balanced 4-wire.
Enable for full-duplex radios. Disable for half-duplex radios.
Tx Pair Monitoring
Enable
This parameter is only available if Line Type is set to Balanced 4-wire.
This enables monitoring audio over the TX pair of wires. When multiple 4-wire
consoles are connected to a single radio, dispatchers should monitor the Tx pair
before they transmit in order to ensure a radio is not already busy.
Receive AGC Enable If this option is enabled, automatic gain control (AGC) will be applied to the RX
amplifier, so that the output signal level is constant despite varying input levels.
Default enabled.
RTP VoIP Vocoder
Selection
Determines the voice quality and bandwidth used. Higher bandwidth settings
result in higher quality. Lower bandwidth settings should only be used when
network bandwidth limitations are encountered. To ensure that the desired voice
quality is used, set this value the same for both the local and remote paired link.
The default setting 64 kbps PCM/G.711 is the only codec that supports MDC-
1200, DTMF, and Tone Remote Control signaling.
RTP VoIP Jitter Buffer
Duration
The buffer is used to reassemble IP voice packet which may be received out-of-
order due to the IP traffic taking different routes between paired links. The audio
to the analog circuit is necessarily delayed by this amount. The value should only
be reduced if the paired devices have very few network nodes between the paired
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