Choppy audio due to Jitter

[Joe Dontz]

UniMRCP version 1.7

The unimrcpserver.xml file is configured for adaptive jitter buffer.  We are having problems with the caller audio not being picked up and/or distorted by our ASR.  When we do a packet capture at the server level and listen, we do hear the complete audio in Wireshark.  In Wireshark, on the same call if we adjust the  jitter buffer down to 0 from 50, we start to loose pieces of the conversation; which is what we are seeing in production today.  A jitter-buffer config of 5 or 10 appears to work.  This lead us to believe that maybe the adaptive jitter buffer settings within Uni are not working for us.  If we turn off adaptive jitter-buffer, what is the correct syntax to manually set the value?  Any other thoughts or things we should check?

We're using the "Vosk Plugin", this is the relevant part: <settings>

    <!-- RTP/RTCP settings -->

    <rtp-settings id="RTP-Settings-1">

      <jitter-buffer>

        <adaptive>1</adaptive>

        <playout-delay>50</playout-delay>

        <max-playout-delay>600</max-playout-delay>

        <time-skew-detection>1</time-skew-detection>

      </jitter-buffer>