Here is a follow up video demo - where Bill, Tom & Lyle are all QSO'n together in this unique way to have yourself a fine CW QSO with your CW buddies using Gstreamer

Experimental setup to have an online CW QSO with your friends, in full QSK, FULL DUPLEX, high quality CW audio fidelity using Gstreamer scripts AND send everyone a PICTURE of yourself too

here are the Gstreamer scripts blueprint used for this example:

SEND a jpeg picture to the "other" QRQ CW OP
*********************************************************************
gst-launch-1.0 -v filesrc location=/home/qrq/kb9xe.jpeg ! decodebin ! videoconvert ! imagefreeze ! jpegenc ! udpsink host=(ip address of the other QRQ CW OP) port=5007

SEND your QRQ CW AUDIO to the "other" QRQ CW OP
*********************************************************************
gst-launch-1.0 jackaudiosrc ! audio/x-raw,channels=1 ! audiorate ! audioconvert ! opusenc bitrate=256000 frame-size=2.5 ! rtpopuspay ! udpsink host=(ip address of the other QRQ CW OP) port=5008

RECEIVE jpeg picture from the "other" QRQ CW OP
*******************************************************************
gst-launch-1.0 -v udpsrc port=5007 ! decodebin ! videoconvert ! ximagesink

RECEIVE QRQ CW AUDIO from the "other" QRQ CW OP
***************************************************************
gst-launch-1.0 -v udpsrc caps="application/x-rtp,channels=1" port=5008 ! rtpjitterbuffer latency=60 ! queue ! rtpopusdepay ! opusdec plc=true ! audioconvert ! jackaudiosink buffer-time=30000

gstreamer:
https://gstreamer.freedesktop.org/

Here is another video in this series about QRQ over GSTREAMER:  this video demo'S a method for getting good "break - in" on the sending op - using a ducking control from a plugin called  "LOUDMAX"

- Brief demo of a QRQ CW "audio over ip" QSO using Gstreamer -

how to break in on the "sending" OP effectively
using a "Ducking control setup" by a PLUGIN called "LOUDMAX" https://loudmax.blogspot.com/

QRQ OVER GSTREAMER - PIPELINES:

here is the blueprint for these scripts:

TRANSMITTER:
gst-launch-1.0 jackaudiosrc ! audio/x-raw,channels=1 ! audiorate ! audioconvert ! opusenc bitrate=128000 frame-size=10 ! rtpopuspay ! queue ! udpsink host=(ip addres of the other op) port=5001

RECEIVER:
gst-launch-1.0 -v udpsrc caps="application/x-rtp,channels=1" port=5001 ! rtpjitterbuffer latency=30 ! queue ! rtpopusdepay ! opusdec ! audioconvert ! jackaudiosink buffer-time=40000