VOICE CHAT by RTSP Server Client using the rtsp-simple-server from Alessandro Ros & Gstreamer Clien

Описание

this is a follow up video from previous video demos here:
https://youtu.be/VtizBdZPVec
https://youtu.be/sveyrDu5hSQ

https://github.com/aler9/rtsp-simple-server/releases
https://github.com/aler9/rtsp-simple-server
https://awesomeopensource.com/projects/gstreamer
https://github.com/aler9

in this video demo, 2 rtsp-simple-servers are created, one for each participant(1 on LINUX, 1 on windows 10) Rtsp Client Gstreamer scripts are used to connect to each other's rtsp-simple-server...and microphones are used as the RTSP SERVER's AUDIO SOURCE....full duplex, 2 way, low latency VOICE CHAT is tested between 2 people

win 10 rtsp Gstreamer MICrophone audio to RTSP SERVER SCRIPT
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C:\gstreamer\1.0\msvc_x86_64\bin
gst-launch-1.0 -v wasapisrc ! queue ! audioconvert ! opusenc bitrate=512000 ! queue ! rtspclientsink location=rtsp://127.0.0.1:8554/mystream latency=0

win 10 RTSPSRC script to receive the other person's VOICE/MIC
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C:\gstreamer\1.0\msvc_x86_64\bin
gst-launch-1.0 rtspsrc location=rtsp://(IP ADDRESS OF THE OTHER PERSON's PC):8554/mystream latency=0 ! "application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)OPUS, sprop-maxcapturerate=(string)48000, sprop-stereo=(string)1, payload=(int)96, encoding-params=(string)2" ! rtpjitterbuffer latency=10 ! rtpopusdepay ! opusdec ! audioconvert ! wasapisink buffer-time=10000

Linux rtsp Gstreamer MICrophone audio to RTSP SERVER SCRIPT
*******************************************************************
gst-launch-1.0 -v jackaudiosrc ! queue ! audioconvert ! opusenc bitrate=650000 ! rtspclientsink location=rtsp://127.0.0.1:8554/mystream latency=0

Linux RTSPSRC script to receive the other person's VOICE/MIC
****************************************************************
gst-launch-1.0 rtspsrc location=rtsp://(IP ADDRESS OF THE OTHER PERSON's PC):8554/mystream latency=0 ! "application/x-rtp, media=(string)audio, clock-rate=(int)48000, encoding-name=(string)OPUS, sprop-maxcapturerate=(string)48000, sprop-stereo=(string)1, payload=(int)96, encoding-params=(string)2" ! rtpjitterbuffer latency=10 ! rtpopusdepay ! opusdec ! audioconvert ! jackaudiosink buffer-time=10000

NOTE: the windows 10 setup was using the laptop's cheap internal MIC on the lid...
NOTE: the linux setup was using a cheap headset MIC

for more info about Gstreamer on windows 10 - see the playlist here:
https://www.youtube.com/playlist?list=PLfFan4sDonOccKApSxnb8mZyUS3CLxSMF