Webrtc flooding lossy tunnel

[uzername123]

Question

i trying to send video over lossy lowspeed (5mbit only/50-80ms latency ) channel , using webrtc stream (as it udp and have internal FEC)
on sender part - stream just served as rtsp and webrtc

runOnDemand: gst-launch-1.0 ...... ! queue max-size-bytes=1024000 ! videoconvert ! x264enc bitrate=2048 bframes=0 tune=zerolatency speed-preset=ultrafast sliced-threads=true noise-reduction=5000 ! video/x-h264 ! h264parse ! rtspclientsink location=rtsp://localhost:8554/stream

on receiver - using string:

source: whep://$SENDER_IP:8889/stream/whep

so all clients connecting to local host that serving rtsp /rtmp/webrtc/hls /whatever
remote unit starting to send stream, but when appearing significant losses and delays in channel - seems mediamtx starting to buffering packets that are late because of delay/lossess, and sending it anyway, so it starting to using full channel bandwidth - easily use 5-10megabits, but stream is CBR 2mbit, how to fix that (discard packets that are too late? )?
also - sender printing such string when traffic increasing to max:
WAR [WebRTC] [session ffbc9fd8] write queue is full