When using the following setup (the error can be reproduced using
simpler sender pipelines), the receiver resynchronises the clock on RTCP
packets. The effect was that a couple seconds were cut out of the
playback because an initial RTCP packet was dropped.
When sending out all RTCP packets (setting sync=FALSE on the RTCP
updsink), the playback is fine.
This syncs rtpsink with rtpsrc (where this property was already set).
gst-launch-1.0 filesrc location=899-en.mp3 \
! mpegaudioparse \
! mpg123audiodec \
! audioconvert \
! audioresample \
! avenc_g722 \
! rtpg722pay
! rtpsink uri=rtp://239.1.2.3:1234
gst-launch-1.0 uridecodebin rtp://239.1.2.3:1234?encoding-name=G722 \
! autoaudiosink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/993>
