srtsink fails to listen on ipv6 address ::
$ gst-launch-1.0 --no-position pulsesrc device=alsa_output.pci-0000_00_1f.3.analog-stereo.monitor ! 'audio/x-raw,format=S16LE,rate=48000,channels=2' ! opusenc bitrate-type=2 bitrate=256000 ! oggmux ! queue ! srtsink uri='srt://[::]:1234/?mode=listener' wait-for-connection=false
Setting pipeline to PAUSED ...
ERROR: from element /GstPipeline:pipeline0/GstSRTSink:srtsink0: Could not open resource for writing.
Additional debug info:
../gstreamer/subprojects/gst-plugins-bad/ext/srt/gstsrtsink.c(160): gst_srt_sink_start (): /GstPipeline:pipeline0/GstSRTSink:srtsink0:
1234
ERROR: pipeline doesn't want to preroll.
ERROR: from element /GstPipeline:pipeline0/GstSRTSink:srtsink0: GStreamer error: state change failed and some element failed to post a proper error message with the reason for the failure.
Additional debug info:
../gstreamer/subprojects/gstreamer/libs/gst/base/gstbasesink.c(5881): gst_base_sink_change_state (): /GstPipeline:pipeline0/GstSRTSink:srtsink0:
Failed to start
ERROR: pipeline doesn't want to preroll.
Failed to set pipeline to PAUSED.
Setting pipeline to NULL ...
Freeing pipeline ...
It'll work if the uri is srt://[::1]:1234/?mode=listener
or srt://0.0.0.0:1234/?mode=listener
, but not what I want.
Arch Linux with packages:
- gst-plugins-bad 1.22.6-1
- srt 1.5.3-1