Files

T

Alessandro RosandGitHub b5b63d02fc support reading and publishing with Media-over-QUIC (#5815 )

Media-over-QUIC is a streaming protocol built upon cutting edge
protocols (QUIC, HTTP3) and browser APIs (WebTransport, WebCodecs).
It's slightly faster than WebRTC, has an advanced data recovery
mechanism (placed at the frame level and not at the packet level), it
supports additional codecs (FLAC) and is less complicated to route.

2026-06-02 23:04:24 +02:00

2.2 KiB

Raw Blame History

GStreamer

GStreamer can publish a stream to the server by acting as a RTSP client, RTMP client, SRT client, WebRTC client or by sending MPEG-TS packets. The recommended way is acting as a RTSP client.

GStreamer as a RTSP client

gst-launch-1.0 rtspclientsink name=s location=rtsp://localhost:8554/mystream \
filesrc location=file.mp4 ! qtdemux name=d \
d.video_0 ! queue ! s.sink_0 \
d.audio_0 ! queue ! s.sink_1

If the stream is video only:

gst-launch-1.0 filesrc location=file.mp4 ! qtdemux name=d \
d.video_0 ! rtspclientsink location=rtsp://localhost:8554/mystream

The resulting stream will be available on path /mystream.

GStreamer as a RTMP client

gst-launch-1.0 -v flvmux name=mux ! rtmpsink location=rtmp://localhost/stream \
videotestsrc ! video/x-raw,width=1280,height=720,format=I420 ! x264enc speed-preset=ultrafast bitrate=3000 key-int-max=60 ! video/x-h264,profile=high ! mux. \
audiotestsrc ! audioconvert ! avenc_aac ! mux.

GStreamer as a SRT client

gst-launch-1.0 -v mpegtsmux name=mux ! srtsink uri="srt://localhost:8890?streamid=publish:mystream&pkt_size=1316" \
videotestsrc ! video/x-raw,width=1280,height=720,format=I420 ! x264enc speed-preset=ultrafast bitrate=3000 key-int-max=60 ! video/x-h264,profile=high ! mux. \
audiotestsrc ! audioconvert ! avenc_aac ! mux.

The resulting stream will be available on path /mystream.

GStreamer and MPEG-TS over UDP

gst-launch-1.0 -v mpegtsmux name=mux alignment=1 ! udpsink host=238.0.0.1 port=1234 \
videotestsrc ! video/x-raw,width=1280,height=720,format=I420 ! x264enc speed-preset=ultrafast bitrate=3000 key-int-max=60 ! video/x-h264,profile=high ! mux. \
audiotestsrc ! audioconvert ! avenc_aac ! mux.

GStreamer as a WebRTC client

Make sure that GStreamer version is at least 1.22, and that if the codec is H264, the profile is baseline. Use the whipclientsink element:

gst-launch-1.0 videotestsrc \
! video/x-raw,width=1920,height=1080,format=I420 \
! x264enc speed-preset=ultrafast bitrate=2000 \
! video/x-h264,profile=baseline \
! whipclientsink signaller::whip-endpoint=http://localhost:8889/mystream/whip

2.2 KiB Raw Blame History