I have a working pipeline begins with :
gst-launch-1.0 rtspsrc location=rtsp://Test_Map:T3$t1ng_2022_Map#x.x.x.x:zzzz/ISAPI/Streaming/Channels/101
which gives Unauthorized error. But vlc can open same stream .
any idea?
gst-launch should open the rtsp
You should use the user-id and user-pw argument with proper ""
than it will work.
I have a parrot drone that has 4 cameras being streamed with RTSP. The 4 cameras are being streamed by the same location (rtsp://192.168.53.1:554/live). Is it possible to choose which stream I want to be displayed using gst-launch-1.0?
When I run:
gst-launch-1.0 rtspsrc location=rtsp://192.168.53.1:554/live latency=0 ! queue ! rtph264depay ! h264parse ! avdec_h264 ! videoconvert ! videoscale ! video/x-raw,width=1920,height=1080 ! autovideosink
I can see one of the streams, the problem is that the stream displayed is a random one from the four.
The output from the gst-launch-1.0 is the following:
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Progress: (open) Opening Stream
Progress: (connect) Connecting to rtsp://192.168.53.1:554/live
Progress: (open) Retrieving server options
Progress: (open) Retrieving media info
Progress: (request) SETUP stream 0
Progress: (request) SETUP stream 1
Progress: (request) SETUP stream 2
Progress: (request) SETUP stream 3
Progress: (open) Opened Stream
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Progress: (request) Sending PLAY request
Progress: (request) Sending PLAY request
Progress: (request) Sent PLAY request
The following images are frames that I extracted from 2 of the 4 streams where, one time the displayed stream was the one from the main camera and another time the one from stereo.
Frame of the main camera stream:
Frame of one stereo camera stream:
I have a Ricoh THETA Z1 360 degrees camera that outputs a 4K 360 stream. I'm using their own libuvc-theta-sample for retrieving the video stream and getting it into Gstreamer. I've used the following pipeline to sink the video stream to a different machine on my network that runs a Gstreamer application that restreams udpsrc into a webrtcbin:
appsrc name=ap ! queue max-size-buffers=1 leaky=downstream ! h264parse ! rtph264pay ! udpsink host=192.168.1.137 port=1935 qos=false sync=false
On my receiving end this works perfectly fine:
As you can see I'm using a WebRTC bin that sinks the video to a WebRTC browser client.
This is all working as expected as long as I'm on localhost. If I try to reach the video stream from a different machine in my network, it does the ICE and SDP offers perfectly, the WebRTC connection is established, but no video is shown. If I replace my pipeline with a pipeline that depays, decodes, and re-encodes the H264 stream, it works perfectly fine. Obviously, this is not what I want because I'm losing (significant) video quality and it takes time to re-encode.
Something that might help discover the issue is the following log that gets spammed in my console. This log doesn't appear when I'm viewing the stream from localhost, but it does appear when trying to view the stream from a different machine within my network.
0:00:07.248383000 22700 00000139F5F3E340 INFO h264parse gsth264parse.c:3741:gst_h264_parse_src_event: received upstream force-key-unit event, seqnum 253 running_time 99:99:99.999999999 all_headers 0 count 0
0:00:07.559580000 22700 00000139F5F3E340 INFO h264parse gsth264parse.c:3741:gst_h264_parse_src_event: received upstream force-key-unit event, seqnum 254 running_time 99:99:99.999999999 all_headers 0 count 0
0:00:07.854883000 22700 00000139F5F3E340 INFO h264parse gsth264parse.c:3741:gst_h264_parse_src_event: received upstream force-key-unit event, seqnum 255 running_time 99:99:99.999999999 all_headers 0 count 0
0:00:08.160170000 22700 00000139F5F3E340 INFO h264parse gsth264parse.c:3741:gst_h264_parse_src_event: received upstream force-key-unit event, seqnum 256 running_time 99:99:99.999999999 all_headers 0 count 0
0:00:08.414529000 22700 00000139F5F3E340 INFO h264parse gsth264parse.c:3741:gst_h264_parse_src_event: received upstream force-key-unit event, seqnum 257 running_time 99:99:99.999999999 all_headers 0 count 0
0:00:08.715480000 22700 00000139F5F3E340 INFO h264parse gsth264parse.c:3741:gst_h264_parse_src_event: received upstream force-key-unit event, seqnum 258 running_time 99:99:99.999999999 all_headers 0 count 0
0:00:09.013550000 22700 00000139F5F3E340 INFO h264parse gsth264parse.c:3741:gst_h264_parse_src_event: received upstream force-key-unit event, seqnum 259 running_time 99:99:99.999999999 all_headers 0 count 0
I've thought about NAT traversal issues with STUN and TURN, but it wouldn't make sense as my Google Chrome WebRTC internals page shows that the WebRTC connection is successfully established. Besides, it does work when re-encoding the stream.
I'm failing to understand why it is working on localhost, but not on other machines within my network. Perhaps someone can shed some light on why it's not working.
Disable firewall on streaming server and client machine then test streaming works or not. If works then you can add your firewall rules for WebRTC and UDP ports .
2)Try streaming with creating direct tunnel using ngrok or other free service with direct IP addresses. Then go with STUN and TURN setup.
Also try following pipeline:
webrtcbin name=sendrecv bundle-policy=max-bundle
udpsrc port=7001 caps = "application/x-rtp, media=(string)video, clock-rate=(int)90000, encoding-name=(string)H264, payload=(int)96" ! rtph264depay !
h264parse ! rtph264pay config-interval=-1 !
queue ! application/x-rtp,media=video,encoding-name=H264,payload=96 ! rtpjitterbuffer ! sendrecv
Not sure at all for your case, but you may try adding rtpjitterbuffer with a few frames latency in the remote case:
... ! udpsrc port=1935 ! application/x-rtp,encoding-name=H264 ! rtpjitterbuffer latency=500 ! rtph264depay ! ...
I'm trying to stream video using gstreamer to html usign busybox in petalinux, within of a zynqmp based platform.
So the main idea is to stream a video pipeline into an embedded web server and open it from a external PC web browser (I'm using chrome).
The index.html file looks like that:
<!DOCTYPE html>
<html>
<p> video test ! </p>
<body>
<video autoplay controls width=1280 height=1024>
<source src="http://192.168.1.12:5000" type="video/mp4">
Your browser does not support the video tag.
</video>
</body>
</html>
And the testing pipeline is:
gst-launch-1.0 -v videotestsrc pattern=snow is-live=true ! video/x-raw,width=1280,height=1024 ! theoraenc ! oggmux ! tcpserversink host=192.168.1.12 port=5000
Also I tried to stream a video camera live from a C application like so:
appsrc -> queue -> omxh264 -> h264parser -> mp4mux (streamable=true fragment-duration = 100) -> queue -> tcpserversink (host=192.168.1.12 port=5000)
(Please note that the above line is a pseudo code to illustrate what the application is doing)
I'm expecting to see the video in the PC web browser but unfortunately I've got a grey box with the video controls instead.
Does anyone know if this would be possible using busybox? or is there something fundamentally wrong with this approach?
I have found the way to stream the videotestsrc element to the internet by calling these functions in the test-video.c file of gst-rtsp-server examples directory:
gst_rtsp_server_set_address(server,"10.xxx.xx.xxx");
gst_rtsp_server_set_service(server,"8555");
and unblocking the ports in the aws console.
But now replacing videotestsrc with the following in test-video.c, like below:
gst_rtsp_media_factory_set_launch (factory, "( "
"filesrc location=/home/ubuntu/DELTA.mpg ! mpeg2dec ! x264enc ! rtph264pay name=pay0 pt=96 "
")");
I am not able to get the stream visible on the vlc player.
Trying the below command produces following output:
ubuntu#ip-xx-xxx-xx-xxx:~/gst-rtsp-server-1.2.3/examples$ /usr/local/bin/gst-launch-1.0 playbin uri=rtsp://10.xxx.xx.xxx:8555/test
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Progress: (open) Opening Stream
Progress: (connect) Connecting to rtsp://10.185.10.118:8555/test
Progress: (open) Retrieving server options
Progress: (open) Retrieving media info
Progress: (request) SETUP stream 0
Progress: (open) Opened Stream
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Progress: (request) Sending PLAY request
Progress: (request) Sending PLAY request
Progress: (request) Sent PLAY request
(lt-test-video1:9027): GLib-CRITICAL **: unblock_source: assertion `!SOURCE_DEST
ROYED (source)' failed
WARNING: from element /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSP
Src:source: Could not read from resource.
Additional debug info:
gstrtspsrc.c(4367): gst_rtspsrc_reconnect (): /GstPlayBin:playbin0/GstURIDecodeB
in:uridecodebin0/GstRTSPSrc:source:
Could not receive any UDP packets for 5.0000 seconds, maybe your firewall is blo
cking it. Retrying using a TCP connection.
♥handling interrupt.
Interrupt: Stopping pipeline ...
Execution ended after 0:00:22.473442001
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
ubuntu#ip-10-xxx-xx-xxx:~/gst-rtsp-server-1.2.3/examples$
What am I doing wrong here ?