I set up the following pipelines:
Server
gst-launch-1.0 -v audiotestsrc ! mulawenc ! tcpserversink host=<IP> port=<PORT>
Client
gst-launch-1.0 tcpclientsrc host=<IP> port=<PORT> ! "audio/x-mulaw, rate=44100, channels=1" ! mulawdec ! audioconvert ! audioresample ! alsasink(windows: directsoundsink)
Using a Windows server and a Linux client I only get a sound for a fraction of a second after closing the pipeline (doesn't matter if I close client or server first). When running both server and client on the same Windows machine I get the same result. However, if I run both server and client on the same Linux machine I get the expected result (sin signal).
What do I need to change to make it work on Windows?
Related
I have a little project where I try to make a low latency video stream from a raspberry pi to a PC. Here for I’m trying Gstreamer. Now I have a working Pipeline on windows with the following command on the pi:
gst-launch-1.0 -v v4l2src device=/dev/video2 ! "image/jpeg,width=1280, height=720,framerate=30/1" ! rtpjpegpay ! udpsink host=192.168.1.94 port=5004
And on the windows receiving site:
gst-launch-1.0 -e -v udpsrc port=5004 ! application/x-rtp, encoding-name=JPEG,payload=26 ! rtpjpegdepay ! jpegdec ! autovideosink
Windows client waiting for server
Windows client and server connected
_
Bud when I try to use the same command on ubuntu it looks like it makes connection bud it won’t open a window with the render.
Ubuntu client waiting for server
Ubuntu client and server connected
Can someone help me get this to work?
Thank you in advance.
Im working on this project :
https://www.hackster.io/jonmendenhall/jetson-nano-search-and-rescue-ai-uav-9ca547
At some point I will need to mount my camera (waveshare ; IMX219-77IR) on top of the drone and I would like to use vlc on Windows or Linux outside of nomachine (because I have installed nomachine server on the nano and the client on windows and because it will run in headless mode),to display what the camera sees when the drone is flying. For this reason I’m trying to configure a gstreamer with RTSP to start a streaming server on the Ubuntu 18.04 that I have installed on the jetson nano.
Below u can see what are the commands that I have issued :
$ ./test-launch "videotestsrc ! nvvidconv ! nvv4l2h264enc ! h264parse ! rtph264pay name=pay0 pt=96"
And on the same Jetson Nano, I opened another console where I ran this pipeline to decode the RTSP stream:
gst-launch-1.0 uridecodebin uri=rtsp://127.0.0.1:8554/test ! nvoverlaysink
I see this picture :
The picture is from videotestsrc plugin. I would like to replace videotestsrc with my video source,but I don't know how to do that.
I tried these combinations,but none of them worked :
./test-launch "v4l2src device=/dev/video0 ! nvvidconv ! nvv4l2h264enc ! h264parse ! queue ! rtph264pay name=pay0 pt=96"
./test-launch "device=/dev/video0 ! nvvidconv ! nvv4l2h264enc ! h264parse ! queue ! rtph264pay name=pay0 pt=96"
but the error is still the same :
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Progress: (open) Opening Stream
Progress: (connect) Connecting to rtsp://127.0.0.1:8554/test
Progress: (open) Retrieving server options
Progress: (open) Retrieving media info
ERROR: from element /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source: Could not read from resource.
Additional debug info:
gstrtspsrc.c(5917): gst_rtsp_src_receive_response (): /GstPlayBin:playbin0/GstURIDecodeBin:uridecodebin0/GstRTSPSrc:source:
Could not receive message. (Timeout while waiting for server response)
ERROR: pipeline doesn't want to preroll.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
Freeing pipeline ...
but why ? I know for sure that my camera (model waveshare ; IMX219-77IR) created a device called /dev/video0 and I know for sure that it works,because this command is able to show my face on the screen :
DISPLAY=:0.0 gst-launch-1.0 nvarguscamerasrc ! 'video/x-raw(memory:NVMM), width=3280, height=2464, format=(string)NV12, framerate=(fraction)20/1' ! nvoverlaysink -e
In the end I would like to take an RSTP stream from an Axis security camera, and then make it available for re streaming using Gstreamers GstRTSPServer object. For now though I cannot even get the example videotestsrc to play in any player.
I have been following the post found here:
GStreamer Launch RTSP Server for ReStreaming IP Camera H264
I have gotten the example code built and running in a console, but am unable to view the stream in any players I find. EXACT code I am using found here:
https://github.com/GStreamer/gst-rtsp-server/blob/master/examples/test-readme.c
Both Windows 10 computers I am using are on the same network, I can ping both directions, firewall is completely off on both computers.
I have tried VLC player, Axis Media Control SDK modified examples, and GStreamer
using commands:
gst-launch-1.0 rtspsrc location=rtsp://127.0.0.1:8554/test latency=0 ! rtph264depay ! avdec_h264 ! videoconvert ! autovideosink
gst-launch-1.0 rtspsrc location=rtsp://10.2.3.10:8554/test latency=0 ! rtph264depay ! avdec_h264 ! videoconvert ! autovideosink
VLC is stating in logs that the connection was actively refused. It seems it is trying to connect to port 80? Not sure if I am reading that correctly. See VLC log below. Full log is pasted here from local machine:
https://pastebin.com/my3kTsRV
and here from a second machine:
https://pastebin.com/g1q47UBm
...Connection to server failed: No connection could be made because the target machine actively refused it
Opening connection to 127.0.0.1, port 80...
...Connection to server failed: No connection could be made because the target machine actively refused it
[06421494] live555 demux debug: connection error -10057
[06421494] live555 demux error: Failed to connect with rtsp://127.0.0.1:8554/test
The Axis Media Control (AMC) is trowing this for an error on both the local and secondary computers:
503 Service Unavailable
The maximum number of clients are already connected.
Trying to connect using the GStreamer command I get the following error, on both local and secondary computers:
ERROR: from element /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0: Unhandled error
Additional debug info:
gstrtspsrc.c(5633): gst_rtspsrc_send (): /GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0:
Service Unavailable (503)
ERROR: pipeline doesn't want to preroll.
With VLC, AMC and GStreamer I am able to pull a stream directly from a camera, so something works at least. I am also able to successfully run other GStreamer examples, without the GstRTSPServer, successfully.
Let me know if you want/need me to try anything.
I am going to use multiple clients on different computers to be able to view video of an IP Camera stream url. Because the Ip camera has limitations on the number of connected clients, I want to setup a streamer for this purpose. I googled and tried GStreamer with different command line options but not yet successful.
Here is a test command line:
gst-launch-1.0 rtspsrc
location="rtsp://root:root#192.168.1.1/axis-media/media.amp?videocodec=h264&resolution=320x240&fps=10&compression=50"
latency=10 ! rtph264depay ! h264parse ! tcpserversink
host=127.0.0.1 port=5100 -e
But when I want to test it with vlc, nothing is played. Is it related to SDP? Does gstreamer can restream sdp from source?
After finding the correct command line, I want to integrate it into a c# application to automate this process.
Any help is welcome.
You need gst-rtsp-server. And to use it you have to write small C/C++ application - example here
upd: If your rtsp source provide h264 video stream you could use following pipeline to restream it without transcoding:
rtspsrc location=rtsp://example.com ! rtph264depay ! h264parse ! rtph264pay name=pay0 pt=96
To re-stream h.264 video from IP camera, below is the Gstreamer pipeline (this works for me)
rtspsrc location=rtsp://IP_CAMERA_URL ! rtph264depay ! video/x-h264, stream-format=byte-stream ! h264parse ! rtph264pay ! application/x-rtp,media=video,encoding-name=H264,payload=96 ! yoursink
On gst-launch-1.0 --version --->
gst-launch-1.0 version 1.14.5
GStreamer 1.14.5
I need to set up a live audio streaming server with gstreamer. Server should be sending live audio to client and at the client side, vlc player should be used to play the incoming stream. I am using the following code
VIDEO_CAPS="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264"
gst-launch -v udpsrc caps=$VIDEO_CAPS port=4444 \
! gstrtpbin .recv_rtp_sink_0 \
! rtph264depay ! ffdec_h264 ! xvimagesink
then gstreamer reports like:
Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock
Please help me with steps for setting up a server using gstreamer a client for performing live streaming
Try reading manual on streaming with VLC here.
Or just:
cvlc rtp://#:4444
Update:
Due to my bad reading skills I slightly misunderstood the question.
Here is how to set up a server:
gst-launch -v pulsesrc ! audioconvert ! audioresample \
! speexenc ! rtpspeexpay \
! udpsink host=224.1.1.1 port=4444 auto-multicast=true
or use multiudpsink to send to multiple clients.