rtspsrc: long and regular pauses when streaming from wifi action cams

[philippe renon]

Thanks to your help Sebastian, I was able to test with a local gstreamer rtspsrc server and the Qt client works like a charm.
The setup was:server:  ./test-launch.exe "( videotestsrc ! x264enc tune=zerolatency ! rtph264pay name=pay0 pt=96 )"
client:  gst-launch-1.0.exe -v -m rtspsrc location=rtsp://127.0.0.1:8554/test latency=30 ! decodebin ! timeoverlay ! autovideosink
I also tested with success a simple rtp scenario:
server : gst-launch-1.0.exe -v -m videotestsrc ! x264enc tune=zerolatency ! rtph264pay ! udpsink host=127.0.0.1 port=5000
client : gst-launch-1.0.exe -v -m udpsrc port=5000 ! application/x-rtp,payload=96,clock-rate=90000 ! rtpjitterbuffer ! rtph264depay ! decodebin ! videoconvert ! timeoverlay ! autovideosink 

But playing a rtsp stream from a wifi "action" camera still has those longish pauses every 10s.And contrary to what I wrote earlier, I can reproduce it with gst-launch 1.10 with both the msys2 build and the official distribution.

client : gst-launch-1.0.exe -v -m rtspsrc location=rtsp://192.x.x.x/AmbaStreamTest latency=30 ! decodebin ! timeoverlay ! autovideosink
Some findings:
Each time the pipeline "pauses", the video sink emits a qos event and drops one frame:
element autovideosink1-actual-sink-d3dvideo sent qos event: live: 1; running time: 30719164049; stream time: 26558070697; timestamp: 30719164049; duration: 33366666 jitter: 3029708354; proportion: 0.15581; quality: 1000000; format: ; processed: 609; dropped: 2;

The rtpjitterbuffer skew is initially around 8 as expected but then starts to increase to reach values in the 200.
I am attaching a dot file of the pipeline at the time a qos event was sent.

Is it possible to disable the rtpjitterbuffer used in the rtspsrc bin.If if not, is it possible to create a rtspsrc from its individual elements with gst-launch ?
Would a wireshark log help ?





    Le Lundi 7 novembre 2016 11h37, philippe renon <philippe_renon at yahoo.fr> a écrit :
 
 

 
I did try test-launch but was missing the parentheses and thus getting assertions that the video source element was not a bin and other such assertions.
Will try with parentheses...
And thanks for the quick and to the point answers.
 

    Le Lundi 7 novembre 2016 10h26, Sebastian Dröge <sebastian at centricular.com> a écrit :
 
 

 On Sun, 2016-11-06 at 11:51 +0000, philippe renon wrote:
> 
> To speed up my testing, I am trying to setup a small rtsp server using gstreamer.
> 
> But the following pipeline fails:
> 
> $ gst-launch-1.0.exe --gst-debug=3 videotestsrc ! x264enc ! rtph264pay name=pay0 pt=96
> [...]

This does not set up an RTSP server but is just an incomplete pipeline
without a sink.

Try using the test-launch example from gst-rtsp-server/examples, e.g.
  ./test-launch "( videotestsrc ! x264enc tune=zerolatency ! rtph264pay name=pay0 pt=96 )"

-- 
Sebastian Dröge, Centricular Ltd · http://www.centricular.com
_______________________________________________
gstreamer-devel mailing list
gstreamer-devel at lists.freedesktop.org
https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel


 
   

 
   
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20161109/4bbdecf8/attachment-0001.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: 0.00.39.921343831-pipeline_qos.dot
Type: application/msword
Size: 31787 bytes
Desc: not available
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20161109/4bbdecf8/attachment-0001.dot>