Re: How can I install WebRTCSink on Ubuntu 22.04

In order to use gcc, you need to also build the net/rtp crate, and have
the `rtpgccbwe` element available.

On Mon, 2023-07-10 at 11:08 +0100, GST Developer wrote:
> Hi Mathieu
> 
> You are correct - the issue was using an older version of webrtcsink
> (from a different git) with the current signalling server - perhaps
> the case/syntax has changed.
> 
> We have rebuilt everything from scratch many times and after more
> work, we have it working !  Many thanks for your input.
> 
> One thing we have discovered is that the default congestion control
> is causing us a problem:
> if we use congestion-control=gcc, gstreamer crashes.
> With congestion-control=homegrown or disabled, it does not crash
> 
> congestion-control=gcc : 
> ERROR: from element
> /GstPipeline:pipeline0/GstAudioTestSrc:audiotestsrc0: Internal data
> stream error.
> Additional debug info:
> ../libs/gst/base/gstbasesrc.c(3127): gst_base_src_loop ():
> /GstPipeline:pipeline0/GstAudioTestSrc:audiotestsrc0:
> streaming stopped, reason error (-5)
> ERROR: from element /GstPipeline:pipeline0/GstWebRTCSink:ws: Panicked
> 
> Otherwise, everything works well, thanks again.
> 
> One final question about the web server component - we have set this
> to use http instead of https since we dont need camera access, and
> all works well - however, it throws an error in the web page if we
> connect via the machine's dns name ("Invalid Host header”). It will
> only work with the external IP address of the machine, even tho the
> DNS resolves to this same address.  This same error happens in the
> default https mode, although this was more expected since the dns
> doesnt match the self-signed certificate.
> 
> We will dig into that code today, but if you have a quick answer to
> this, it would be very useful.
> 
> Many thanks again for this excellent code and your support.
> 
> > On 10 Jul 2023, at 10:53, Mathieu Duponchelle
> > <[email protected]> wrote:
> > 
> > Are you sure you are using the correct version of webrtcsink? Did
> > you
> > build it from the repo and have your plugin path point to this
> > version?
> > 
> > On Sun, 2023-07-09 at 09:09 +0100, GST Developer wrote:
> > > Thanks for your guidance Mathieu !!!
> > > 
> > > I have been able to work through the documentation you linked to
> > > and
> > > have been able to build the gst-plugins-rs components and get
> > > 2/3rd
> > > of the example running.
> > > Referring to the 3 shell windows mentioned here:
> > > gstreamer.freedesktop.org
> > > 
> > > 
> > > 
> > > The first shell is working - the signalling server appears to be
> > > running
> > > 
> > > The second shell is working, the gstwebrtc-api appears to be
> > > running
> > > a webserver, and I can connect to the page on port 9090 with a
> > > web
> > > gui showing a potential “start capture” and a “remote streams”
> > > section
> > > 
> > > However, The third does not work:
> > > 
> > > gst-launch-1.0 webrtcsink name=ws meta="meta,name=gst-stream"
> > > videotestsrc ! ws. audiotestsrc ! ws.
> > > 
> > > attempts to start, and I can see it has contacted the signalling
> > > server, but it fails with the error sequence below:
> > > 
> > > Setting pipeline to PAUSED ...
> > > Pipeline is live and does not need PREROLL ...
> > > Pipeline is PREROLLED ...
> > > Setting pipeline to PLAYING ...
> > > New clock: GstSystemClock
> > > Redistribute latency...
> > > Redistribute latency...
> > > ERROR: from element /GstPipeline:pipeline0/RsWebRTCSink:ws:
> > > GStreamer
> > > encountered a general stream error.
> > > Additional debug info:
> > > plugins/src/webrtcsink/imp.rs(1240): webrtcsink::webrtcsink::imp
> > > ():
> > > /GstPipeline:pipeline0/RsWebRTCSink:ws:
> > > Signalling error: Unknown message from server:
> > > {"type":"welcome","peerId":"384ad93b-1089-4fb8-93f9-
> > > 8a3e9ed709ba"}
> > > Execution ended after 0:00:00.122527906
> > > Setting pipeline to NULL ...
> > > Freeing pipeline …
> > > 
> > > meanwhile, the signalling server has responded:
> > > 
> > > 2023-07-09T08:03:49.938533Z  INFO ThreadId(01)
> > > gst_webrtc_signalling_server: Accepting connection from
> > > 127.0.0.1:39162
> > > 2023-07-09T08:03:49.938717Z DEBUG ThreadId(61) accept_async:
> > > gst_plugin_webrtc_signalling::server: new
> > > 2023-07-09T08:03:49.938901Z DEBUG ThreadId(61) accept_async:
> > > tungstenite::handshake::server: Server handshake done.    
> > > 2023-07-09T08:03:49.938926Z  INFO ThreadId(61) accept_async:
> > > gst_plugin_webrtc_signalling::server: New WebSocket connection
> > > this_id=0d76097e-e8fe-45d5-9769-b184bdc30ba8
> > > 2023-07-09T08:03:49.938959Z DEBUG ThreadId(61) accept_async:
> > > gst_plugin_webrtc_signalling::server: close time.busy=192µs
> > > time.idle=57.0µs
> > > 2023-07-09T08:03:49.939310Z  INFO ThreadId(56)
> > > gst_plugin_webrtc_signalling::server: Received message
> > > Ok(Text("{\"type\":\"setPeerStatus\",\"roles\":[\"producer\"],\"m
> > > eta\
> > > ":{\"name\":\"gst-stream\"}}"))
> > > 2023-07-09T08:03:49.939379Z DEBUG ThreadId(56)
> > > set_peer_status{peer_id="0d76097e-e8fe-45d5-9769-b184bdc30ba8"
> > > status=PeerStatus { roles: [Producer], meta: Some(Object {"name":
> > > String("gst-stream")}), peer_id: None }}:
> > > gst_plugin_webrtc_signalling::handlers: new
> > > 2023-07-09T08:03:49.939400Z  INFO ThreadId(56)
> > > set_peer_status{peer_id="0d76097e-e8fe-45d5-9769-b184bdc30ba8"
> > > status=PeerStatus { roles: [Producer], meta: Some(Object {"name":
> > > String("gst-stream")}), peer_id: None }}:
> > > gst_plugin_webrtc_signalling::handlers: registered as a producer
> > > peer_id=0d76097e-e8fe-45d5-9769-b184bdc30ba8
> > > 2023-07-09T08:03:49.939419Z DEBUG ThreadId(56)
> > > set_peer_status{peer_id="0d76097e-e8fe-45d5-9769-b184bdc30ba8"
> > > status=PeerStatus { roles: [Producer], meta: Some(Object {"name":
> > > String("gst-stream")}), peer_id: None }}:
> > > gst_plugin_webrtc_signalling::handlers: close time.busy=26.3µs
> > > time.idle=14.2µs
> > > 2023-07-09T08:03:49.981322Z DEBUG ThreadId(61)
> > > tungstenite::protocol:
> > > Received close frame: None    
> > > 2023-07-09T08:03:49.981347Z DEBUG ThreadId(61)
> > > tungstenite::protocol:
> > > Replying to close with Frame { header: FrameHeader { is_final:
> > > true,
> > > rsv1: false, rsv2: false, rsv3: false, opcode: Control(Close),
> > > mask:
> > > None }, payload: [] }    
> > > 2023-07-09T08:03:49.981357Z  INFO ThreadId(61)
> > > gst_plugin_webrtc_signalling::server: Received message
> > > Ok(Close(None))
> > > 2023-07-09T08:03:49.981361Z  INFO ThreadId(61)
> > > gst_plugin_webrtc_signalling::server: connection closed: None
> > > this_id=0d76097e-e8fe-45d5-9769-b184bdc30ba8
> > > 2023-07-09T08:03:49.981375Z DEBUG ThreadId(61)
> > > remove_peer{peer_id="0d76097e-e8fe-45d5-9769-b184bdc30ba8"}:
> > > gst_plugin_webrtc_signalling::server: new
> > > 2023-07-09T08:03:49.981401Z DEBUG ThreadId(61)
> > > remove_peer{peer_id="0d76097e-e8fe-45d5-9769-b184bdc30ba8"}:
> > > gst_plugin_webrtc_signalling::server: close time.busy=18.1µs
> > > time.idle=8.46µs
> > > 2023-07-09T08:03:49.981506Z DEBUG ThreadId(56)
> > > remove_peer{peer_id="0d76097e-e8fe-45d5-9769-b184bdc30ba8"}:
> > > gst_plugin_webrtc_signalling::handlers: new
> > > 2023-07-09T08:03:49.981537Z  INFO ThreadId(56)
> > > remove_peer{peer_id="0d76097e-e8fe-45d5-9769-b184bdc30ba8"}:
> > > gst_plugin_webrtc_signalling::handlers: removing peer
> > > peer_id=0d76097e-e8fe-45d5-9769-b184bdc30ba8
> > > 2023-07-09T08:03:49.981551Z DEBUG ThreadId(56)
> > > remove_peer{peer_id="0d76097e-e8fe-45d5-9769-b184bdc30ba8"}:
> > > gst_plugin_webrtc_signalling::handlers: close time.busy=14.5µs
> > > time.idle=33.3µs
> > > 
> > > 
> > > I wonder if you might be willing to comment on this error and
> > > offer a
> > > suggestion on what might be causing it ?
> > > 
> > > Also, if anyone else has been able to follow the steps at :
> > > rswebrtc
> > > gstreamer.freedesktop.org
> > > favicon.png
> > > 
> > > 
> > > 
> > > could you share your experience ?
> > > 
> > > Many thanks !!
> > > 
> > > > On 4 Jul 2023, at 16:43, Mathieu Duponchelle
> > > > <[email protected]> wrote:
> > > > 
> > > > Indeed, those should be absolute URLs (patch welcome), but that
> > > > doesn't
> > > > render the rest of the document invalid does it? In particular,
> > > > the
> > > > pipelines in the usage section are what you are looking for.
> > > > 
> > > > On Tue, 2023-07-04 at 15:07 +0100, GST Developer via gstreamer-
> > > > devel
> > > > wrote:
> > > > > Thanks Mathieu.
> > > > > 
> > > > > I had been working through the documentation, but the links
> > > > > to
> > > > > the
> > > > > signalling server and the API, from this section of the
> > > > > documentation
> > > > > you linked to,
> > > > > at 
> > > > > https://gstreamer.freedesktop.org/documentation/rswebrtc/index.ht
> > > > > m
> > > > > l are returning a 404 error.
> > > > > 
> > > > > rswebrtc
> > > > > (from gst-plugin-webrtc)
> > > > > All-batteries included GStreamer WebRTC producer and
> > > > > consumer,
> > > > > that
> > > > > try their best to do The Right Thing™.
> > > > > It also provides a flexible and all-purposes WebRTC
> > > > > signalling
> > > > > server
> > > > > (gst-webrtc-signalling-server) and a Javascript API
> > > > > (gstwebrtc-
> > > > > api)
> > > > > to produce and consume compatible WebRTC streams from a web
> > > > > browser.
> > > > > also, as referenced later in the documentation.
> > > > > I suspect I will need to build or install this signalling
> > > > > server
> > > > > in
> > > > > order to proceed.
> > > > > 
> > > > > Many thanks again for your help, which is most appreciated :-
> > > > > )
> > > > > 
> > > > > 
> > > > > > On 4 Jul 2023, at 13:46, Mathieu Duponchelle
> > > > > > <[email protected]> wrote:
> > > > > > 
> > > > > > These pipelines are simply incorrect, why don't you just
> > > > > > read
> > > > > > the
> > > > > > documentation that is provided?
> > > > > > 
> > > > > > https://gstreamer.freedesktop.org/documentation/rswebrtc/index.html?gi-language=c#usage
> > > > > > 
> > > > > > 
> > > > > > On Tue, 2023-07-04 at 13:37 +0100, GST Developer via
> > > > > > gstreamer-
> > > > > > devel
> > > > > > wrote:
> > > > > > > Many Thanks Matthieu !!!
> > > > > > > 
> > > > > > > Using ubuntu 22,04LTS, it takes a few quirks, such as not
> > > > > > > installing
> > > > > > > meson with apt (since that gets 0.61.2 which causes an
> > > > > > > error
> > > > > > > later),
> > > > > > > instead using 
> > > > > > > pip3 install --user meson==0.62.0
> > > > > > > and later maybe editing a few files to stop hard coded
> > > > > > > links
> > > > > > > to
> > > > > > > /usr/bin/meson from failing ‘;
> > > > > > > Perhaps a few other tweaks, sorry I have done so many i
> > > > > > > dont
> > > > > > > have
> > > > > > > a
> > > > > > > list
> > > > > > > Then finally adding the path using export GST_PLUGIN_PATH
> > > > > > > 
> > > > > > > Finally, webrtcsink is available in gst-inspect-1.0 !!
> > > > > > > 
> > > > > > > However, I try an example found online: 
> > > > > > > gst-launch-1.0 videotestsrc ! video/x-
> > > > > > > raw,width=1024,height=768,framerate=30/1 ! timeoverlay !
> > > > > > > x264enc
> > > > > > > !
> > > > > > > rtph264pay config-interval=1 pt=96 ! webrtcsink
> > > > > > > and I get an error:
> > > > > > > WARNING: erroneous pipeline: could not link rtph264pay0
> > > > > > > to
> > > > > > > rswebrtcsink0
> > > > > > > 
> > > > > > > realistically I guess really I want something more like
> > > > > > > this
> > > > > > > example
> > > > > > > I found which seems to have some addressing info : 
> > > > > > > gst-launch-1.0 -v videotestsrc ! video/x-
> > > > > > > raw,width=640,height=480
> > > > > > > !
> > > > > > > x264enc ! rtph264pay ! application/x-
> > > > > > > rtp,media=video,encoding-
> > > > > > > name=H264,payload=96 ! rswebrtcsink
> > > > > > > uri=ws://localhost:8080/stream
> > > > > > > but this uses something called “rswebrtcsink”….which
> > > > > > > doesnt
> > > > > > > exist...
> > > > > > > 
> > > > > > > Ultimately what I would like is to run a gstreamer
> > > > > > > pipeline
> > > > > > > which
> > > > > > > ends in a webrtcsink, so I can connect a browser
> > > > > > > directly,
> > > > > > >  ideally
> > > > > > > with the webrtcsink providing the encoding, the
> > > > > > > signalling,
> > > > > > > everything needed to get the picture into the browser.  
> > > > > > > Perhaps
> > > > > > > I
> > > > > > > misunderstand the capabilities, but the various data I
> > > > > > > can
> > > > > > > find
> > > > > > > suggests this is possible.
> > > > > > > 
> > > > > > > Could you advise a simple syntax to send a test audio and
> > > > > > > video
> > > > > > > stream to a browser using webrtcsink ?
> > > > > > > 
> > > > > > > Many, many thanks for your help.
> > > > > > > 
> > > > > > > 
> > > > > > > 
> > > > > > > > On 4 Jul 2023, at 11:50, Mathieu Duponchelle
> > > > > > > > <[email protected]> wrote:
> > > > > > > > 
> > > > > > > > Compiling the webrtc crate after compiling gstreamer
> > > > > > > > from
> > > > > > > > source
> > > > > > > > and
> > > > > > > > entering the uninstalled environment should work, if
> > > > > > > > you
> > > > > > > > encounter
> > > > > > > > issues please file them on gitlab with steps to
> > > > > > > > reproduce.
> > > > > > > > 
> > > > > > > > As for installing GStreamer with webrtcsink included, I
> > > > > > > > don't
> > > > > > > > know
> > > > > > > > if
> > > > > > > > distributions actually package gst-plugins-rs to be
> > > > > > > > honest
> > > > > > > > :)
> > > > > > > > 
> > > > > > > > On Mon, 2023-07-03 at 18:25 +0100, GST Developer via
> > > > > > > > gstreamer-
> > > > > > > > devel
> > > > > > > > wrote:
> > > > > > > > > Hi Folks.
> > > > > > > > > 
> > > > > > > > > I am attempting to use WebRTCSink on Ubuntu 22.04
> > > > > > > > > LTS,
> > > > > > > > > but
> > > > > > > > > despite 2
> > > > > > > > > days trial and error I have yet to find any way to
> > > > > > > > > install
> > > > > > > > > it.
> > > > > > > > > 
> > > > > > > > > I have tried following guidance for installation of
> > > > > > > > > all
> > > > > > > > > the
> > > > > > > > > plug
> > > > > > > > > in
> > > > > > > > > groups via apt install, and also compiling gstreamer
> > > > > > > > > from
> > > > > > > > > source,
> > > > > > > > > but
> > > > > > > > > whilst I can use webrtcbin, none of my attempts have
> > > > > > > > > yet
> > > > > > > > > found
> > > > > > > > > webrtcsink to be available.
> > > > > > > > > 
> > > > > > > > > I also went down the root of trying to compile
> > > > > > > > > from 
> > > > > > > > > https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/tree/m
> > > > > > > > > ain/net/webrtc
> > > > > > > > > but this tends to get hung up on now having the
> > > > > > > > > correct
> > > > > > > > > version
> > > > > > > > > of
> > > > > > > > > various gstreamer or other components.
> > > > > > > > > 
> > > > > > > > > Is there a simple way to install GStreamer with
> > > > > > > > > WebRTCSink
> > > > > > > > > included ?
> > > > > > > > > 
> > > > > > > > > Many thanks for any clues !
> > > > > > > 
> > > > > 
> > > 
>