Re: How can I install WebRTCSink on Ubuntu 22.04

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:
https://gstreamer.freedesktop.org/documentation/rswebrtc/index.html?gi-language=c#usage
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\"],\"meta\":{\"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 :
https://gstreamer.freedesktop.org/documentation/rswebrtc/index.html?gi-language=c#usage
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.htm
>> 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 !
>>>> 
>>