webrtc example -- wss://webrtc.nirbheek.in:8443 not available -- send video only example

[lionel mazeyrat]

I try to test the webrtc example.

wss://webrtc.nirbheek.in:8443 seems not to be still available....

I've tried to replace it with a javalin wss and to bypass stun server

Is it possible to configure the webrtcbin to send video only ? no cideo/audio stream to receive.

I try to send rstp stream with webrtc...and to integrate a ws javalin server for the signaling protocol with a JS client.

[Neil C Smith]

On Fri, 13 Aug 2021 at 13:14, lionel mazeyrat <lionel....@gmail.com> wrote:
> I try to test the webrtc example.
> wss://webrtc.nirbheek.in:8443 seems not to be still available....

Looks up at the moment. The example is a port of the upstream
GStreamer one using that server, which is run by one of the GStreamer
devs I think. It's a little problematic, and the webpage doesn't seem
to request audio permissions in the browser correctly so that has to
be done manually. If there's a better example server to use we could
consider changing.

> Is it possible to configure the webrtcbin to send video only ? no cideo/audio stream to receive.

Leaving out the audio parts of the pipeline should work. It's also
the way to work around the audio permissions issue!

Best wishes,

Neil


--
Neil C Smith
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[lionel mazeyrat]

the https web page is ok but I can't reach the wss://webrtc.nirbheek.in:8443 from java code or a test tool : postman.

For the webrtcbin, I had ever delete the audio part.

But I don't known if it could be annoying if there is no stream received. 

I use the html part of:

https://gitlab.freedesktop.org/gstreamer/gst-examples/-/blob/master/webrtc/sendonly/webrtc-unidirectional-h264.c

but the video doesn't play despide all the steps seems to be good.....

In the C example, there is somme additionnal settings, but I can't figure out how to set it in java with the WebRTCBin

receiver_entry->webrtcbin =

      gst_bin_get_by_name (GST_BIN (receiver_entry->pipeline), "webrtcbin");

  g_assert (receiver_entry->webrtcbin != NULL);

  g_signal_emit_by_name (receiver_entry->webrtcbin, "get-transceivers",

      &transceivers);

  g_assert (transceivers != NULL && transceivers->len > 1);

  trans = g_array_index (transceivers, GstWebRTCRTPTransceiver *, 0);

  g_object_set (trans, "direction",

      GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, NULL);

  if (video_priority) {

    GstWebRTCPriorityType priority;

    priority = _priority_from_string (video_priority);

    if (priority) {

      GstWebRTCRTPSender *sender;

      g_object_get (trans, "sender", &sender, NULL);

      gst_webrtc_rtp_sender_set_priority (sender, priority);

      g_object_unref (sender);

    }

  }

[lionel mazeyrat]

If there is a mean to debug what's wrong ???

stream object seem to be received in the JS client

[lionel mazeyrat]

For sending video only without audio I use the following pipeline

But I'm not sure of this pipeline because nothing is displayed in the client...

Do you think it is ok ?

private static final String PIPELINE_DESCRIPTION
= "videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay"
+ " ! queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 !"
+ " webrtcbin name=webrtcbin bundle-policy=max-bundle";

Le vendredi 13 août 2021 à 15:07:51 UTC+2, neil a écrit :

[Neil C Smith]

On Fri, 13 Aug 2021 at 22:05, lionel mazeyrat <lionel....@gmail.com> wrote:
>
> For sending video only without audio I use the following pipeline
> But I'm not sure of this pipeline because nothing is displayed in the client...
>
> Do you think it is ok ?
> private static final String PIPELINE_DESCRIPTION
> = "videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay"
> + " ! queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 !"
> + " webrtcbin name=webrtcbin bundle-policy=max-bundle";

Off hand, not sure. When testing this with the example, I think I
just commented out lines
https://github.com/gstreamer-java/gst1-java-examples/blob/master/WebRTCSendRecv/src/main/java/org/freedesktop/gstreamer/examples/WebRTCSendRecv.java#L59-L60

However, I would recommend using the upstream GStreamer mailing list
for questions that are not specific to the Java integration.