H264 to WebRTC

[Simon Brown]

Hi,

I'm very new to Gstreamer, so please bear with me.

I have an h264 video stream and an aac audio stream in an Mpeg2 Transport stream.  I would like to send this to a webpage using WebRTC.  Everything I read suggests it's possible, but all the examples I've found encode the video first.  Is there a simple pipeline that will let me take the mpeg2ts and output webrtc?

This is all running on an embedded system so I don't have the CPU power to re-encode the video.

Regards,

Simon

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[Pascal Jacquemart]

Hi Simon,

I think it depends on the H264 profile but it should be possible.
You should be able to use something like

filesrc location=video.ts ! tsdemux ! rtph264pay !
application/x-rtp,media=video,encoding-name=H264,payload=96 ! webrtcbin

Cheers,        Pascal

On Wed, 11 Sep 2019 17:24:11 +0100
Simon Brown <[hidden email]> wrote:

> Hi,
> I'm very new to Gstreamer, so please bear with me.
> I have an h264 video stream and an aac audio stream in an Mpeg2
> Transport stream.  I would like to send this to a webpage using
> WebRTC.  Everything I read suggests it's possible, but all the
> examples I've found encode the video first.  Is there a simple
> pipeline that will let me take the mpeg2ts and output webrtc?
>
> This is all running on an embedded system so I don't have the CPU
> power to re-encode the video.
>
> Regards,
> Simon

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[Simon Brown]

On Thu, 12 Sep 2019 at 02:19, Pascal Jacquemart <[hidden email]> wrote:

Hi Simon,

I think it depends on the H264 profile but it should be possible.
You should be able to use something like

filesrc location=video.ts ! tsdemux ! rtph264pay !
application/x-rtp,media=video,encoding-name=H264,payload=96 ! webrtcbin

Cheers,        Pascal

Thanks Pascal,

I'm trying this now on an Ubuntu system, and two things spring to light:

1) webrtcbin isn't available - it looks like this is available from source for gstreamer bad plugins, but not from install of the bad plugins?  

2) It looks like it's only doing the video - is there a secondary path that I can send the audio on?

Cheers,

Simon

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[Pascal Jacquemart-3]

Yes there are few things to be aware of:

- gstwebrtcbin was introduced in 1.14. I am not surprise if you still have an older version.

- Yes you can add an audio path but we will probably need to re-encode the audio. See https://github.com/centricular/gstwebrtc-demos

- You cannot use a static pipeline to use the webrtcbin because you need to plug a custom signalling. So you have to write code and you cannot use gst-launch directly.

Cheers,       Pascal

On 13 Sep 2019, at 21:41, Simon Brown <[hidden email]> wrote:

On Thu, 12 Sep 2019 at 02:19, Pascal Jacquemart <[hidden email]> wrote:

Hi Simon,

I think it depends on the H264 profile but it should be possible.
You should be able to use something like

filesrc location=video.ts ! tsdemux ! rtph264pay !
application/x-rtp,media=video,encoding-name=H264,payload=96 ! webrtcbin

Cheers,        Pascal

Thanks Pascal,

I'm trying this now on an Ubuntu system, and two things spring to light:

1) webrtcbin isn't available - it looks like this is available from source for gstreamer bad plugins, but not from install of the bad plugins?  

2) It looks like it's only doing the video - is there a secondary path that I can send the audio on?

Cheers,

Simon

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[Ashok]

I'm new to gstreamer and gstreamer webrtcbin. I ran the following command as
you posted.

gst-launch-1.0 filesrc location=video.ts ! tsdemux ! rtph264pay !
application/x-rtp,media=video,encoding-name=H264,payload=96 ! webrtcbin

Looks like it is running

Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock

After that how I stream the video to the web page. Can you please help me?



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[Christopher Lee]

Based on your pipeline description, I'd recommend taking a look at the "sendonly" example from centricular's demo applications (https://github.com/centricular/gstwebrtc-demos).

On Mon, Apr 20, 2020 at 7:19 AM Ashok <[hidden email]> wrote:

I'm new to gstreamer and gstreamer webrtcbin. I ran the following command as
you posted.

gst-launch-1.0 filesrc location=video.ts ! tsdemux ! rtph264pay !
application/x-rtp,media=video,encoding-name=H264,payload=96 ! webrtcbin

Looks like it is running

Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstSystemClock

After that how I stream the video to the web page. Can you please help me?



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