Sending to the browser a RTP Stream source about webrtc HOT 4 CLOSED

Welcome @rosimildo.
A quick overview of what we have right now. If you look at the gstreamer-send example you'll see the following line:

webrtc.RegisterDefaultCodecs()

This line initializes some default 'codecs' which are shipped with the library.
Note that when referring to a 'codec' we mean a collection of:

• Some meta data that defines the codec. This is used in WebRTC to find matching capabilities between the peers.
• A Payloader used to split the stream of media data into packets for sending over the network.

The actual conversion of video is (at least currently) out of scope of our efforts.
In addition, you can define your own codec. One way to do this is used in the save-to-disk example:

webrtc.RegisterCodec(webrtc.NewRTCRtpOpusCodec(webrtc.DefaultPayloadTypeOpus, 48000, 2))

You can even go further and define a completely new codec. You can use the contents of NewRTCRtpOpusCodec as an example.
We also support multiple peer connections with different codecs. This can be done by creating your own MediaEngine and registering it with the peer connection using SetMediaEngine.

A fully generic way to ingest media would have to identify the codec and supply the required details to the library as described above. I currently don't know of any such efforts.

Hopefully this provides some insight. Let me know if you have any additional questions.

from webrtc.

Thanks Backkem for your quick answer!

For now, I am browsing the project to get familiar with, and hopefully soon to be able to play with the examples..... thanks!

from webrtc.

Great, If you have any questions while browsing you can always find us in the #pion channel of the Gopher slack.

from webrtc.

Hey @rosimildo, thanks for checking out pion-WebRTC!

We have examples of GStreamer generating RTP traffic (and consuming it) I am happy to add more examples though (I think this covers what your looking for though)

Feel free to reopen this if you have a specific example that would help developers learning this library! And always happy to chat in Slack.

from webrtc.