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README: link to blog post, document multiparty example

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Also add TODO stubs for MCU and SFU
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Nirbheek Chauhan 2018-02-17 08:10:59 +05:30
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webrtc/README.md
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@ -1,7 +1,9 @@
## GStreamer WebRTC demos # GStreamer WebRTC demos
All demos use the same signalling server in the `signalling/` directory All demos use the same signalling server in the `signalling/` directory
## Build
The GStreamer WebRTC implementation has now been merged upstream, so all The GStreamer WebRTC implementation has now been merged upstream, so all
you need is the latest GStreamer git master, as of 2 February 2018 or later: you need is the latest GStreamer git master, as of 2 February 2018 or later:
@ -18,11 +20,23 @@ Or with Meson gst-build:
https://cgit.freedesktop.org/gstreamer/gst-build/ https://cgit.freedesktop.org/gstreamer/gst-build/
You may need to install the following packages using your package manager:
json-glib, libsoup, libnice, libnice-gstreamer1 (the gstreamer plugin for libnice)
## Documentation
Currently, the best way to understand the API is to read the examples. This post breaking down the API should help with that:
http://blog.nirbheek.in/2018/02/gstreamer-webrtc.html
## Examples
### sendrecv: Send and receive audio and video ### sendrecv: Send and receive audio and video
* Serve the `js/` directory on the root of your website, or open https://webrtc.nirbheek.in * Serve the `js/` directory on the root of your website, or open https://webrtc.nirbheek.in
- The JS code assumes the signalling server is on port 8443 of the same server serving the HTML - The JS code assumes the signalling server is on port 8443 of the same server serving the HTML
* Build and run the sources in the `gst/` directory on your machine * Build the sources in the `gst/` directory on your machine
```console ```console
$ gcc webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o webrtc-sendrecv $ gcc webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o webrtc-sendrecv
@ -31,3 +45,29 @@ $ gcc webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstrea
* Open the website in a browser and ensure that the status is "Registered with server, waiting for call", and note the `id` too. * Open the website in a browser and ensure that the status is "Registered with server, waiting for call", and note the `id` too.
* Run `webrtc-sendrecv --peer-id=ID` with the `id` from the browser. You will see state changes and an SDP exchange. * Run `webrtc-sendrecv --peer-id=ID` with the `id` from the browser. You will see state changes and an SDP exchange.
* You will see a bouncing ball + hear red noise in the browser, and your browser's webcam + mic in the gst app * You will see a bouncing ball + hear red noise in the browser, and your browser's webcam + mic in the gst app
TODO: Port to Python and Rust.
### multiparty-sendrecv: Multiparty audio conference with N peers
* Build the sources in the `gst/` directory on your machine
```console
$ gcc mp-webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o mp-webrtc-sendrecv
```
* Run `mp-webrtc-sendrecv --room-id=ID` with `ID` as a room name. The peer will connect to the signalling server and setup a conference room.
* Run this as many times as you like, each will spawn a peer that sends red noise and outputs the red noise it receives from other peers.
- To change what a peer sends, find the `audiotestsrc` element in the source and change the `wave` property.
- You can, of course, also replace `audiotestsrc` itself with `autoaudiosrc` (any platform) or `pulsesink` (on linux).
* TODO: implement JS to do the same, derived from the JS for the `sendrecv` example.
### TODO: Selective Forwarding Unit (SFU) example
* Server routes media between peers
* Participant sends 1 stream, receives n-1 streams
### TODO: Multipoint Control Unit (MCU) example
* Server mixes media from all participants
* Participant sends 1 stream, receives 1 stream