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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
## Build
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:
@ -18,11 +20,23 @@ Or with Meson 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
* 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
* Build and run the sources in the `gst/` directory on your machine
* Build the sources in the `gst/` directory on your machine
```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
@ -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.
* 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
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