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113 lines
4.7 KiB
Markdown
113 lines
4.7 KiB
Markdown
# GStreamer WebRTC demos
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All demos use the same signalling server in the `signalling/` directory
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## Downloading GStreamer
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The GStreamer WebRTC implementation has now been merged upstream, and is in the GStreamer 1.14 release. Binaries can be found here:
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https://gstreamer.freedesktop.org/download/
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## Building GStreamer from source
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If you don't want to use the binaries provided by GStreamer or on your Linux distro, you can build GStreamer from source.
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The easiest way to build the webrtc plugin and all the plugins it needs, is to [use Cerbero](https://gstreamer.freedesktop.org/documentation/installing/building-from-source-using-cerbero.html). These instructions should work out of the box for all platforms, including cross-compiling for iOS and Android.
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One thing to note is that it's written in Python 2, so you may need to replace all instances of `./cerbero-uninstalled` (or `cerbero`) with `python2 cerbero-uninstalled` or whatever Python 2 is called on your platform.
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## Building GStreamer manually from source
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For hacking on the webrtc plugin, you may want to build manually using the git repositories:
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- http://cgit.freedesktop.org/gstreamer/gstreamer
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- http://cgit.freedesktop.org/gstreamer/gst-plugins-base
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- http://cgit.freedesktop.org/gstreamer/gst-plugins-good
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- http://cgit.freedesktop.org/gstreamer/gst-plugins-bad
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- http://cgit.freedesktop.org/libnice/libnice
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You can build these with either Autotools gst-uninstalled:
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https://arunraghavan.net/2014/07/quick-start-guide-to-gst-uninstalled-1-x/
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Or with Meson gst-build:
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https://cgit.freedesktop.org/gstreamer/gst-build/
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You may need to install the following packages using your package manager:
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json-glib, libsoup, libnice, libnice-gstreamer1 (the gstreamer plugin for libnice)
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## Filing bugs
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Please only file bugs about the demos here. Bugs about GStreamer's WebRTC implementation should be filed on the [GStreamer bugzilla](https://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-plugins-bad).
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You can also find us on IRC by joining #gstreamer @ FreeNode.
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## Documentation
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Currently, the best way to understand the API is to read the examples. This post breaking down the API should help with that:
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http://blog.nirbheek.in/2018/02/gstreamer-webrtc.html
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## Examples
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### sendrecv: Send and receive audio and video
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* Serve the `js/` directory on the root of your website, or open https://webrtc.nirbheek.in
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- The JS code assumes the signalling server is on port 8443 of the same server serving the HTML
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* Open the website in a browser and ensure that the status is "Registered with server, waiting for call", and note the `id` too.
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#### Running the C version
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* Build the sources in the `gst/` directory on your machine
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```console
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$ gcc webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o webrtc-sendrecv
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```
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* Run `webrtc-sendrecv --peer-id=ID` with the `id` from the browser. You will see state changes and an SDP exchange.
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#### Running the Python version
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* python3 -m pip install --user websockets
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* run `python3 sendrecv/gst/webrtc-sendrecv.py ID` with the `id` from the browser. You will see state changes and an SDP exchange.
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> The python version currently requires the master branches from `gst-plugins-bad` and `gst-plugins-base`.
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<!---
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TODO: replace the note above when 1.16 is released
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-->
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With all versions, you will see a bouncing ball + hear red noise in the browser, and your browser's webcam + mic in the gst app.
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You can pass a --server argument to all versions, for example `--server=wss://127.0.0.1:8443`.
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<!---
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TODO: Port to Rust.
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-->
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### multiparty-sendrecv: Multiparty audio conference with N peers
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* Build the sources in the `gst/` directory on your machine
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```console
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$ 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
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```
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* 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.
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* 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.
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- To change what a peer sends, find the `audiotestsrc` element in the source and change the `wave` property.
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- You can, of course, also replace `audiotestsrc` itself with `autoaudiosrc` (any platform) or `pulsesink` (on linux).
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* TODO: implement JS to do the same, derived from the JS for the `sendrecv` example.
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### TODO: Selective Forwarding Unit (SFU) example
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* Server routes media between peers
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* Participant sends 1 stream, receives n-1 streams
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### TODO: Multipoint Control Unit (MCU) example
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* Server mixes media from all participants
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* Participant sends 1 stream, receives 1 stream
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