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Please visit https://because-why-not.com for
up-to-date setup information, guides and examples!
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Introduction:
WebRTC Video Chat is a unity plugin which allows you to stream audio, video and send text messages between
two programs. You can use it to create a live video and audio stream similar to features of Skype, Google Hangout
or Teamspeak and integrate it directly into your own unity project.
The assets comes with a fully functional open source example to demonstrate how to create a video chat. If this
is all you need you can simply change the UI to your liking and use it without any programming.
How does it work?
The plugin automatically handles all audio/video and network functionality for you. All you need is to
create a Call object and connect it to the other side and using a shared password. An event handler will
then return events when a user connects, sends a message, a new video frame is received and so on.
The video frames will be returned as a raw image. You can then copy it into a texture (see example) or
use it for other libraries e.g for facial recognition or applying filters.
So far Unity Editor in Windows + Standalone Windows x86 and x64 is supported. Browsers and mobile platforms
will be supported in the near future. Conference calls with multiple users will be added
in an update as well. Note that the network functionality requires a server to work. I will provide
a development server for free but you will need to set up your own server for production.
Checkout http://because-why-not.com/webrtc/webrtc-network/ for more information, documentation, API
and test a free sample application.
Note: This asset uses shared dependencies with WebRtcNetwork. Please read the description of WebRtcNetwork
as well.
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Setup:
Make sure you tick "Run in background" in your player settings. + add the example scenes to your build settings.
Make sure your platform is set to PC / Mac / Linux Standalone.
for Android:
Don't forget to configure your Assets\Plugins\Android\AndroidManifest.xml file. You can find an
example at WebRtcNetwork\Plugins\android\AndroidManifest.xml.
You need at least following permission:
<uses-permission android:name="android.permission.INTERNET" />
<uses-permission android:name="android.permission.ACCESS_NETWORK_STATE" />
<uses-permission android:name="android.permission.CHANGE_NETWORK_STATE" />
For Video:
<uses-permission android:name="android.permission.CAMERA" />
<uses-feature android:name="android.hardware.camera" android:required="false" />
<uses-feature android:name="android.hardware.camera.autofocus" android:required="false" />
<uses-feature android:name="android.hardware.camera.front" android:required="false" />
For Audio:
<uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
<uses-permission android:name="android.permission.RECORD_AUDIO" />
Also have a look at AndroidHelper.cs and UnityCallFactory.Instance.SetLoudspeakerStatus. WebRTC treats the audio connection as a phone call on Android thus
the speaker has to be turned on to have a proper volume.
iOS:
Unity iOS Build Settings:
- Set minimum iOS Version to 9.3 (lower might work but isn't tested)
- Backend should be IL2CPP
- Make sure to set Camera and Microphone Usage Description. If those fields are empty newer iOS versions will crash
Building the xcode project:
- The script IosPostBuild.cs will automatically change flags in your build project while Unity is creating the XCode project
It will make sure the native part of the library is added as an embedded framework.
- (Only needed for versions below Unity 2017.2)You still need to set a flag by yourself in xcode:
"Project settings -> Build phases -> Copy Frameworks -> set the flag Code Sign On Copy
- Depending on your personal setup you might then still need to setup signing / other unity and ios specific build settings. Please check
with Unity and iOS documentation first before requesting support.
Also have a look at UnityCallFactory.Instance.SetLoudspeakerStatus to turn on / off the phones speaker.
Example CallApp:
The callscene contains two instances of the CallApp allowing you to test the library within a single application or
run the same program on two different pc's and connect. It supports streaming audio, video and sending text messges
to the other side. Simply enter a shared text or password on both sides and press the join button. The app will
connect both apps and start streaming audio and video (if available).
Important folders, files and classes:
This library shares dependencies with WebRtcNetwork. Please see the WebRtcNetwork/readme.txt for everything about
WebRtcNetwork.
