The signaling server for the Unity asset WebRtcVideoChat is now open source under the BCD license with its original typescript source code. You can find it on github under its new name awrtc_signaling.
The current master branch is already prepared for the next version V0.984 and contains several changes such as updated dependencies, stability improvements and support for heatbeat messages which will allow the client side to detect disconnects quicker. It is also backwards compatible to all earlier versions. An update to the new version is highly recommended.
Please feel free to contribute and report issues & suggestions to the github issue tracker.
The pure java script/typescript API and Unity WebGL plugin is now open source and freely available on github: https://github.com/because-why-not/awrtc_browser. It also contains a browser version of the CallApp which can connect to the Unity version out of the box. You can give it a try here.
The first test builds for the unity version of WebRTC Video Chat V0.984 are now available in the download area. The current version is not yet stable but will be updated on a weekly basis until it is ready for release. Please report any errors you find via e-mail or use the github issue tracker. See changelog to learn more about the next update.
A new patch is now pending review from Unity and should be released in a few days. The original version can trigger an exception if UnityCallFactory.GetVideoDevices() is called in WebGL if the project was build with Unity 2018.2 (and possibly other 2018 versions). The error is also triggered by the CallApp example.
You will get the following error message:
TypeError: Module.stringToUTF8 is not a function,TypeError: Module.stringToUTF8 is not a function
If you are using Unity 2018 with WebGL and don’t want to wait until the patch is released: You can open the file awrtc_unity.jslib and replace “Module.stringToUTF8” with “stringToUTF8”.
Update: The Asset has been updated and you will get the patched version automatically the next time you download & import the asset.
The first release candidates for V0.983 are available in the Download Area. Details for all changes and new features can be found at: Changelog .
V0.983 also adds a the first purchasable extra that won’t be part of the asset version: The latest development version for UWP/HoloLens. Please inquire by email for more detail.
The next version is now available in the Unity Asset Store. Here is the change log:
– Removed obsolete wrapper methods. Recent changes in WebRTC broke most of them
– Update to WebRTC 64 + switching to newer compiler versions (XCode 9, VS 2017, newer NDK)
– Cleanup of log calls
– ICall.SetMute / IsMute added to silence the local microphone on all platforms
– Added mute button in the CallApp
– added bool Send(string, bool, ConnectionId) to ICall and IBasicNetwork to support high throughput data transfer
– Rework of the frame delivery system. RawFrame is obsolete and will be replaced with IFrame (will be completed in V0.983)
– for testing: some platforms can deliver frames in i420p which reduces CPU load (see SimpleCall example)
– Added first changes to support direct access to some WebRTC functionality in the future
– Android supports native video now (camera 1 or camera 2 API depending on the device). The old camera system is deactivated by default.
– added AndroidHelper.IsFrontFacing / IsBackFacing. For all other platforms WebcamTexture.devices can be used for this.
– Android native plugin is bundled into an .aar file now. Unity will automatically include necessary changes for the AndroidManifest.xml
– Android comes with a java plugin now to support platform specific features in the future
– Android IL2CPP support has been added
– iOS plenty of bugfixes and performance improvements
– iOS comes with an Objective C plugin now to support platform specific features in the future
– WebGL version updated to use newer API calls
– WebGL fixed several bugs related to various image resolutions / rotated images received from mobile phones
– Bugfix: using invalid device names could cause crashes on some platforms
– Bugfix: Video shouldn’t stop anymore if the scene is changed on Android
– Bugfix: Fixed problems with the loud speaker mode for iOS device
– Bugfix: multi-threading issues / stalling on shutdown or startup
A new download area has been added to the webpage. It can be used to download the latest hotfixes, recent builds and older releases of WebRTC Video Chat. It also contains the latest build for the version 0.982!
You can log in using your invoice ID from the Unity Asset Store purchase of WebRTC Video Chat.
Here is the changelog:
- Documentation updated. Please visit https://www.because-why-not.com/webrtc to find out more
- Tutorials for signaling server & stun/turn server setup added: https://www.because-why-not.com/webrtc/tutorials-server-side/
- ICall received new methods to send String & byte to one or multiple users using UDP or TCP style channels
- Android video has been further optimized (full native camera access will come soon)
- CallApp has a new configuration button allowing users to change resolution and other settings
- CallApp shows now additional information if the image is clicked once
- CallApp will now save the last settings (based on the name if its GameObject to still allow multiple instances in one application)
- extra folder added. These are additional features requested by users and can be used at your own risk.
- extra VideoInput added. It can be used to stream custom images or from a Unity camera
- mobile devices won’t switch off screen any more of CallApp is used
- extra OneToMany added. It is an example how to use the IMediaInterface to stream to multiple recives.
- extra VideoConference added. An example how to create conference calls using ICall interface.
- example folder added. It contains several minimal examples to help new users.
See more here:https://www.because-why-not.com/webrtc/examples/
- mac native libraries uses proper bundle format now
- iOS workaround added. If a phone call ends Unity will turn off sound for all AudioSources. IosHelper.UnitySetAudioSessionActive can be called after the call ended to switch
the Audio back into the correct mode. All AudioSources need to be restarted after it.
- signaling server contains now a webserver to make testing and use of https://letsencrypt.org/ easier
Note that new features mentioned as “extra” might not work on all platforms and will only receive limited support for now.
The focus of the next few updates will be on updating the code to keep up with recent changes in WebRTC, improving the performance and general bugfixing, especially for the mobile versions.
If you find any problems please send an email to firstname.lastname@example.org with a description of the problem + an example project that can reproduce the problem. Ideally, use the new minimal examples as base to reproduce any errors.
Test servers + email were partly inaccessible in the past few hours as the hosting company decided to delete all DNS entries while moving from one plan to another.
Everything should be back to normal now. No further changes are planned.
The servers will be updated within the next 24 hours. That means webserver as well as signaling, stun and turn might be unavailable for a few hours.
Version 0.98 of WebRTC Video Chat is now awaiting Unity’s approval and will be released soon.
Here is the change log:
- 0.98 – iOS support (arm + arm64). Please check the readme.txt for how to build it.
- Removed log messages / errors appearing with Unity 2017.2
- C++ side was rewritten entirely replacing all callbacks with a polling based system
- Using the debugger in Unity should work better now
- Video Frames will now be automatically dropped if the Update method isn’t called quickly enough (e.g. during a FPS drop)
- IMediaNetwork is now supported and can be created using UnityCallFactory.Instance.CreateMediaNetwork
It works similar to WebRTC Network’s IBasicNetwork interface but adds audio and video to make broadcasting to
audio / video to multiple users easier
- UnityCallFactory.SetLoudspeakerStatus / GetLoudspeakerStatus added. Mobile devices treat WebRTC calls like regular phone calls
requring the speakers to be manually turned on.
- WebGL version was adapted to recent changes in Chrome. The CallEnded event should now be triggered during disconnects due to network failure
- CallApp was improved to better support mobile platforms + a new button was added to change SetLoudspeakerStatus
- Numerous smaller bugfixes and changes based on user reports received in the past few months
There were also a lot of internal changes that make the way free for some new features that will be released within the next few weeks as part of the 0.98x versions!
Please note that the servers will change with the update and the old servers will be shutdown within the next few weeks! There might also be some downtime for this website within the next few days!