this: https://github.com/arturoc/ofxGstRTP can do video & audio in sync over rtp, do nat transversal using stun… which are the protocols that webrtc uses internally but it won’t be directly compatible with webrtc so if you need to communicate with other webrtc client it won’t work but if you just need to send video/audio between 2 OF apps it’ll do
interesting, thanks Arturo! We’re actually already using ofxGstreamer to playback RTSP streams…
the requirement for this one is to actually integrate with a server that’s managing/hosting/archiving (and potentially providing the signaling needed to setup a webRTC connection.) I am definitely considering a “roll your own” approach using gstreamer to decode WebRTC (if it’s even possible), but this seems it might be a massive rabbit hole.
Just a heads-up that Gstreamer 1.14+ now has the webrtc component. I did some work on ofxGStreamer to get it compiling w/ macOS via the brew tap installed GStreamer, or via manual builds of gstreamer components.
Hi ! Great topic !
For an artistic project, I need to make an OF app that can send and receive webRTC audio/video to/from other devices (using a web app)…
I’m trying your solution but nothing works when I test examples. Both, Arturo and autr “ofxGStreamer”.
Are there out to date ?
Or the problem come from my setup (Mac OS X 10.15.7 / Xcode 11.6) ?
Or maybe I do not correctly install or make something ?
With Arturo addon and ofxGstRTP examples, the error is :
‘file_wrapper.h’ file not found
In audio_processing_impl.cc
With autr addon and his ofxGStreamer examples, the error is :
d: file not found: @rpath/lib/libgstnet-1.0.0.dylib for architecture x86_64
clang: error: linker command failed with exit code 1 (use -v to see invocation)