Hi, been looking at Membrane for a few days now and it seems pretty great. I don’t have any prior experience in this area so the guides have been very helpful.
I want to relay some web-rtc audio (in both directions) from a browser into a server (e.g. a membrane server) and then into pipewire nodes running on the same server. My default approach is to use Membrane for the signalling and then a Membrane pipeline to get the packets into raw audio frames, which I will then push into pipewire, through either a couple of pipewire ‘Unix FIFO’ modules, or a ‘Simple Protocol’ module. These modules are then linked to the ‘real’ pipewire nodes. Does this sound like a sane approach?
Follow up question:
My other thought when I looked at the available pipewire modules, was to use the pipewire RTP modules and then just advertise their endpoints during signalling (in Membrane). But I don’t know if this is feasible, as even ex_webrtc seems to want own the rtp endpoints after setup?
Thx!




















