-
Notifications
You must be signed in to change notification settings - Fork 27
Description
We have successfully integrated this library in our application, and currently use it to bridge PSTN phone calls to our video service. For this to work, we rely on the nodejs process being able to send audio to remote participants.
However, we stumbled upon an issue when joining p2p calls with more than one remote participant (SFU calls using Mediasoup works fine). Whenever the 2nd remote participant joins, audio is only sent from the Nodejs process to the last participant.
Looking at existing issues, seems like node-webrtc#745 is quite similar, although in our implementation, we only create a single local audio track.
Diving into the code, it seems like the RTCAudioSource implementation is a bit strange. Internally, the audio source only keeps track of one single _sink. Wouldnt this break if there are more than one sink, eg multiple RTP senders?
Im hoping to confirm this with a local fix and build, but thought Id share in advance.