Hello everyone,
I'm working on a audio/video/chatting app, and I've got some issues with the way AVAudioSession is handled by WebRTC.
I am specifically overriding the audio port so that audio goes through loud speakers using this: "AVAudioSession.sharedInstance().overrideOutputAudioPort(.speaker)", but after peer connection gets established (Meaning it's ICE state changes to "Connected") somehow WebRTC changes audio route to go through internal speakers. I want audio to not change route just because I get connected to another peer.
Maybe I'm missing some obvious thing in the API? Can anybody help me?