We are witnessing a large adoption of web based audio/video communication platforms that run in a browser and can be easily integrated with the web environment. Most of these solutions are in the context of videoconferencing, but the recent limitation to people mobility encouraged the extension of such platforms also to the context of networked music performance. This master thesis focuses on showing an alternative approach to peer-to-peer high quality and low latency audio streaming by exploring an unconventional solution based on WebRTC's DataChannels, instead of WebRTC's MediaStreams. Uncompressed audio is extracted from WebRTC's MediaStream, processed by Web Audio API's AudioWorklets and sent on WebRTC's DataChannels in order to have more control on the overall transmission protocol and to avoid possible communication delays introduced by compression, filtering and other audio processing MediaStreams may perform.