What's the most straightforward way of receiving and sending a real time audio between a VoIP calling service and a node app that's only connected to the internet? It needs to be able to 'dial' a call, and send/ receive audio.
Right now, the architecture I've figured out is roughly to use Twilio's Elastic SIP trunking, then setup a SIP server like Asterisk that proxies RTP to WebRTC and connect that to Twilio, and then use something like JsSIP (although I'm not even sure if it allows getting an audio stream in a node environment) as a SIP over WebRTC client, but this is extremely complicated to setup, and just feels like overkill.
Is there an easier way/ service that provides this functionality, or at least an existing guide on how to do this?
You have following options
1) WebRTC
2) EAGI(audio to script file#3, one way).
3) asterisk to JACK
4) create your own c/c++ handler and do in whatever format you want.