I want to develop a application that can do real time call transcribe in a call center environment, you must provide your description of solution in detail on how you can push the audio stream from a traditional call center architecture (IVR/ACD) or in open source environment like Asterisk.
how to classify which audio stream belongs to which agent.
Currently we already developed a demo code which can receive audio stream from Twilio through ngrok but we understand that in real call center environment it is more complicated which involves multiple agents working at the same time.
we study the IBM solution in voice gateway and understand that SBC (session border contoller) probably needed to fork the call and audio stream to voice gateway and further to Rest Server.
we need an expert like you to work with us to achieve that.
If you are an expert and have working experience in this area, Please submit your proposal with detail for us to considered. Looking forward to hearing from you.
11 freelancers are bidding on average $2388 for this job
We are a team of professional developer's having 3+ experience. We also work on similar Idea. Can help you to achieve within timeframe with quality of work.