Playing RTMP Live Streaming into Conference Bridge

Hi All,

I am new to Asterisk.

I am building an application that will call (or allow people to call) and listen to a live stream.

I searched the forums and the Internet for solutions and the most common I found was to use the “Music on Hold” and pipe a stream into that using a custom class. From what I could tell that is not good as it creates an instance of the class for every person on “hold” which could be a lot of users and CPU load.

My thought is the create a Conference Bridge (and have people that joined muted) and figure out how to programmatically “play” the live stream into the conference bridge as the administrator, as if the admin joined on the phone and was speaking.

Maybe when the admin hangs up (eg the live stream is over) it closes the bridge or something.

I am running the latest Asterisk and in the last week got familiar with Dialplans, etc.

I am just missing the basic concept of how to get the stream audio (from ffmpeg?) into the bridge?

I wanted to reach out to others who might have some insight or point me in the right direction.

Thanks!

-Rob