From what I can tell from the Wiki page at https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MP3Player, if someone dials in and starts playing a stream, mp3player will load up the URL and inject it into the current call.

But what about if 20 or 30 people call in, and it’s firing off 30
copies of mp3Player? Won’t that start to degrade bandwidth?

In an ideal world, I was wondering it there was any way of all callers
“tapping into” the same audio socket, and if there were no callers, the player would be inactive, and the first “new” caller would kick the stream into life again.

Does that make any sense at all?

Thanks!

Asterisk 13.7.2 (but excited about 13.8 with bundled PJsip!) on Ubuntu
15.10 Wiley.