Bridging a Live Video Stream to a SIP Call (video)

[ivanxx1]

Hello Community,

I am exploring a custom video integration with Asterisk and, being relatively new to its advanced capabilities like this, I was hoping to get some feedback on a concept I’m considering.

My goal is to have Asterisk answer an incoming SIP video call and then play content from an external, continuous live stream (like RTMP or RTSP) back to the caller.

My initial idea is to use an AGI script to launch an external FFmpeg process. The script would manage the FFmpeg instance, which would be responsible for pulling the video from the live source and streaming it to the SIP call’s RTP endpoint.

I understand this is likely not a standard, out-of-the-box feature. Before I invest too much time developing this, I was hoping to start a discussion here. Does this general approach (AGI + FFmpeg) seem viable from a conceptual standpoint? Are there any obvious pitfalls or architectural considerations I might be missing?

I’m truly open to any thoughts or alternative ideas. I’d be grateful for any feedback on the concept itself.

Thank you in advance for sharing your time and expertise.

[jcolp]

@ivanxx1

Does this general approach (AGI + FFmpeg) seem viable from a conceptual standpoint?

This isn’t supported at all. Meaning there’s no C code implemented to do it, so if you go down this road - you’re going to have to modify Asterisk.

In general there isn’t really anything implemented to allow this. You’re far enough off the normal path that you’re down the road of writing/modifying C code no matter what.