Dear Baresip Developers,

I am testing your SIP client to create an IP telephony setup combined with AI-generated voice responses.

I am not a professional C developer, so I am running and testing your client in Docker using the corresponding repository:
https://github.com/baresip/docker

During my work, I encountered a few questions:

Does the client support redirecting the RTP data stream to an external server where I could process it?

Does your client support handling multiple channels? For example, if multiple clients call my number, can I answer all of them?

I also found that you have an option to enable the gst streaming module, which I believe could help with this. However, I was unable to install it correctly in the Docker image.

I tried adding it via the flag:

-DMODULES="gst"

I also uncommented it in the config file, but when launching the container, I still receive a message saying that the module was not found.

Could you advise me on how to get it running?

And in general, am I on the right track in trying to implement my task this way?

I appreciate your comments in advance.