I'm recording an audio conference using RestComm and the Telestax MediaServer. I would like to forward the RTP data of that conference, so I could do some real-time processing with it using gstreamer. I was taking a look to the documentation but I didn't find how to do it easily with RestComm.
Some guidance/documentation here would be welcomed so I can start to work on it (configuration or maybe extending RestComm if needed)
Did you try joining a mute participant to the conference. This muted participant would be your SIP + gstreamer client that could do some real time processing on the media coming from the audio conference ?