How to achieve real time streaming/processing of RTP Packets

[anbhowmi]

Hello, I am very new to audio processing. I followed the project guide, and now I have Drachtio + RTPengine running on Docker and the node app running in my system. SBCs are configured to send traffic through and I am able to see call recording as PCAP inside the Docker container.
My question is after RTPengine allocates the Endpoints, I can parse the SDP to get the media ports for both caller and callee. I tried to pass that to ffmpeg to generate .wav files, but got Invalid Input error.
What can be done to have Real Time processing of the RTP streams?