Hello, can u correct me if im wrong. currently im using webrtc to use jsep given by sip when incoming call comes and sink from pc giving me the pcm samples which i can use for external uses, do i really need to use rtp forward here because it is just 1 to 1 peer …
Not sure I understand the question, but RTP forwarding will forward still encoded audio frames: Janus and the SIP plugin will not decode the audio.