I am using JsSIP to make SIP calls. I am able to see Signalling messages like From, To, Via in the console.
In the same way, can we see RTP and RTCP packet transmission messages?
No. You can see the signalling part (SIP or JSEP), but not the media part in the console. RTP/RTCP are the media protocols.
You can diagnose RTP headers and RTCP packets by using chrome's webrtc-internals dump files.
https://tokbox.com/blog/how-to-get-a-webrtc-diagnostic-recording-from-chrome-49/