Is my observation correct that RTP/RTCP packets from a webRTC stream cannot be analyzed in Wireshark running on the same desktop to analyze RTP packets because the browser would have encrypted them using DTLS/SRTP?
I know there are some browser APIs to help but is there any other approach? libpcap if used to write some tool will probably have the same problem.
Firefox has support for dumping the decrypted RTP/RTCP packets into the log files, described here. Chrome does not have something similar unfortunately.
If you use a server, some of them like Janus have the ability to generate similar dumps, see here