Analyzing RTP packets from browser's webRTC stream using Wireshark or similar tool

Question

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.

Guthrie

Answer 1

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

Fractionate

I guess that keeping a middleware in between will need to do the entire DTLS-SRTP handshake, copy the rtp/rtcp packets and blindly forward them to the other peer but then we are dealing with all related issues which Janus-type SFUs/MCUs deal with. The Firefox solution should suffice. Thanks – Guthrie

Can u please help me on this question? – Guthrie