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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.

asinix
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1 Answers1

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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

Philipp Hancke
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  • 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 – asinix Nov 14 '18 at 08:14
  • Can u please help me on this [question?](https://stackoverflow.com/questions/53673672/webrtc-relationship-between-channels-tracks-streams-vis-a-vis-rtp-ssrc-and-r) – asinix Dec 07 '18 at 17:09