[Freeswitch-dev] Eavesdrop doesn't work for WebRTC calls

[Yan Brenman]

Hello,

I have been running the 1.4.beta branch of the FreeSWITCH to do some 
WebRTC related testing. And I noticed that "eavesdrop" application,
although working nicely for the any other types of clients, doesn't work 
for the WebRTC. I am not getting neither video nor audio (I did merge
changes from the video-media-bug branch and do get video "eavesdropped" 
nicely using jitsi clients) using SIPML5 clients.

I will be more than happy to do any code modifications and testing 
required, just need some initial input as I am relatively new to WebRTC.
I looked at the current implementation of "eavesdrop" and everything 
seems to make perfect sense. Why it doesn't work for the WebRTC then?

Please let me know if there is anything I can provide.

Any help is greatly appreciated.

Thanks
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