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<font size="-1">Hello,<br>
<br>
I have been running the 1.4.beta branch of the FreeSWITCH to do
some WebRTC related testing. And I noticed that "eavesdrop"
application,<br>
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<br>
changes from the video-media-bug branch and do get video
"eavesdropped" nicely using jitsi clients) using SIPML5 clients.<br>
<br>
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. <br>
I looked at the current implementation of "eavesdrop" and
everything seems to make perfect sense. Why it doesn't work for
the WebRTC then?<br>
<br>
Please let me know if there is anything I can provide.<br>
<br>
Any help is greatly appreciated.<br>
<br>
Thanks</font>
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