<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">how about on master?<div><br><div style=""><div>On Mar 15, 2014, at 10:19 PM, Yan Brenman <<a href="mailto:ybrenman@xopnetworks.com">ybrenman@xopnetworks.com</a>> wrote:</div><br class="Apple-interchange-newline"><blockquote type="cite">
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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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