<div dir="ltr">Thank you very much for the reply. Yes, I just got another reply suggesting to try master branch.<div>The reason I was working with 1.4.beta branch is because that&#39;s what</div><div>Wiki is suggesting. I am going to do this now.</div>
<div><br></div><div>Thanks</div><div>Yan</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Mar 21, 2014 at 8:00 AM, Michael Jerris <span dir="ltr">&lt;<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">how about on master?<div><br><div><div>On Mar 15, 2014, at 10:19 PM, Yan Brenman &lt;<a href="mailto:ybrenman@xopnetworks.com" target="_blank">ybrenman@xopnetworks.com</a>&gt; wrote:</div>
<br><blockquote type="cite">
  

    
  
  <div bgcolor="#FFFFFF" text="#000000">
    <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 &quot;eavesdrop&quot;
      application,<br>
      although working nicely for the any other types of clients,
      doesn&#39;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
      &quot;eavesdropped&quot; 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 &quot;eavesdrop&quot; and
      everything seems to make perfect sense. Why it doesn&#39;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>
  </div>

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