<div dir="ltr"><font size="-1">I just tried the "master" branch and situation there with the
"eavesdrop" application is absolutely the same as it is on the
"1.4.beta" branch. It doesn't work for the WebRTC clients and it does work for the
non-WebRTC clients (I used jitsi client to test). Having said that
I was only getting audio working with "eavesdrop" for non-WebRTC clients but
I am pretty sure if I merge in some code I used to from the video-media-bug branch - I will get the video too.</font><br><div><font size="-1"><br></font></div><div><font size="-1">Just to describe you my test scenario - I am placing and inbound
leg with I am "answering" and "park". Then from the FS_CLI I am
"originating" an outbound leg which is "eavesdrop" on the inbound "parked" leg. I
hope you don't see anything wrong with it.</font><font size="-1"><br></font></div><div><font size="-1"><br></font></div><div><font size="-1">Thanks</font></div><div><font size="-1">Yan</font></div><div><font size="-1"><br>
</font></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"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></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 <<a href="mailto:ybrenman@xopnetworks.com" target="_blank">ybrenman@xopnetworks.com</a>> 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 "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>
</div>
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