Sorry, it's answered and parked. And I expect to hear audio and see video, as i do with non-WebRTC client (jitsi for example).<br><br>On Monday, March 24, 2014, Michael Jerris <<a href="mailto:mike@jerris.com">mike@jerris.com</a>> 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">What do you expect to be hearing when eavesdropping on a parked channel?<div><br><div>
<div>On Mar 21, 2014, at 9:05 PM, Yan Brenman <<a href="javascript:_e(%7B%7D,'cvml','ybrenman@xopnetworks.com');" target="_blank">ybrenman@xopnetworks.com</a>> wrote:</div><br><blockquote type="cite">
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<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.<br>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<br>
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<br>branch - I will get the video too.<span> </span><br><br>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<br>
outbound leg which is "eavesdrop" on the inbound "parked" leg. I hope you don't see anything wrong with it.<br><br>Thanks again<br><br></font><div>On 3/21/2014 8:38 AM, Ken Rice wrote:<br></div><blockquote type="cite">
<font face="Monaco, Courier New"><span style="font-size:11pt">You should be testing webrtc stuff on the master branch... Theres been a lot of changes going on in tree.. The video-media-bug branch is probably incomplete... I don’t think its even close to ready for merging at this point...<br>
<br><br>On 3/18/14 2:01 PM, "Yan Brenman" <<a>yb91423@gmail.com</a>> wrote:<br><br></span></font><blockquote><font face="Monaco, Courier New"><span style="font-size:11pt">Hello,<br><br>I was trying to get the eavesdrop application work on the 1.4.beta branch for the WebRTC clients (works great for all other<br>
types of clients) and could not. Neither audio nor video (I merged the code from the video-media-bug branch for the video)<br>is going through. For the client I tried both - sipml5 and jssip clients.<br><br>Please let me know if there is any other piece of information I can provide.<br>
<br>I will be more then happy to do any kind of code changes and testing - just need some starting points. Especially considering<br>that I am very new to the WebRTC.<br><br>Every piece of advice is greatly appreciated. <br>
<br>Thanks<br>Yan<br><br></span></font></blockquote></blockquote></div></blockquote></div><br></div></div></blockquote>