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    <font size="-1">And just for completeness - eavesdrop on the bridged
      audio call doesn't work either.<br>
      <br>
    </font>
    <div class="moz-cite-prefix">On 3/24/2014 10:41 AM, Yan Brenman
      wrote:<br>
    </div>
    <blockquote cite="mid:53306E53.2000407@xopnetworks.com" type="cite">
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      <font size="-1">I understand the situation with the video
        (although like I said - I did merged code from video-media-bug
        branch and got video working for non-webrtc clients).<br>
        <br>
        Eavesdropping on the answered and parked call leg from WebRTC
        client with audio only gives the same&nbsp; result - no audio on the
        eavesdropping leg. Although,<br>
        according to the log, everything went ok without any problems. I
        compared the SDP session dump from the creating eavesdropping
        leg to the dump where I<br>
        would just "uuid_bridge" two webrtc call legs - and there are no
        differences there (like I thought the eavesdropping leg would
        somehow end up using the<br>
        wrong cocec or something). <br>
        <br>
        I have not tried to eavesdrop an already bridged call as ideally
        this is not the scenario I would be planning to use - I don't
        want caller leg (the one we are <br>
        eavesdropping on) to get terminated if call leg it is bridged to
        hangs up. <br>
        And I have not tried the recording yet. Will do it next.<br>
        <br>
        Thanks<br>
        <br>
      </font>
      <div class="moz-cite-prefix">On 3/24/2014 7:10 AM, Michael Jerris
        wrote:<br>
      </div>
      <blockquote
        cite="mid:F79CA111-85B6-435C-B3A7-EE69D02BF964@jerris.com"
        type="cite">
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        I'll leave the video question totally alone as thats code that
        isn't even merged in yet, and still very under development.
        &nbsp;What happens when you eavesdrop a bridged call with audio only?
        &nbsp;Does call recording behave differently?
        <div><br>
          <div>
            <div>On Mar 24, 2014, at 9:58 AM, Yan Brenman &lt;<a
                moz-do-not-send="true" href="mailto:yb91423@gmail.com">yb91423@gmail.com</a>&gt;

              wrote:</div>
            <br class="Apple-interchange-newline">
            <blockquote type="cite">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 &lt;<a
                moz-do-not-send="true" href="mailto:mike@jerris.com">mike@jerris.com</a>&gt;

              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 &lt;<a
                          moz-do-not-send="true"
                          href="javascript:_e(%7B%7D,'cvml','ybrenman@xopnetworks.com');"
                          target="_blank">ybrenman@xopnetworks.com</a>&gt;

                        wrote:</div>
                      <br>
                      <blockquote type="cite">
                        <div bgcolor="#FFFFFF" text="#000000"
style="font-family:Helvetica;font-size:12px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><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>&nbsp;</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&#8217;t think its even
                                close to ready for merging at this
                                point...<br>
                                <br>
                                <br>
                                On 3/18/14 2:01 PM, "Yan Brenman" &lt;<a
                                  moz-do-not-send="true">yb91423@gmail.com</a>&gt;

                                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.&nbsp;<br>
                                  <br>
                                  Thanks<br>
                                  Yan<br>
                                  <br>
                                </span></font></blockquote>
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