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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 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>
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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. What
happens when you eavesdrop a bridged call with audio only? Does
call recording behave differently?
<div><br>
<div>
<div>On Mar 24, 2014, at 9:58 AM, Yan Brenman <<a
moz-do-not-send="true" href="mailto:yb91423@gmail.com">yb91423@gmail.com</a>>
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 <<a
moz-do-not-send="true" 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
moz-do-not-send="true"
href="javascript:_e(%7B%7D,'cvml','ybrenman@xopnetworks.com');"
target="_blank">ybrenman@xopnetworks.com</a>>
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> </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
moz-do-not-send="true">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>
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