<div dir="ltr">ZRTP happens in the media stream, what you're seeing is a zid in the sdp, which doesn't mean that it will start the ZRTP exchange, your client or FS will have to trigger the start by doing the hello, which isn't what you want.</div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Feb 10, 2015 at 11:58 AM, Peter Steinbach <span dir="ltr"><<a href="mailto:lists@telefaks.de" target="_blank">lists@telefaks.de</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
<div>Hello Brian,<br>
<br>
thanks for the hint. In the meantime, I tried both, setting zrtp
globally in Freeswitch and having zrtp in the original dialplan
and in the {}-Variables for dialing the B party. This did not
help.<br>
<br>
Then I totally disabled ZRTP inside FSs and in the dialplans and
grepped on the network. Then I issued a "uuid_media off" for
starting a reinvite between the 2 endpoints. On the Freeswitch
console, I could see, that ZRTP is exchanged in the SDP, although
Freeswitch does a reinvite with ZRTP.<br>
Having a look at the channel variables, I could see that<br>
<ul>
<li>variable_rtp_local_sdp_str does _not_ contain ZRTP</li>
<li>variable_switch_m_sdp _does_ contain ZRTP</li>
<li>variable_switch_r_sdp _does_ contain ZRTP</li>
</ul>
<p>But ZRTP however is not started on both sides. Then I killed
Freeswitch during a call and started again, the calls were
automatically recovered, and ZRTP was there. But this was surely
done with a normal invite, right?<br>
</p>
<p>I tried with CsipSimple on Android and Jitsi nightly on Ubuntu.
Both clients behave the same. I assume, that they will not start
ZRTP on a reinvite.<br>
</p>
<p>So unless there is now way to reistablish a call (I tried
uuid_transfer to the same number with autoanswer, but target
number was still considered as busy), I think it will not work.<br>
</p>
<p>Any other hints?<br>
</p>
Best regards <br><span class="HOEnZb"><font color="#888888">
Peter</font></span><div><div class="h5"><br>
<br>
<br>
<br>
On 02/10/15 16:05, Brian West wrote:<br>
</div></div></div><div><div class="h5">
<blockquote type="cite">
<div dir="ltr">You can't set the zrtp_secure_media=true after a
session is created, it needs to be a global or you set it inside
the {} on the bridge or originate line. Then you're outlining
uuid_media off, which would make the media go between the two
endpoints, FreeSWITCH isn't even involved in the zrtp exchange
at that point. Technically your endpoints should be the ones
doing this and ZRTP should be disabled in FreeSWITCH, You really
only need ZRTP with MiTM if you MUSt transcode or you're
terminating directly on the FreeSWITCH instance. I should
really just rip out MiTM support.</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Tue, Feb 10, 2015 at 5:53 AM, Peter
Steinbach <span dir="ltr"><<a href="mailto:lists@telefaks.de" target="_blank">lists@telefaks.de</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000"> Hello,<br>
<br>
I want to establish ZRTP, after I am bridging 2 calls,
which have been active already before.<br>
<br>
My scenario is as follows:<br>
<ul>
<li>Caller A is calling Freeswitch, the call is answered
and an announcement is played</li>
<li>In the meantime Freeswitch calls another Party B and
connects it to an internal IVR, where also
announcements are played.</li>
<li>Now, after a while, I decide to connect the calls
together, then send an "uuid_media off <uuid>"
to the calling party A leg, reinvites occur and bypass
media works.</li>
<li>However the 2 SIP clients (Csipsimple) do not start
a ZRTP session</li>
<li>If caller A calls B directly via bypass media, the
ZRTP is active, by the way.<br>
</li>
</ul>
<p>I suspect that ZRTP somehow should be forced or
triggered. Is there a way to do that? I already invited
the B leg with "zrtp_secure_media=true" and issued</p>
<p> uuid_setvar 1bd508af-19c7-4e38-8621-731baaf33920
zrtp_secure_media true<br>
uuid_media off 1bd508af-19c7-4e38-8621-731baaf33920<br>
to the A leg. But this did not work.<br>
</p>
<p>The main question should be: How can I trigger a ZRTP
connection on an existing call?<span><font color="#888888"><br>
</font></span></p>
<span><font color="#888888"> <br>
<pre cols="72">--
With kind regards
Peter Steinbach
Telefaks Services GmbH
<a href="mailto:lists" target="_blank">mailto:lists</a> (att) <a href="http://telefaks.de" target="_blank">telefaks.de</a>
Internet: <a href="http://www.telefaks.de" target="_blank">www.telefaks.de</a>
</pre>
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-- <br>
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<div dir="ltr">
<p><font face="courier new, monospace"><b><i><font size="4">Brian
West</font></i></b><br>
<span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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</font></p>
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</blockquote>
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<br>
<pre cols="72">--
With kind regards
Peter Steinbach
Telefaks Services GmbH
<a href="mailto:lists" target="_blank">mailto:lists</a> (att) <a href="http://telefaks.de" target="_blank">telefaks.de</a>
Internet: <a href="http://www.telefaks.de" target="_blank">www.telefaks.de</a>
</pre>
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<br>
Official FreeSWITCH Sites<br>
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<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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