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<div dir="ltr">concerning:<br>
<br>
>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>
Do you see a way for re-establishing the call in a way that ZRTP
may be triggered on the client's side? Is there a way to
re-establish a call beyond a SIP reinvite? Or maybe force some
new codecs to be negociated?<br>
<br>
Best regards<br>
Peter<br>
</div>
<br>
<br>
<br>
On 02/10/15 20:59, Brian West wrote:<br>
</div>
<blockquote
cite="mid:CAEJMVkATrpeFm32k414tfpxfNH0J0cdboj3kjpCDYHUjE9Toww@mail.gmail.com"
type="cite">
<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 moz-do-not-send="true"
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
moz-do-not-send="true"
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 moz-do-not-send="true" href="mailto:lists" target="_blank">mailto:lists</a> (att) <a moz-do-not-send="true" href="http://telefaks.de" target="_blank">telefaks.de</a>
Internet: <a moz-do-not-send="true" href="http://www.telefaks.de" target="_blank">www.telefaks.de</a>
</pre>
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-- <br>
<div>
<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
moz-do-not-send="true"
href="mailto:brian@freeswitch.org"
target="_blank">brian@freeswitch.org</a></span></font></p>
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size="1"><img moz-do-not-send="true"
src="http://billing.freeswitch.org/templates/default/img/whmcslogo.png"><br>
</font></p>
<p><font face="courier new, monospace"><b><i>Twitter:
@FreeSWITCH , @briankwest</i></b><br>
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href="http://www.freeswitchbook.com"
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<pre>_________________________________________________________________________
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</blockquote>
<br>
<br>
<pre cols="72">--
With kind regards
Peter Steinbach
Telefaks Services GmbH
<a moz-do-not-send="true" href="mailto:lists" target="_blank">mailto:lists</a> (att) <a moz-do-not-send="true" href="http://telefaks.de" target="_blank">telefaks.de</a>
Internet: <a moz-do-not-send="true" href="http://www.telefaks.de" target="_blank">www.telefaks.de</a>
</pre>
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<br>
_________________________________________________________________________<br>
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href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
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<br>
Official FreeSWITCH Sites<br>
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</div>
-- <br>
<div class="gmail_signature">
<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
moz-do-not-send="true"
href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
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moz-do-not-send="true"
src="http://billing.freeswitch.org/templates/default/img/whmcslogo.png"><br>
</font></p>
<p><font face="courier new, monospace"><b><i>Twitter:
@FreeSWITCH , @briankwest</i></b><br>
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href="http://www.freeswitchbook.com" target="_blank">http://www.freeswitchbook.com</a><br>
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href="http://www.freeswitchcookbook.com"
target="_blank">http://www.freeswitchcookbook.com</a></font></p>
<p><font face="courier new, monospace"><b>T:</b>+19184209001
| <b>F:</b>+19184209002 | <b>M:</b>+1918424WEST (9378)<br>
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<pre wrap="">_________________________________________________________________________
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Official FreeSWITCH Sites
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</blockquote>
<br>
<br>
<pre class="moz-signature" cols="72">--
With kind regards
Peter Steinbach
Telefaks Services GmbH
<a class="moz-txt-link-freetext" href="mailto:lists">mailto:lists</a> (att) telefaks.de
Internet: <a class="moz-txt-link-abbreviated" href="http://www.telefaks.de">www.telefaks.de</a>
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