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<p>Hello,</p>
<p>for phones that are using tls, you can do the following tests:</p>
<p> - for incoming traffic: proto==TLS<br>
- for outgoing traffic: after lookup location, the R-URI ($ru)
should have 'transport=tls'</p>
<p>For RTPEngine there are some flags to specify you want or not
SDES SRTP, I used them few times in the past, but I don't recall
them by heart -- the docs should have them.</p>
<p>Cheers,<br>
Daniel<br>
</p>
<br>
<div class="moz-cite-prefix">On 26.07.17 06:40, David Cunningham
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CAHGbP-_p6pcSLcUK2RcX=Te0RHw5wgAf-1V7BY7v9e6RBLg_vQ@mail.gmail.com">
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<div>Hello,<br>
<br>
</div>
We're configuring Kamailio 4.2 with rtpengine to act as a
midpoint between a telephone using TLS/SRTP and Asterisk.
There are examples out there for TLS/SRTP with WebRTC, but
we're using a plain hard phone, not WebRTC.<br>
<br>
</div>
<div>Would anyone be able to point us towards a Kamaiio
configuration which:<br>
<br>
</div>
<div>a) Tests if the destination phone (stored using usrloc)
uses TLS.<br>
<br>
</div>
<div>b) Sends RTP for calls to a TLS phone to rtpengine to be
encrypted. We can assume all phones using TLS want to use
SRTP.<br>
<br>
</div>
<div>Thanks very much in advance.<br>
</div>
<div>
<div>
<div><br>
-- <br>
<div class="gmail_signature"
data-smartmail="gmail_signature">
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<div>
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<div>David Cunningham, Voisonics Limited<br>
<a href="http://voisonics.com/" target="_blank"
moz-do-not-send="true">http://voisonics.com/</a><br>
USA: +1 213 221 1092<br>
Australia: +61 (0) 2 8063 9019<br>
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</blockquote>
<br>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla
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