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<p>Thank you very much Arsen.</p>
<p>Knowing that this was supposed to work was what I needed to know.
I was only missing the NAT rule back to my phone system on the TLS
port, which is where the BYE packet was being sent to.</p>
<p>All works perfectly now.<br>
</p>
<div class="moz-cite-prefix">On 9/19/20 4:05 AM, Arsen Semenov
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CAC3hFAp0tzCveJOWbCOX7SS81Ko=xuE0yKsSsUFRe40KjFFP7A@mail.gmail.com">
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<div dir="auto">
<div dir="auto">Hello Conrad, </div>
<div dir="auto"><br>
</div>
<div dir="auto">Bit hard to say exactly without looking into
logs/dumps but it seems like your call is long enough so
router (if you have one) could timeout on nat tcp
connection, thus the “bye” request can not reach uac. Just
guessing. </div>
<div dir="auto"><br>
</div>
<div dir="auto">Please check whether you have tcp keepalive
enabled, example:</div>
<div dir="auto">tcp_keepalive=yes</div>
<div dir="auto">tcp_crlf_ping=yes</div>
<div dir="auto">tcp_keepcnt=3</div>
<div dir="auto">tcp_keepidle=30</div>
<div dir="auto">tcp_keepintvl=30</div>
<div dir="auto"><br>
</div>
</div>
</div>
<div dir="auto">Hope it helps.</div>
<div dir="auto"><br>
</div>
<div><br>
<div class="gmail_quote">
<div dir="ltr" class="gmail_attr">On Sat, 19 Sep 2020 at
8:59 AM, <a href="mailto:conradcordeiro@gmail.com"
target="_blank" moz-do-not-send="true">conradcordeiro@gmail.com</a>
<<a href="mailto:conradcordeiro@gmail.com"
target="_blank" moz-do-not-send="true">conradcordeiro@gmail.com</a>>
wrote:<br>
</div>
<blockquote class="gmail_quote" style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-style:solid;padding-left:1ex;border-left-color:rgb(204,204,204)">Hello,<br>
<br>
<br>
<br>
Thank you for your reading this and for your help.<br>
<br>
<br>
<br>
I'm a Kamailio newbie and managed to set up an edge proxy,
which works <br>
<br>
perfectly on UDP traffic. I'm now attempting to deploy
TLS/SRTP and <br>
<br>
everything almost works perfectly. The single issue I'm
having is that <br>
<br>
when either of the parties to an SRTP/TLS call disconnect,
the other <br>
<br>
party's call remains active. With UDP, when one of the
parties <br>
<br>
disconnects the call, the other leg of the call receives
the BYE command <br>
<br>
and the call automatically disconnects.<br>
<br>
<br>
<br>
This is how I have our infrastructure set up:<br>
<br>
<br>
<br>
1. Twilio SIP Trunk with Secure Media enabled.<br>
<br>
<br>
<br>
2. Kamailio 5.4.1 set up with TLS module, set to listen on
TLS port <br>
<br>
5061, SSL certificates from Let's Encrypt, route set to
our phone system.<br>
<br>
<br>
<br>
3. Phone system is Asterisk.<br>
<br>
<br>
<br>
As per above, everything works almost perfectly with
TLS/SRTP. The only <br>
<br>
issue is that calls will not disconnect when one of the
sides hang up. <br>
<br>
If I disable TLS/SRTP and use UDP only, everything works.<br>
<br>
<br>
<br>
Audio is just fine with TLS/SRTP.<br>
<br>
<br>
<br>
Does anyone know why calls with SRTP/TLS will not
disconnect <br>
<br>
automatically when one of the parties ends the call?<br>
<br>
<br>
<br>
Thank you,<br>
<br>
<br>
<br>
Conrad<br>
<br>
<br>
<br>
<br>
<br>
_______________________________________________<br>
<br>
Kamailio (SER) - Users Mailing List<br>
<br>
<a href="mailto:sr-users@lists.kamailio.org"
target="_blank" moz-do-not-send="true">sr-users@lists.kamailio.org</a><br>
<br>
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href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users"
rel="noreferrer" target="_blank" moz-do-not-send="true">https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users</a><br>
<br>
</blockquote>
</div>
</div>
<br>
<br>
</div>
-- <br>
<div dir="ltr" class="gmail_signature"
data-smartmail="gmail_signature">Sent from Gmail Mobile</div>
<br>
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<pre class="moz-quote-pre" wrap="">_______________________________________________
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