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<p>Hello,<br>
</p>
<div class="moz-cite-prefix">On 06.08.20 16:26,
<a class="moz-txt-link-abbreviated" href="mailto:amitsharma@coraltele.com">amitsharma@coraltele.com</a> wrote:<br>
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<div class="WordSection1">
<p class="MsoNormal">Hello Daniel,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thank a lot for such an elaborative reply,
it will really help in many ways. It is clear that in case of
big system the Progress call transition may not be possible.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I want to know two more things : <o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<ol style="margin-top:0in" type="1" start="1">
<li class="MsoListParagraph"
style="margin-left:0in;mso-list:l0 level1 lfo4">Can we built
Re-Homing around Kamailio (Move call from Kamailio to
Freeswitch). Is there any possibility of doing it.</li>
</ol>
</div>
</blockquote>
<p><br>
</p>
<p>Both applications are open source and you can develop extensions
in both of them to share call data and maybe you will get what you
want. But from the SIP specification, there is no mechanism to
migrate a server from a proxy (like Kamailio) to an endpoint/b2bua
like FreeSwitch. Therefore at this moment is no option to do that.<br>
</p>
<p>As I mentioned, another kamailio can route just fine requests
belonging to a call initiated via another Kamailio. FreeSwitch (or
other B2BUA/endpoint) can do re-INVITE and recover the call after
a crash and restart on the same system or on another system, but
that because it was part of the call and it is allowed to change
its contact/IP address <br>
</p>
<blockquote type="cite"
cite="mid:00d001d66bfd$837a8130$8a6f8390$@coraltele.com">
<div class="WordSection1"><br>
<ol style="margin-top:0in" type="1" start="1">
<li class="MsoListParagraph"
style="margin-left:0in;mso-list:l0 level1 lfo4">How can
capture sip-info from Kamailio to Freeswitch. Means DTMP
pressed.</li>
</ol>
</div>
</blockquote>
<p>I do not understand this one, maybe you can elaborate more.</p>
<p>Cheers,<br>
Daniel<br>
</p>
<blockquote type="cite"
cite="mid:00d001d66bfd$837a8130$8a6f8390$@coraltele.com">
<div class="WordSection1">
<ol style="margin-top:0in" type="1" start="1">
<li class="MsoListParagraph"
style="margin-left:0in;mso-list:l0 level1 lfo4"><o:p></o:p></li>
</ol>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thanks in advance.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Amit Sharma<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<div>
<div style="border:none;border-top:solid #E1E1E1
1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b>From:</b> Daniel-Constantin Mierla
<a class="moz-txt-link-rfc2396E" href="mailto:miconda@gmail.com"><miconda@gmail.com></a> <br>
<b>Sent:</b> Wednesday, August 5, 2020 6:33 PM<br>
<b>To:</b> Kamailio (SER) - Users Mailing List
<a class="moz-txt-link-rfc2396E" href="mailto:sr-users@lists.kamailio.org"><sr-users@lists.kamailio.org></a>;
<a class="moz-txt-link-abbreviated" href="mailto:amitsharma@coraltele.com">amitsharma@coraltele.com</a><br>
<b>Subject:</b> Re: [SR-Users] Call in Progress Recovery
in Redundancy<o:p></o:p></p>
</div>
</div>
<p class="MsoNormal"><o:p> </o:p></p>
<p>Hello,<o:p></o:p></p>
<p>first we need to clarify that it seems you are actually not
looking for redundancy of active transactions, which I tried
to focus on the answer during the ClueCon session last
evening.<o:p></o:p></p>
<p>My answer there related to htable was about the ability to
route CANCEL requests where the INVITE was forwarded.<o:p></o:p></p>
<p>Like Julien replied on another email, SIP has couple of
mechanism to "recover" or "go through" in case of transaction
states being lost. For example, with UDP if the proxy is
restarted after receiving the INVITE and not sending any
reply, then there is a retranmission of the INVITE for couple
of times (can be up to 30seconds or more, depending on UA
settings). So the INVITE comes again to the proxy, which can
handle it (assuming a fast enough restart). Then, if the
INVITE was forwarded, the responses to it can be routed
without any problem, using the Via headers.<o:p></o:p></p>
<p>Considering that the SIP transaction is about keeping the
states of routing the request until a final response is sent
out, one of the main benefits is the ability to re-route the
request to a new address if the first selected destination
doesn't answer (aka, serial forking). But if you have
one-to-one routing policy (like receiving from the phone and
sending to a freeswitch), then you can also do stateless
forwarding. In such case, if you migrate the ip to another
Kamailio node, it can route the replies even when the request
was routed by previous active node.<o:p></o:p></p>
<p>As far as I can remember from some demos at past cluecon
events, the FreeSwitch call recovery was based on re-INVITEs,
which means the call has to be established to know where to
send the re-INVITE, be aware of caller/callee contact
addresses, codecs, routing headers, ... Recovering a progress
call from a B2BUA like FreeSwitch can be as difficult as for a
proxy, if you want to cover over possible scenarios related to
serial and parallel forking, branches added on the fly when a
new registration comes in, different retransmission timers per
branches, storage of most relevant replies for branches, etc
... just to enumerate from the impact on the SIP
specification, but each application has a lot of event
callbacks, structures and parameters associated with a
transaction (e.g., for accounting, message logging, ...), ...
so the eco-system around a SIP transaction is very fluid,
shifting to another node could be impossible.<o:p></o:p></p>
<p>For example, consider that first retransmission has to be
done in 500ms, followed by 1sec, 2sec, 4sec -- in a case of a
shared IP active-standby system, detection that node is done
typically takes a few seconds itself, so retransmission steps
can be lost for sure.<o:p></o:p></p>
<p>Kamailio itself is not a B2BUA so it cannot re-INVITE inside
a call, but many Kamailio systems can route SIP
requests/replies from the same call (e.g., INVITE routed by
Kamailio A and the BYE by Kamailio B), it is a matter of what
you set in Record-Route headers, or do anycast routing to a
cluster of Kamailio nodes. When you hear about getting out of
the call, is about RTP (audio/video) streams, because from
signaling point of view, a B2BUA is an endpoint in each of the
two legs of the calls, it can do re-INVITE to move RTP streams
to be end-to-end, but it has to stay in the signaling path. An
endpoint can get out of the call via a transfer to another
endpoint, but then it cannot transfer the call back to it.<o:p></o:p></p>
<p>Also, let's say the call is completed without going to
freeswitch with the initial INVITE, afterward you cannot hand
it over to Freeswitch. But you can route initial INVITE to
Kamailio, do not do record-routing, and send it to freeswitch.
By not doing record-routing, requests within dialog
(re-INVITE, BYE, etc..) and not coming to Kamailio, they go
directly to FreeSwitch. But you have to be careful with natted
devices, typically they can get messages back only from the
box where they sent the initial INVITE.<o:p></o:p></p>
<p>The discussion can be long here, as I tried to say, if you
have the very simple scenario of one-to-one routing rule, then
even going (sip-transaction-)stateless can work, but to cover
all cases with parallel/serial forking and multiple active
branches at different stages of processing is not working.<o:p></o:p></p>
<p>My feeling is that you were thinking from your experience
with freeswitch/b2bua systems, where when you restart the
b2bua in a ringing state the call does not complete. But if
use Kamailio to route the call from Alice to Bob, it gets to
ringing state, then you can restart kamailio and call gets
completed (the answer -- the 200ok response -- is routed by
Kamailio correctly). Of course, depending on what other
modules you use, some specific processing may be lost for such
calls, but case by case, there can be solutions.<o:p></o:p></p>
<p>Cheers,<br>
Daniel<o:p></o:p></p>
<div>
<p class="MsoNormal">On 05.08.20 12:36, <a
href="mailto:amitsharma@coraltele.com"
moz-do-not-send="true">amitsharma@coraltele.com</a> wrote:<o:p></o:p></p>
</div>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<p class="MsoNormal">Dear Daniel/Team,<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">I had raised one question in “Workshop 3
– Kamailio” at Cluecon 2020(Last Night), i.e. Can Progress
Call(Ringing Calls) be recovered in case of redundancy with
Kamailio. You were told me that straight way it is not
possible but try with hash table once. I had tried following
link <a
href="https://wazo-platform.org/blog/kamailio-ha-dispatcher-and-dmq"
moz-do-not-send="true">https://wazo-platform.org/blog/kamailio-ha-dispatcher-and-dmq</a>
and able to recover Call in progress within 2-3 nodes.<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<ol style="margin-top:0in" type="1" start="1">
<li class="MsoListParagraph"
style="margin-left:0in;mso-list:l1 level1 lfo3">My one
question is that either this approach will work in
production or not.<o:p></o:p></li>
<li class="MsoListParagraph"
style="margin-left:0in;mso-list:l1 level1 lfo3">I have
been using Freeswitch for last 6-7 years but “Call in
Progress Recovery in Redundancy” is not possible there in
“Freeswitch”, So I tried Kamailio and got success. My
Second question is that can it be possible that Call
established on Kamailio and after call set up Kamailio
leave that call and handed over it to Freeswitch for
further processing(Like Re-homing available in OpenSIPS).
This will save years of time that I have invested building
features around Freeswitch.<o:p></o:p></li>
</ol>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal">Please suggest me the best way possible
to achieve this.<o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"> <o:p></o:p></p>
<p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:"Arial",sans-serif;color:gray">Thanks
& Regards,</span></b><o:p></o:p></p>
<p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:"Arial",sans-serif;color:gray">Amit
Sharma</span></b><o:p></o:p></p>
<p class="MsoNormal"><b><span
style="font-size:10.0pt;font-family:"Arial",sans-serif;color:gray">(Sr.
Team Leader)</span></b><o:p></o:p></p>
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<br>
<o:p></o:p></p>
<pre>_______________________________________________<o:p></o:p></pre>
<pre>Kamailio (SER) - Users Mailing List<o:p></o:p></pre>
<pre><a href="mailto:sr-users@lists.kamailio.org" moz-do-not-send="true">sr-users@lists.kamailio.org</a><o:p></o:p></pre>
<pre><a href="https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users" moz-do-not-send="true">https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users</a><o:p></o:p></pre>
</blockquote>
<pre>-- <o:p></o:p></pre>
<pre>Daniel-Constantin Mierla -- <a href="http://www.asipto.com" moz-do-not-send="true">www.asipto.com</a><o:p></o:p></pre>
<pre><a href="http://www.twitter.com/miconda" moz-do-not-send="true">www.twitter.com/miconda</a> -- <a href="http://www.linkedin.com/in/miconda" moz-do-not-send="true">www.linkedin.com/in/miconda</a><o:p></o:p></pre>
<pre>Funding: <a href="https://www.paypal.me/dcmierla" moz-do-not-send="true">https://www.paypal.me/dcmierla</a><o:p></o:p></pre>
</div>
</blockquote>
<pre class="moz-signature" cols="72">--
Daniel-Constantin Mierla -- <a class="moz-txt-link-abbreviated" href="http://www.asipto.com">www.asipto.com</a>
<a class="moz-txt-link-abbreviated" href="http://www.twitter.com/miconda">www.twitter.com/miconda</a> -- <a class="moz-txt-link-abbreviated" href="http://www.linkedin.com/in/miconda">www.linkedin.com/in/miconda</a>
Funding: <a class="moz-txt-link-freetext" href="https://www.paypal.me/dcmierla">https://www.paypal.me/dcmierla</a></pre>
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