[SR-Users] Call in Progress Recovery in Redundancy

Thu Aug 6 16:26:00 CEST 2020

Hello Daniel,

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.

I want to know two more things : 

1.	Can we built Re-Homing around Kamailio (Move call from Kamailio to
Freeswitch). Is there any possibility of doing it.
2.	How can capture sip-info from Kamailio to Freeswitch. Means DTMP
pressed.

Thanks in advance.

Amit Sharma

From: Daniel-Constantin Mierla <miconda at gmail.com> 
Sent: Wednesday, August 5, 2020 6:33 PM
To: Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>;
amitsharma at coraltele.com
Subject: Re: [SR-Users] Call in Progress Recovery in Redundancy

Hello,

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.

My answer there related to htable was about the ability to route CANCEL
requests where the INVITE was forwarded.

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.

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.

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.

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.

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.

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.

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.

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.

Cheers,
Daniel

On 05.08.20 12:36, amitsharma at coraltele.com
<mailto:amitsharma at coraltele.com>  wrote:

Dear Daniel/Team,

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
https://wazo-platform.org/blog/kamailio-ha-dispatcher-and-dmq and able to
recover Call in progress within 2-3 nodes.

1.	My one question is that either this approach will work in production
or not.
2.	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.

Please suggest me the best way possible to achieve this.

Thanks & Regards,

Amit Sharma

(Sr. Team Leader)

(An ISO 9001:2008 company)

Mobile:  <tel:9891612004> tel:9891612004

PH:  +91 120 2595870

Ext.:  <tel:870> tel:870

Email :  <mailto:amitsharma at coraltele.com> amitsharma at coraltele.com

Web :  <blocked::http://www.coraltele.com> www.coraltele.com

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-- 
Daniel-Constantin Mierla -- www.asipto.com <http://www.asipto.com> 
www.twitter.com/miconda <http://www.twitter.com/miconda>  --
www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> 
Funding: https://www.paypal.me/dcmierla
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