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@gmail.com>
Sent: Wednesday, August 5, 2020 6:33 PM
To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>; amitsharma@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@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)

 

 Copy of 34643416.jpg


(An ISO 9001:2008 company)

 

Mobile: tel:9891612004

PH:  +91 120 2595870

Ext.: tel:870

Email : amitsharma@coraltele.com

Web : www.coraltele.com

 

https://docs.google.com/uc?export=download&id=0BwYYBqN87-tfM2JiYmhMSFdSZkk&revid=0BwYYBqN87-tfZW5iNExpazJqc01WUDZpOWFXd09SakhuUWJRPQ

 



_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla