This might be tricky to move fast  registrations from the asterisk to kamailio if asterisk for example,  handles a lot of logic and used as presence service and queues manager.
The other approach  might work here: to go with kamailio as a load balancer between asterisks but keep registrations of all users across the asterisks first till migration of other functionality will be done. Usable scenario in that case - to share registration of the user between multiple asterisks using kamailio.

However regarding trunks: kamailio is a good candidate to be used as entrypoint for trunks and as main point for outgoing calls from the trunks. It can handle as IP2IP based relationship as registration based ( see UAC module ).

If your providers allow you to send RTP traffic from specific ips but not from ip of the endpoint SIP message came from - you can run rtp directly from the asterisks.

*Annoying mode on
Asterisk is not an RTP proxy in any case
*Annoying mode off

On Wed, 5 Jan 2022, 17:41 Henning Westerholt, <hw@gilawa.com> wrote:
Hello,

there are of course many options depending on your requirements etc..

But if your infrastructure has grown over a certain size, then common architectures are:

- using kamailio of load balancer in front of asterisk for security/scalability
- using kamailio additionally to handle also certain SIP applications, like registration handling

Again, generalization - Kamailio should handle the registration more scalable and more reliable as asterisk.

Cheers,

Henning

--
Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com

-----Original Message-----
From: sr-users <sr-users-bounces@lists.kamailio.org> On Behalf Of Nauman Sulaiman (SESSIONTALK)
Sent: Wednesday, January 5, 2022 3:13 PM
To: sr-users@lists.kamailio.org
Subject: [SR-Users] Kamailio call flows with Asterisk

Hi,

We are using Asterisk as a PBX with users directly registered to Asterisk and Asterisk registering to SIP trunks. We are now looking to put Kamailio in front of Asterisk to handle SIP registrations from the SIP clients.

In a ‘typical’ architecture should we keep the SIP trunk registrations on Asterisk or is Kamailio used for this? We want to keep Asterisk as the RTP proxy so we don’t want a call setup by Kamailio with RTP then going direct between user agents.

Regards
Nauman


__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
  * sr-users@lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the sender!
Edit mailing list options or unsubscribe:
  * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
__________________________________________________________
Kamailio - Users Mailing List - Non Commercial Discussions
  * sr-users@lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the sender!
Edit mailing list options or unsubscribe:
  * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users