6 Jan
2022
6 Jan
'22
1:33 a.m.
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(a)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(a)lists.kamailio.org> On Behalf Of Nauman
Sulaiman (SESSIONTALK)
Sent: Wednesday, January 5, 2022 3:13 PM
To: sr-users(a)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
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