[SR-Users] Kamailio call flows with Asterisk

[Yuriy Gorlichenko]

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 at 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 at lists.kamailio.org> On Behalf Of Nauman
> Sulaiman (SESSIONTALK)
> Sent: Wednesday, January 5, 2022 3:13 PM
> To: sr-users at 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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