Hm. To use RTPEngine I need Kamailio, correct? Or can I use RTPEngine
just with Asterisk?
If I need Kamailio: Kamailio in the default configuration is behaving
differently than Asterisk in terms of the contents of the SIP header
fields that it sends by default to the carrier IP (please see my first
E-Mail in this thread where I compared the headers that Kamailio sends
vs. the headers that Asterisk sends). This breaks the connection to the
carrier so to speak (INVITE's not being accepted by the carrier). Thus,
when I replace Asterisk with Kamailio, I need to modify Kamailio's
config in a way that I get it to send exactly the same headers than
Asterisk would.
If I don't need Kamailio: That would be great, but I'm thinking the load
of just Asterisk + RTPEngine would be still higher than Kamailio +
RTPEngine. I don't need any of the Asterisk features really, just need
to forward and receive SIP packets, and I guess Kamailio is performing
much better than Asterisk at this task.
Am 22.09.2023 um 17:38 schrieb Alex Balashov via sr-users:
So, forgive the silly question, but why do you need to do anything except to
engage RTPEngine? Why rewrite any other SIP headers?
On Sep 22, 2023, at 5:11 AM, Markus via sr-users <[email protected]>
wrote:
Hi Alex,
I'm trying to replace the Asterisk box with an instance of Kamailio+RTPEngine
because the Asterisk box is heavily overloaded and calls that are passing
through this box are encountering packet loss. The idea behind it is that the
bundle of Kamailio+RTPEngine will be less CPU-intense than Asterisk and that
the machine this bundle runs on would be able to handle the current call load
without packet loss.
With "drop-in replacement" I meant that no changes on the upstream carrier side
can be made for the moment (they're slow), thus I'm having to use the IP of the Asterisk
box for the Kamailio+RTPengine bundle.
The purpose of the Asterisk box (and, once I got it to work, Kamailio+RTPEngine
will be the replacement) is to route SIP voice calls from several other
Asterisk boxes in the LAN to the carrier.
That single overloaded Asterisk box is the gateway to the carrier so to speak.
And its IP 2.2.2.2 which is authorized to send INVITE's towards the carrier
can't get changed on the carrier side for the moment.
Thanks :)
Markus
Am 22.09.2023 um 03:56 schrieb Alex Balashov:
Hi Markus,
Can you elaborate upon the way in which you are using Kamailio+RTPEngine "as a
drop-in replacement"? Drop-in replacement for what? Or that is to say, what are you
trying to accomplish here, functionally?
I have the suspicion that what you're doing is probably best accomplished in a
different and more straightforward way. :-)
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