23 Sep
2023
23 Sep
'23
4:13 p.m.
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
<sr-users(a)lists.kamailio.org> 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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