Hello,

to use rtpengine you need some kind of server like Kamailio, it does not work 
with asterisk.

Many people use Kamailio for carrier interconnection, in the end it is (as you 
already stated) just a matter of the correct configuration.

Cheers,

Henning

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

> -----Original Message-----
> From: Markus via sr-users <sr-users@lists.kamailio.org>
> Sent: Samstag, 23. September 2023 15:13
> To: Kamailio (SER) - Users Mailing List <sr-users@lists.kamailio.org>
> Cc: Markus <unive...@truemetal.org>
> Subject: [SR-Users] Re: Modifying SDP as drop-in replacement for overloaded
> Asterisk box - looking for help/paid consulting fast
> 
> 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-
> us...@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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