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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-- 
Alex Balashov
Principal Consultant
Evariste Systems LLC
Web: https://evaristesys.com
Tel: +1-706-510-6800

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