How to organize the correct operation of Asterisk on several network interfaces?

Question

Asterisk

I registered SRV records in NS of the format
_sip._udp [1.1.1.1] priority 0 port 5061
_sip._udp [2.2.2.2] priority 1 port 5061
_sip._udp [3.3.3.3] priority 2 port 5061
The softphone normally ate the DNS record and if forcibly bring down the line 1.1.1.1:5061, then it connects without problems to another channel. Mission accomplished. But then another priest appeared. The IP address 1.1.1.1 is registered in the master and connecting via CHAN_SIP in the second and third channels leads to no sound during calls (and, apparently, the interlocutor does not hear anything).
Tried: As soon as you change Extrenal IP in Aster = all chiki-farts. I also tried changing codecs, no effect.
On the CHAIN_PJSIP driver - everything is exactly in this situation. It works without changing the IP address in the system and for clients it is almost seamless switching to another channel. But unfortunately, I specially compiled Asterisk with an OPUS codec for conversations even with Edge from a mobile phone (by the way, it works fine), but it does not work with CHAIN_PJSIP (there is no patch at the moment)
PS: I will not be able to correctly track which the channel collapsed to change the data in the aster or to which a specific person connected. There is an emergency 4th and 5th entry in the SRV, which lead to servers in Europe and China, where there is end-to-end routing through the VPN tunnel to Aster. Well, you yourself understand - the Chinese firewall, sanctions and, in general, an alarming time.

Answer 1

nf_nat_sip seems to be able to change the address in rtp packets

Answer 2

I advise you to make a VPS somewhere in a stable place and resolve issues with channels using routing tools in tunnels - the same bonding them. On the VPS, connect the wires.