One VoIP carrier will change their SIP server.
They informed us this information.
If connecting to other VoIP carrier from this VoIP carrier, there will problem may occur about RTP in case of some IP-PBX system.
Normaly for Incoming call during 183 with RTP, RTP sequence number increasing continuously correctly.
But after 200 OK, in case of some VoIP system it will big skip for RTP sequence number.
I mean it will increasing like 40000 40001 40002 during 183, but after 200 OK it will skip to like 50000.
According to VoIP carrier some IP-PBX will chek RTP sequence number and if sequence number will skip a lot, it will stop handle RTP as invalid RTP.
So it will become one way audio.
We concern that how about Asterisk?
Using Asterisk 18.