I have two identical servers I am doing some call volume testing on with SIPp,(we’ll call them Ast1 & Ast2). both servers have an identical iptables config, no NAT between and are on the same LAN, as well as similiar sip.conf/extensions.conf
Ast1 can call Ast2 without issue at high call volumes, lets say 700 concurrent calls for 1-2 minutes with RTP added. no issues
But if I call Ast1 from Ast2 I start to see SIP retransmissions and about 50-90% call failure on SIPp. It looks like on the Asterisk console though, the calls roll through the dialplan just fine. This issue only presents it self if SIPp is placing multiple concurrent calls. if I make 1 call, 10 seconds long for 10 total calls. no issues. but if I make 10 concurrent calls 10 seconds long 10 total calls, I am looking at 1 successful call and 9 failed. with multiple Invites retransmitted.
I have read retransmissions are a product of NAT issues or firewall issues. There is no NAT in use here and have checked the firewall configs. Both servers have the same settings at this point. Why would I see failures only coming from one way?
Thank you for any input