rene_mobile’s avatarrene_mobile’s Twitter Archive—№ 1,811

    1. After too many hours trial&error and debugging in the last two months, I am now officially giving up on SIP over NAT. Broken by design...
  1. …in reply to @rene_mobile
    I may still check if @asteriskpbx SIP-TLS/SRTP works over NAT (@freeswitch doesn't). Plan B is IAX2 to mobile clients with bridge to SIP.