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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...
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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.
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If only existing phone gateways would start to support newer signaling protocols like Axolotl and Proteus with SRTP in the same channel.