How to connect Asterisk SIP Clients over Duel Layers of NAT
A common problem with SIP is it’s difficulty handling NAT(especially over multiple layers). A common symptom is no audio stream. The Asterisk CLI shows a successful call connection succeeding followed by an error in /var/log/asterisk/messages:
[Jun 14 17:16:26] WARNING[2967] chan_sip.c: Retransmission timeout reached on transmission 4e2a8586-3a91c1cf-e73f6b64@10.1.10.115 for seqno 2 (Critical Response) — See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
Packet timed out after 6399ms with no response