Jul
03
2017
0

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

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