PJSIP NAT AUDIO transport=transport-udp

I have a server and a remote endpoint both behind NAT. I am having audio issues and see that the sdp message is showing the phone ip not the server. In reading the asterisk documentation it suggests having the “transport=transport-udp” enabled in the endpoint context. I didn’t see that option for the endpoint. Would anyone have any suggestions?

;===============ENDPOINT BEHIND NAT OR FIREWALL===============================
348 ;
349 ; This example assumes your transport is configured with a public IP and the
350 ; endpoint itself is behind NAT and maybe a firewall, rather than having
351 ; Asterisk behind NAT. For the sake of simplicity, we’ll assume a typical
352 ; VOIP phone. The most important settings to configure are:
353 ;
354 ; * direct_media, to ensure Asterisk stays in the media path
355 ; * rtp_symmetric and force_rport options to help the far-end NAT/firewall
356 ;
357 ; Depending on the settings of your remote SIP device or NAT/firewall device
358 ; you may have to experiment with a combination of these settings.
359 ;
360 ; If both Asterisk and the remote phones are a behind NAT/firewall then you’ll
361 ; have to make sure to use a transport with appropriate settings (as in the
362 ; transport-udp-nat example).
363 ;
364 ;[6002]
365 ;type=endpoint
366 ;transport=transport-udp
367 ;context=from-internal
368 ;disallow=all
369 ;allow=ulaw
370 ;auth=6002
371 ;aors=6002
372 ;direct_media=no
373 ;rtp_symmetric=yes
374 ;force_rport=yes
375 ;rewrite_contact=yes ; necessary if endpoint does not know/register public ip:port
376 ;ice_support=yes ;This is specific to clients that support NAT traversal
377 ;for media via ICE,STUN,TURN. See the wiki at:
378 ;https://wiki.asterisk.org/wiki/x/D4FHAQ
379 ;for a deeper explanation of this topic.
380
381 ;[6002]
382 ;type=auth
383 ;auth_type=userpass
384 ;password=6002
385 ;username=6002
386
387 ;[6002]
388 ;type=aor
389 ;max_contacts=2
390
391

Standard config for pjsip extensions works for this arrangement, you don’t have to delve into conf files. What version of Asterisk, if not at 13.17.1, upgrade to current, there are recently fixed bugs with remote pjsip extensions.

I am using freepbx 14 and asterisk 14.6.1.

Everything was working fine till I upgraded one of the modules (ring groups). I then got one way audio. Recovered from a backup image before the module update and working again. I now have the broken image on another machine and I debugged the sip packet sdp and noticed that the endpoint ip is offered instead of the asterisk asterisk ip. This is the current state and can be duplicated. Thanks for you help.