@gtcnet wrote:
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
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