https://origsvn.digium.com/svn/asterisk/branches/1.4
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r78569 | file | 2007-08-08 10:51:01 -0300 (Wed, 08 Aug 2007) | 4 lines
(closes issue #10335)
Reported by: adamgundy
Update sip.conf to include another scenario where directrtpsetup will fail.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78570
65c4cc65-6c06-0410-ace0-
fbb531ad65f3
; the call directly with media peer-2-peer without re-invites.
; Will not work for video and cases where the callee sends
; RTP payloads and fmtp headers in the 200 OK that does not match the
- ; callers INVITE.
+ ; callers INVITE. This will also fail if canreinvite is enabled when
+ ; the device is actually behind NAT.
;canreinvite=nonat ; An additional option is to allow media path redirection
; (reinvite) but only when the peer where the media is being