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authorJoshua Colp <jcolp@digium.com>2009-06-26 20:19:49 +0000
committerJoshua Colp <jcolp@digium.com>2009-06-26 20:19:49 +0000
commit48f7381af038934ab14cffa25ccbf6e86d5f83ef (patch)
tree856b129a30dff0b3dfecb4176120bbc9c0abe515 /configs
parent9c6f164227c75e58cea271848476cb31fef79121 (diff)
Fix the 'nat' option to actually do RFC3581 as expected and extend the configurable values for finer control.
(closes issue #8855) Reported by: mikma Tested by: klaus3000, file git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203735 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'configs')
-rw-r--r--configs/sip.conf.sample9
1 files changed, 4 insertions, 5 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 245f87145..3aadc633d 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -635,11 +635,10 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; However, this is only useful if the external traffic can reach us.
; The following settings are allowed (both globally and in individual sections):
;
-; nat = no ; default. Use NAT mode only according to RFC3581 (;rport)
-; nat = yes ; Always ignore info and assume NAT
-; nat = never ; Never attempt NAT mode or RFC3581 support
-; nat = route ; route = Assume NAT, don't send rport
-; ; (work around more UNIDEN bugs)
+; nat = no ; Default. Use rport if the remote side says to use it.
+; nat = force_rport ; Force rport to always be on.
+; nat = yes ; Force rport to always be on and perform symmetric RTP.
+; nat = comedia ; Use rport if the remote side says to use it and perform symmetric RTP.
;----------------------------------- MEDIA HANDLING --------------------------------
; By default, Asterisk tries to re-invite the audio to an optimal path. If there's