[asterisk-commits] oej: branch oej/earlyrtpfix r49087 - /team/oej/earlyrtpfix/configs/

[asterisk-commits at lists.digium.com]

Author: oej
Date: Sun Dec 31 05:11:54 2006
New Revision: 49087

URL: http://svn.digium.com/view/asterisk?view=rev&rev=49087
Log:
Add docAdd docss

Modified:
    team/oej/earlyrtpfix/configs/sip.conf.sample

Modified: team/oej/earlyrtpfix/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/team/oej/earlyrtpfix/configs/sip.conf.sample?view=diff&rev=49087&r1=49086&r2=49087
==============================================================================
--- team/oej/earlyrtpfix/configs/sip.conf.sample (original)
+++ team/oej/earlyrtpfix/configs/sip.conf.sample Sun Dec 31 05:11:54 2006
@@ -303,6 +303,12 @@
 				; at call setup (a new feature in 1.4 - setting up the
 				; call directly between the endpoints instead of sending
 				; a re-INVITE).
+
+;directrtpsetup=no		; Experimental feature: In 1.4, there's an experimental
+				; feature for direct call setup between phones, where the
+				; media is setup peer2peer at call setup instead of using
+				; re-invites. This is known to have issues and is disabled
+				; by default in the 1.4 release.
 
 ;canreinvite=nonat		; An additional option is to allow media path redirection
 				; (reinvite) but only when the peer where the media is being