Adding information about Marks direct-RTP hack to the docs...

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45329 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index b101b9058d5fbaae6668e87a8d7010f1e41f7a27..ef4fa2cda71db0033fce8678a10c608a28002a0c 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -256,6 +256,11 @@ srvlookup=yes                      ; Enable DNS SRV lookups on outbound calls
                                ; behind a NAT, or for some other reason wants Asterisk to
                                ; stay in the audio path, you may want to turn this off.
 
+                               ; In Asterisk 1.4 this setting also affect direct RTP
+                               ; at call setup (a new feature in 1.4 - setting up the
+                               ; call directly between the endpoints instead of sending
+                               ; a re-INVITE).
+
 ;canreinvite=nonat             ; An additional option is to allow media path redirection
                                ; (reinvite) but only when the peer where the media is being
                                ; sent is known to not be behind a NAT (as the RTP core can