From: Olle Johansson Date: Tue, 17 Oct 2006 17:39:18 +0000 (+0000) Subject: Adding information about Marks direct-RTP hack to the docs... X-Git-Tag: 1.4.0-beta3~7 X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=590698e583fe7cd20dfef5823d3ff2bf72cd8748;p=thirdparty%2Fasterisk.git 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 b101b9058d..ef4fa2cda7 100644 --- 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