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authorOlle Johansson <oej@edvina.net>2007-02-02 00:26:25 +0000
committerOlle Johansson <oej@edvina.net>2007-02-02 00:26:25 +0000
commitcfe66e6b26931c9cd92f2f4f1d36694b1b8baad6 (patch)
treebfdbfed71e156ca94b0c7eeaad4514a52cd504c8 /configs/sip.conf.sample
parent44a9af35761c45010dc320c4786d683bdadaefb3 (diff)
Patch based on this patch with small changes for trunk...
Merged revisions 53109 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4 lines Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps stuff. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'configs/sip.conf.sample')
-rw-r--r--configs/sip.conf.sample6
1 files changed, 6 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 83073996d..df22910a9 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -333,6 +333,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; call directly between the endpoints instead of sending
; a re-INVITE).
+;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
+ ; 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.
+
;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