diff options
| author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-02 00:24:03 +0000 |
|---|---|---|
| committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-02 00:24:03 +0000 |
| commit | 0cddd4907260bb50181cfc5ba02b7ffc45f0ba1a (patch) | |
| tree | 77eb44c928e7d6d1d2fb0247d63d93d60a91fbcd /configs | |
| parent | d70a2fe5465c850ae29d4f61f97d7d3ccfb9b6ea (diff) | |
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: http://svn.digium.com/svn/asterisk/branches/1.4@53109 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
| -rw-r--r-- | configs/sip.conf.sample | 6 |
1 files changed, 6 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index 2948eebab..293eafa63 100644 --- a/configs/sip.conf.sample +++ b/configs/sip.conf.sample @@ -307,6 +307,12 @@ 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 |