diff options
| author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-02 00:26:25 +0000 |
|---|---|---|
| committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2007-02-02 00:26:25 +0000 |
| commit | 9cee6624de380378c8689d4354807c101e67f76c (patch) | |
| tree | bfdbfed71e156ca94b0c7eeaad4514a52cd504c8 /configs/sip.conf.sample | |
| parent | b638ef0a340dcbc851cacc052df4514254ce9628 (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.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@53110 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs/sip.conf.sample')
| -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 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 |