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author | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-10-17 17:39:18 +0000 |
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committer | oej <oej@f38db490-d61c-443f-a65b-d21fe96a405b> | 2006-10-17 17:39:18 +0000 |
commit | 7268d40ab9da9f123c885756ab7db48817efe657 (patch) | |
tree | 826db3030db664bca56a04fc194d12b03b9017a9 /configs | |
parent | 5a4a0f68af54b4ff6e84a5049b793764f3b9df95 (diff) |
Adding information about Marks direct-RTP hack to the docs...
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@45329 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'configs')
-rw-r--r-- | configs/sip.conf.sample | 5 |
1 files changed, 5 insertions, 0 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample index b101b9058..ef4fa2cda 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 |