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-rw-r--r--configs/sip.conf.sample18
1 files changed, 13 insertions, 5 deletions
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 178e29b00..e85e0c778 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -223,12 +223,20 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;canreinvite=yes ; Asterisk by default tries to redirect the
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
- ; support this (especially if one of them is
- ; behind a NAT).
+ ; support this (especially if one of them is behind a NAT).
; The default setting is YES. If you have all clients
- ; behind a NAT, or for some other reason wants
- ; Asterisk to stay in the audio path,
- ; you may want to turn this off
+ ; behind a NAT, or for some other reason wants Asterisk to
+ ; stay in the audio path, you may want to turn this off.
+
+;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
+ ; determine it based on the apparent IP address the media
+ ; arrives from).
+
+;canreinvite=update ; Yet a third option... use UPDATE for media path redirection,
+ ; instead of INVITE. This can be combined with 'nonat', as
+ ; 'canreinvite=update,nonat'. It implies 'yes'.
;----------------------------------------- REALTIME SUPPORT ------------------------
; For additional information on ARA, the Asterisk Realtime Architecture,