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-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 eb5334693..6557cd551 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -248,6 +248,12 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; route = Assume NAT, don't send rport
; (work around more UNIDEN bugs)
+;----------------------------------- MEDIA HANDLING --------------------------------
+; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
+; no reason for Asterisk to stay in the media path, the media will be redirected.
+; This does not really work with in the case where Asterisk is outside and have
+; clients on the inside of a NAT. In that case, you want to set canreinvite=nonat
+;
;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