Clarify RTP timers. Sorry, grandma.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48105 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 7f9649a38a4b53df371c1abeebf7f1aac9894899..cbd892de8e967baa85ce6a5d9f70986a39bc7b4f 100644 (file)
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -90,9 +90,11 @@ srvlookup=yes                        ; Enable DNS SRV lookups on outbound calls
 ;language=en                   ; Default language setting for all users/peers
                                ; This may also be set for individual users/peers
 ;relaxdtmf=yes                 ; Relax dtmf handling
-;rtptimeout=60                 ; Terminate call if 60 seconds of no RTP activity
-                               ; when we're not on hold
-;rtpholdtimeout=300            ; Terminate call if 300 seconds of no RTP activity
+;rtptimeout=60                 ; Terminate call if 60 seconds of no RTP or RTCP activity
+                               ; when we're not on hold. This is to be able to hangup
+                               ; a call in the case of a phone disappearing from the net,
+                               ; like a powerloss or grandma tripping over a cable.
+;rtpholdtimeout=300            ; Terminate call if 300 seconds of no RTP or RTCP activity
                                ; when we're on hold (must be > rtptimeout)
 ;trustrpid = no                        ; If Remote-Party-ID should be trusted
 ;sendrpid = yes                        ; If Remote-Party-ID should be sent