Clarify RTP timers. Sorry, grandma.

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

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