Asterisk sip rtptimeout


Terminate call if 60 seconds of no RTP activity when we're not on hold
Added in June 2004 to CVS-HEAD



See also

Note from MarkSter's writing on bugtrack :

However, I've added an option called "rtptimeout" which can be used to
automatically hangup the call if no RTP traffic is received within that
number of seconds. It can be specified globally or on a per-peer basis.
Please be aware that if your ata ignores our request to disable silence
suppression (e.g. you have an ATA186) then this could cause it to
terminate the call when one party is silent for that number of seconds.

Further, if the ATA or phone puts us on hold, then explicitly they are no
longer sending RTP traffic, so the rtptimeout setting is ignored.
However, an alternative setting, rtpholdtimeout, can be specified (it
SHOULD be greater than rtptimeout). That places an ABSOLUTE TIMEOUT on
how long we will allow ourselves to be placed on hold.

These aren't beautiful, but they're at least some sort of a work around
for yet another architectural limitation of SIP. As you may have already
guessed, IAX does not have this problem.

Go back to Asterisk config sip.conf
Created by: JustRumours, Last modification: Mon 27 of Sep, 2004 (07:45 UTC) by dercol
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