[asterisk-dev] SIP session-timers: concept, discussion

[John Todd]

At 2:29 PM +0800 2007/7/18, Dinesh Nair wrote:
>On Tue, 17 Jul 2007 16:20:27 -0700, John Todd wrote:
>
>>  A worse condition arises when a network
>>  becomes disconnected, and particularly stupid equipment doesn't
>>  notice that RTP stops in one direction(1).  (This brings up an
>
>in 1.2.22, chan_sip.so has a rtptimeout and rtpholdtimeout parameter in
>sip.conf which hangsup the call if the timeout expires and there have been
>no rtp packets in the period. doesnt this fit the bill for what you're
>suggesting ?


I'll quote myself from the footnote/subscript in the message I sent 
to the list:

>(1) So why not just use the RTP detection routines in Asterisk?  Two
>reasons come to mind immediately, but there may be more.  Firstly:
>often, RTP is not travelling through the Asterisk server that is
>responsible for the call setup, and forcing RTP through that server
>may be impractical or impossible.  Secondly, Session-Timers (can)
>work on the equipment at both ends of a call so any disconnection
>[can|will] cause both ends to hang up.  The RTP timers in Asterisk
>are great for closing out the call on the Asterisk system, but in the
>case of a network failure (without Session-Timers) it may be the case
>that the non-Asterisk side may continue the call "forever", which is
>typically a big problem when there are charges for call minutes.


JT