[Asterisk-Dev] Changes in chan_sip: rtptimeout

[John Todd]

I really like the new changes I see in CVS for chan_sip for 
rtptimeout and rtpholdtimeout.  Does anyone know if these are 
bi-directional or unidirectional?

Let's say I have two SIP devices, UA-A and UA-B.  They make a call, 
and they are connected through Asterisk and the RTP stream goes 
through Asterisk.  UA-A then gets unplugged from it's ethernet.  UA-B 
will still continue to send media to Asterisk->UA-A, but UA-A isn't 
there.

Does rtptimeout look at only one direction of the RTP flow, or both? 
In other words, is the timer reset with _any_ RTP packets in the 
call, or does the clock start ticking when only one side stops 
sending?

JT