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
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