[asterisk-users] SIP Disconnects from Network - Asterisk Does not hangup

[Kevin P. Fleming]

Kirill 'Big K' Katsnelson wrote:

> The caveat here is that it is perfectly normal NOT to transmit any RTP
> data in case of long silence. This is why the SIP timers were introduced
> in the first place: there is no correct way to detect when the client is
> going away, as no activity is a good session state.

That's only true when Asterisk tells the other endpoint that it is
allowed to use voice activity detection and silence suppression, which
at this point it does not do. In spite of that, there are many endpoints
that do it anyway, which then causes strange problems on calls,
including calls getting dropped if an RTP timeout is in use.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming at digium.com
Check us out at www.digium.com & www.asterisk.org