[Asterisk-Users] Dropped SIP connections never being closed?

[Olivier]

2006/5/30, Kevin P. Fleming <kpfleming at digium.com>:
>
> You can use 'rtptimeout' to make Asterisk notice when the RTP stream has
> stopped and then drop the call,


 Could that be done at network level (with some kind of network monitoring)
?
Regards
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