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

[Kirill 'Big K' Katsnelson]

On 100222 1818, Kevin P. Fleming wrote:
> 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, 

Oh yes, I've seen these problems first person, mostly manifesting 
themselves as dropped syllables after a period of silence if not 
complete loss of a call, but I assumed it was not a negotiated option 
but rather left to unilateral decision of an endpoint. Thank you for the 
correction!

  -kkm
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