[Asterisk-Users] Asterisk SIP architecture question

[Olle E. Johansson]

Matt Riddell wrote:
> Kevin P. Fleming wrote:
> 
>>Matt Riddell wrote:
>>
>>
>>>So how does Asterisk know that the media stream has been disconnected
>>>between
>>>the two remote hosts?
>>
>>It doesn't... nor does any other SIP softswitch. See my other reply for
>>a possible solution.
> 
> 
> I agree that you could code a fix, but saying my advice is bogus because you
> could code a fix for Asterisk to avoid it is slightly wrong.
> 
> The fact remains, if you need *very* accurate cdr's then you either don't do
> canreinvite=yes for the peer or you code something so that Asterisk notices
> that the rtp has stopped.  The fact remains that without these, the most
> accurate CDR is going to come from the provider.
> 

If the audio goes through asterisk without re-invites, you could use the
rtptimeouts to detect a dead phone.

/O