[Asterisk-Dev] Security issue mumblings - SIP

[Olle E. Johansson]

Kevin P. Fleming wrote:
> Olle E. Johansson wrote:
> 
>> According to specs we have to start listening when we send an SDP and
>> are able to start sending audio when we get an SDP. I agree that the ACK
>> would be the time that the call "started" but that's not really
>> implemented. In Asterisk the call is UP when we get or send a 200 OK.
> 
> 
> That is intentional, and will not be changed. The same is true on PRI
> links, where we consider the call 'up' as soon as we send CONNECT,
> without waiting for the CONNECT ACKNOWLEDGE. This is proper behavior and
> is 'by design' :-)
> 
> Of course you are correct in saying that if the ACK is never received we
> will tear the call down, but during that interval that the call is up,
> the call will be 'billed'.

Sorry for not expressing myself better.
I fully agree with you that this is the correct way for billing, even
though the SIP dialog should not be considered established until we
receive an ACK.

/O