[asterisk-users] Keeping call up without SIP (Asterisk) in the middle

[Janet]

I know this was discussed years ago - but I'm looking into whether things
have changed.  Imagine this scenario:

 

1.       Phone A call Phone B through Asterisk.  (A -- > Asterisk -- > B)

2.       All 3 devices have public IP addresses, and Asterisk is configured
for directmedia / reinvites.

3.       Phone A and B are having a successful call with direct RTP.

4.       Asterisk shutdowns down (pull the power) and the SIP connection
closes (maybe a FIN is sent, maybe not)

 

My questions are:

1.       Will he call drop?  

2.       Immediately or after some SIP packet times out?

3.       Is there a way to keep the call up without Asterisk/SIP?  (This was
discussed before and the practical answer was no)  

 

I'm curious if anything has changed. The only solution put forward years ago
was adding a proxy in front of Asterisk which redirects SIP between phone,
but that discussion had lots of negatives / debate.

 

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