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.