High availability

Is it possible that when an Asterisk server crashes, that the SIP and RTP traffic be rerouted in a way that any calls on that asterisk server not actually be dropped, but remain active? Basically, is it possible to prevent a user’s call from dropping by transfering it to another server? I haven’t found this question posted anywhere so I hope it’s not a duplicate.

Thanks for your help.

directmedia=yes (unlikely to be supported by service providers) can provide some resillience for RTP. My guess is that any mechanism to provide resillience for SIP would be the part most likely to crash!

Thanks David. I failed to mention that we frequently do call monitoring and recording on our servers. So are you saying that it’s possible if we are not in the media stream? Also, the second part of your reply… Are you saying that any sort of mechanism to handle the sip traffic would be a problem? Sorry if these are stupid questions. I’m still learning a lot about this stuff. :smile:

I was saying that software that tries to synchronise the state of a master and standby system at SIP traffic rates is probably a more likely point of failure than the software it is trying to protect.

Synchronising state at the RTP traffic rate, if you actually need to stay in the stream, would be even worse!

Gotcha. So is there a way to actually redirect the traffic and not actually keep maintaining state? Or does that make it impossible for the second machine to take over?

Asterisk SCF was built to provide this kind of capability where Asterisk can’t do it.

I’ll take a look. Thanks!

Thanks for the info. It seems promising. It looks to be a bit young for any serious use. I’ll keep my eye on the project.
So am I to believe that the best possible goal would be to have quick fail-over to another Asterisk server and just live with the calls that would be dropped?

That’s the best you can hope for with Asterisk today.