Using *RT for HA purposes was: [Asterisk-Users]
RealtimeMultipleAsterisk boxes, iaxusers
Douglas Garstang
dgarstang at oneeighty.com
Thu Jan 5 23:52:57 MST 2006
The dispatcher module in OpenSER can load balance calls based on a hash of the SIP call-id. Supposedly the latest version even supports failover. Ooooo, fancy.
Doug.
-----Original Message-----
From: tijmen van den brink [mailto:tijmen.vandenbrink at gmail.com]
Sent: Wed 1/4/2006 2:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: Re: Using *RT for HA purposes was: [Asterisk-Users] RealtimeMultipleAsterisk boxes, iaxusers
I did some research about Asterisk and High Availability and some sort of load balancing. The High Availability issue isnn't much of a problem. I did it with heartbeat en realtime. But the load balancing issue is realy a problem. You want a load balancer to make decisions based on call ID. The call ID is stored in the SIP header (layer 7) and for all I know there are only a few load balancers that can make decisions based on this layer and those load balancers are not SIP aware. So for now I don't think load balancing with *servers could be easily achieved.
On 1/4/06, Kevin P. Fleming <kpfleming at digium.com> wrote:
Asterisk wrote:
> In my case I would be using DNS round robin. So a UA would only be
> registering to one * server at a time. So wouldn't in fact be an
> active/passive?
No. You have said that you want the _other_ servers to be aware of that
phone's registration and be able to deliver calls to it directly. That
will not work.
If you want the other servers to send calls to that phone through the
server it registered with, then yes, that can easily be done.
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Tijmen van den Brink
Wilhelminaweg 46
3441 XC Woerden
Tel: 0642233831
MSN: tijmenvandenbrink at hotmail.com
Skype: tijmenvdbrink at skype.com
SIP:697116 at fwd.pulver.com
More information about the asterisk-users
mailing list