The DNS approach does not handle single or multiple system failures, only very elementary load balancing over a lengthy period of time.
------------------------ > You may want to consider a simpler aproach, why don't you balance the load > via DNS? > If you put in a zone file various A records for the same machine, but with > different > IP's, BIND will catch the trick and send a different IP (from the pool yo > defined) each > time a DNS request arrives. That's a simple way of doing that, it will > definively work > for termination, but you may have to think more who to cope with origiation > (outgoing > calls), since different clients will be connected to different servers. > > > --- [EMAIL PROTECTED] wrote: > > We use it on our web and mail server to load ballance across multiple > hosts. The way we have it configured > it will maintain a session for 15 minutes between a client and a > specific server. So long as you have > qualify=yes in your configuration files, each client will continue to > talk to the one server until they are turned off/ > deactivated for at least 15 minutes (or whatever time period you > configure into it). I've not tested LVS with > Asterisk, but it may be the right direction for you to take. > > Cheers, > -Shaun > > Matthew Boehm wrote: > > >I've read several other emails and pages on the wiki but none give any > >deffinate answers. if you have 20 asterisk servers each with 4 pri's, all > >running RealTime Extensions and RealTime SIPBuddies from the same MySQL > >server, what prevents you from putting all 20 servers behind a single load > >balancer? That way all of your UA's can use the same IP to register to; vs > >maintaining which customer is assigned to which machine. > > > >perhaps its just that i am not that familiar with load balancers. i was > >under the impression that a load balancer could/would send each recieved > >packet to a different server. > >this doesn't matter in the case of register requests since all asterisk > >boxes share same SIP registry database. > > > >but what about invite requests and the rtp stream? you would have a majorly > >broken conversation if each packet in the rtp stream went to a different > >asterisk box. > > > >or are load balancers SIP aware? or is there some sort of session control > >that the balancer is aware of and will send all packets in a "sip session" > >to the same asterisk box? > > > >and then what about meet me conferences? if 10 UA's all dial a conference > >DID number and all 10 get balanced to 10 different servers then they are all > >sitting in seperate rooms right? > > > >hints, opinions, facts...all welcome and appreciated. > > > >-Matthew > > > >_______________________________________________ > >Asterisk-Users mailing list > >[email protected] > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > __________________________________ > Do you Yahoo!? > All your favorites on one personal page � Try My Yahoo! > http://my.yahoo.com > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ---------------End of Original Message----------------- _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
