# moving to -users as this belongs there. It is a nice idea to run several Asterisk processes simultenously, it will defineately help with multithreading. However I would suggest trying less instances - that would perhaps give greater benefit, as Asterisk has it's own threading. For example 8 instances of Asterisk / 4 instances.. However, in this case - if You go for splitting everything up, You could just simply drop in more machines. I think it would be more cost-effective to have 8 machines with 2 cores each. and that would additionally provide better I/O performance.
Anyway, You can try throwing those calls and see how much can You get. As for directrtp=yes - i'm not sure what it does, but perhaps it's meant to be canreinvite=yes? Set it for each peer, and make sure You dial to peer, not to IP (as I recall - this didn't work globally) Regards, Atis On Wed, Apr 22, 2009 at 10:31 AM, Venefax <vene...@gmail.com> wrote: > Yes, I have the box. And I will get the calls next week. I was thinking to > use the Asterisk feature where you can start different Asterisk using -C > \path_to\config\file, and start 15 instances. But to be able to load balance > it is a nightmare, since many clients do not accept or follow redirects (SIP > 302 Moved). I am out of tricks, unless I setup another technology for load > balancing but then why not use the same (x) technology for everything? What > technology would that be that can handle 10.000 sip connections, not > touching the media? My Cisco 7301 would not scale so far out. > > -----Original Message----- > From: asterisk-dev-boun...@lists.digium.com > [mailto:asterisk-dev-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen > Sent: Wednesday, April 22, 2009 3:19 AM > To: asterisk-...@lists.digium.com > Subject: Re: [asterisk-dev] How to get to 10.000 open calls > > On Wed, Apr 22, 2009 at 02:48:11AM -0400, Venefax wrote: >> I am using 1.6.2 and directrtp=yes. I need to scale to 10.000 open calls > on >> a box with 1288 GB or RAM and 16 Cores. Is there any modification to the >> source code that would be obvious, any bottlenecks? I will never to >> transcoding and the media should, theoretically, flow outside. I have 15 > IP >> addresses already configured in the same box, on two different nics, to >> spread the interrupts. Is this a dream or will this work with some > tweaking? > > Do you have the system now? > > While it's most likely be a dream, identifying the current bottlenecks > might be useful :-) > > Just a few uneducated guesses of my own: > > * More than one IP per NIC won't help and only cause some administrative > issues > * I'm not sure how much the extra memory can help. I suspect htat if you > boot the system with mem=<whatever_needed_for_16GB> the results won't > differ greatly > * It would also be interesting to see how the results scale with various > values numbers of cores. This is again something you can set at boot > (numcpus=N). I wonder just how far from linear it will be. > > -- > Tzafrir Cohen > icq#16849755 jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > -- Atis Lezdins, VoIP Project Manager / Developer, IQ Labs Inc, a...@iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
