[asterisk-users] [asterisk-dev] How to get to 10.000 open calls

[Atis Lezdins]

# 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 <venefax at 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-bounces at lists.digium.com
> [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Tzafrir Cohen
> Sent: Wednesday, April 22, 2009 3:19 AM
> To: asterisk-dev at 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.cohen at xorcom.com
> +972-50-7952406           mailto:tzafrir.cohen at xorcom.com
> http://www.xorcom.com  iax:guest at 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,
atis at iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835