[asterisk-dev] Regarding SIP performance
Steve Langstaff
steve.langstaff at citel.com
Fri Oct 13 04:31:10 MST 2006
This is just off the top of my head, so excuse me if I am talking
rubbish, but perhaps the canreinvite checking could be done *before* the
channels to the endpoints have been established, so that a call that
will be 'reinvited' does not go though the RTP setup/cleardown mechanism
on Asterisk.
-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Chirag
Vaishnav
Sent: 13 October 2006 12:24
To: asterisk-dev at lists.digium.com
Subject: [asterisk-dev] Regarding SIP performance
Thanks. But I am looking for imrovement in code. Can we really do
something to make it able to handle more number of calls? If we can do
that we can get better performance on any machine. Like using sun spark
you can fire 300 call/sec then you should be able to do much more.
May be asterisk gives low performance because it is made to create
channel between to different protocal and it handle RTP/RTCP also. If we
think about sip to sip and peer to peer only can we do something to
improve performance.
I know there are softweres like OPEN SER to do so. but I have lot of
reason to use asterisk. So... help me. atleast give me some idea.
- Chirag
>From: Bob Atkins <bob at digilink.net>
>Reply-To: Asterisk Developers Mailing List
><asterisk-dev at lists.digium.com>
>To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
>Subject: Re: [asterisk-dev] Regarding SIP performance
>Date: Fri, 13 Oct 2006 02:15:58 -0700
>
>
>
>Chirag Vaishnav wrote:
>
>>
>>Hi !
>>
>>I am chirag from india and I am working with SIP part of asterisk.
>>Currently I am doing testing with SIPP tool. Now the thing is that
>>asterisk is not able to handle more then 550 concurrent calls with
>>call rate 100 call/sec (on my amd 64bit dual core machine with 2 gb
>>ram) which is to less. I am doing this with canreinvite mode so there
>>is no issue of RTP traffic. I just want to astablish peer to peer call
>>thru asterisk. As i have said doing 550 call with 100 call/sec ( I
>>hang up calls after some time so this is including all invite and
>>hangup related massages) it consume full processing power and
>>retransmission starts if i increase number of calls.
>>
>>If i keep firing 100 calls/sec without hangup retrans starts after
>>3200 calls.
>>
>>Can any one tell me why it is so? is there any way to increase number
>>of concurrent calls with same or more fire rate?
>
>Yes - run Asterisk under Solaris on a Sun Sparc system. If you want
>serious performance (>300 calls/sec) and reliability, Sun's Sparc
>systems are the way to go. Something like a dual 1GHz SunFire 280r or
>perhaps an older 10 processor E4500 that can be expanded to 14
>processors and has the ability to hot swap processors and memory
>/_*without*_/ being re-booted! Running asterisk on Solaris/Sun Sparc
>systems will outperform any Intel platform by several multiples. Before
>anyone brings up price difference - the 280r is 2 generations old and
>the E4500 is 3 generations old and both machines can be easily found
>for <$3000 and will outperform the latest Intel/AMD platforms costing
considerably more.
>
>Unfortunately, it seems that the predominate concentration of effort on
>Asterisk is for an Intel platform which, in my opinion and 15+ years of
>experience providing commercial services simply do not standup to the
>performance and reliability of Sun Sparc systems. And, from a business
>standpoint Sun Sparc systems provide more than 3 times the ROI of any
>Intel platform.
>
>And no - I don't work for Sun nor do I have any interest in selling
>hardware. I am just passing on our own experience. We are running a
>commercial production hosted PBX system using Asterisk on Sun Sparc
>systems very successfully.
>
>--
>*Bob Atkins * /President/CEO/
>
>*DigiLink, Inc. <http://www.digilink.net>* Business Inter-net-working
>*/The Cure for the Common ISP!/*
>
>
>
>Phone: (310) 577-9450
>Fax: (310) 577-3360
>eMail: bob at digilink.net
>
>
>
><< bob.vcf >>
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