[asterisk-users] Asterisk 1.2.24 simultaneous call limits.

Fri Sep 21 16:29:02 CDT 2007

The errors are gone after I set the ulimit -n to 32786. The -x is
unlimited. 

I found that AMI is very inefficient. During the time of issue a lot of
AMI command, sip calls can't go into the asterisk box. I don't know if
you guys have the same experience.

________________________________

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James
Texter
Sent: Friday, September 21, 2007 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.2.24 simultaneous call limits.

What do you have ulimit -n and ulimit -x set to?

Thanks,

James Texter

On Fri, 2007-09-21 at 08:51 -0400, Wai Wu wrote: 

	I am not so sure if the interrupts has any thing to do with it.
I run some more test just now and I am getting these error on the
console of the call receiving machine. All it does is wait for 45
seconds. I think there is more can be done on the Linux configuration,
but I just don't know what.

	Sep 21 08:42:30 WARNING[22820]: channel.c:565 ast_channel_alloc:
Channel allocation failed: Can't create alert pipe!
	Sep 21 08:42:30 WARNING[22820]: chan_sip.c:2797 sip_new: Unable
to allocate SIP channel structure
	Sep 21 08:42:30 NOTICE[22820]: chan_sip.c:10843
handle_request_invite: Unable to create/find channel
	Sep 21 08:43:03 WARNING[22820]: channel.c:565 ast_channel_alloc:
Channel allocation failed: Can't create alert pipe!
	Sep 21 08:43:03 WARNING[22820]: chan_sip.c:2797 sip_new: Unable
to allocate SIP channel structure
	Sep 21 08:43:03 NOTICE[22820]: chan_sip.c:10843
handle_request_invite: Unable to create/find channel

	-----Original Message-----
	From: asterisk-users-bounces at lists.digium.com on behalf of
Gordon Henderson
	Sent: Fri 9/21/2007 3:40 AM
	To: Asterisk Users Mailing List - Non-Commercial Discussion
	Subject: Re: [asterisk-users] Asterisk 1.2.24 simultaneous call
limits.

	On Thu, 20 Sep 2007, Wai Wu wrote:

	>
	> Hi everyone,
	>
	> I am running into wall today with simultaneous call limits. I
have two
	> Asterisk machines (fast 3GHz C2D with 2GB of ram). I tried to
create a
	> lot of sip calls from one machine to the other by issuing AMI
Originate
	> commands to one machine. The machine that makes calls plays a
message
	> (demo-intruct) upon the other machine answer. The machine
receives the
	> calls just waits for 40 seconds then hangs up. Throught the
manager
	> connection, I was creating 10 calls per-second. I also have
sip phone
	> registered with the calling machine. At around 150 to 200
calls. When I
	> call the machine that's making all the calls, most of the
calls couldn't
	> go through. For the ones that went through, most of them will
drop off
	> within seconds of the call. But here is catch. When I run
'top', the cpu
	> is idling 97%. My question is. Is there a limit on the number
of
	> simultaneous calls Asterisk can handle? I know I have very
fast systems.
	> Shouldn't they be able to handle that many calls? What is your
take?

	200 calls using g711 needs 16Mb/sec of network bandwidth - each
way. (200 
	* 80Kbs) This is well within the limits of a 100Mb network
interface.

	However it also needs 50 packets per second of 160 bytes + IP
overhead 
	each way, per call, so thats 20,000 packets/second, and that
might well be 
	the bottleneck for your system, not just in the hardware issues
required 
	to shovel that many packets over the various buses, but the
Linux overhead 
	of schedulling each of the 200 threads to take/send that data in

	real-time.

	You might want to run iperf on each machine with nothing else
going and 
	see just how many UDP packets of 160 bytes you can push between
the 
	machines before packet loss starts.

	Gordon

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