VoIP Mailing List Archives :: View topic - [asterisk-users] Asterisk concurrent calls count

Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community

[asterisk-users] Asterisk concurrent calls count

Goto page Previous 1, 2, 3, 4 Next

VoIP Mailing List Archives Forum Index -> Asterisk Users

View previous topic :: View next topic

Author

Message

sherwood.mcgowan at gm...
Guest

Posted: Fri May 16, 2008 12:26 pm Post subject: [asterisk-users] Asterisk concurrent calls count

David Backeberg wrote:

Quote:

Has anybody ever tried to roll their own VoIP or Zaptel load
simulator? How did they do it?

SIPP can help with benchmarking SIP calls and you can loop back T1
calls if you have two machines with T1 cards or even one machine with
multiple T1 ports.

SIPp looks like it's exactly the right tool for voip load generation /
simulation. Sweet!

_______________________________________________
-- 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

I've used SIPp before for benchmarking, it works quite well

asterisk.org at sedwar...
Guest

Posted: Fri May 16, 2008 12:52 pm Post subject: [asterisk-users] Asterisk concurrent calls count

On Fri, 16 May 2008, Steve Totaro wrote:

Quote:

On Fri, May 16, 2008 at 9:47 AM, David Backeberg <dbackeberg at gmail.com> wrote:

Quote:

Has anybody ever tried to roll their own VoIP or Zaptel load
simulator? How did they do it?

SIPP can help with benchmarking SIP calls and you can loop back T1
calls if you have two machines with T1 cards or even one machine with
multiple T1 ports.

Then just look at top. Make a few test calls and see if they are choppy....

What value do you look at with top? (Especially with multiple
processor/core servers.) I have an old 1.2.7 server with "custom features"
hacked in that leaks memory. We know audio quality goes to hell when
Asterisk hoards more than 100mb.

How do you quantify "choppy?" Anybody volunteer to write "app-MOS?"

Thanks in advance,
------------------------------------------------------------------------
Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

sherwood.mcgowan at gm...
Guest

Posted: Fri May 16, 2008 12:59 pm Post subject: [asterisk-users] Asterisk concurrent calls count

Alexander Olekhnovich wrote:

Quote:

Hi Asterisk Users,

I'm interested in how many concurrent calls Asterisk can process
without troubles. I mean 1 Asterisk server (software) like either
proxy or media server (any numbers will be appropriate).

1. Is there any limitations by the software? What is this number?
2. What is the maximum count of concurrent calls you've ever seen/tested?

--
Thanks in advance
Alexander Olekhnovich
------------------------------------------------------------------------

_______________________________________________
-- 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

Rather than jump into the heavy list of replies, in which there's some
heated discussion, I thought I'd offer a quick $0.02:

Asterisk's concurrent call capabilities is limited (as far as I know)
only by the hardware you're using and the implementation. By this I mean
that the amount of transcoding, meetme conferences, SIP/IAX/Zap
channels, recording, CDR backend, etc...all take their toll on your
hardware's capabilities.

I'll give you two examples:
1. On a Dual 1.5Ghz XEON, 2GB RAM server running CentOS 4.5(unsure on
this anymore) with only Asterisk 1.4 TRUNK in 1995 in a SIP only
environment with ONLY ulaw encoding, I've seen 500+ concurrent calls
with over 2K users on a single machine. All clients were set for
canreinvite=no, and qualify=yes. This system did not show degradation of
performance.

2. I'm currently working with a client that has a Dual 2.5 Ghz, 2GB RAM
server, running Debian Etch. They are running two EM Wink T1 Trunks, and
51 Zap phones locally running through Adtran Total Access Channel Banks,
12 POTS lines running through a Rhino channel bank, and 27 SIP Phones.
Concurrent calls only run at around 43 calls currently, although I've
seen it as high as 53, and ALL calls are recorded other than local
spying on channels and inter-extension calls. Additionally, this server
has PostgreSQL and Apache running on it to allow administration to
review CDRs and pull recordings, and a Zabbix monitoring agent daemon
sending data to a local network Zabbix server. This server showed
little or no degradation in call quality or service (even with Sox and
Speexmix running in the background converting recordings via a
background script) until just recently when we changed T1 providers and
got EM Wink instead of the requested PRI. Before we had 99.999% of all
calls complete from dial to hangup with no issues. Now we're at 98.8%,
with calls being dropped in midconversation. I have not found the answer
to what is causing the server to drop calls, other than after the
switchover to EM_W our Zaptel accuracy started degrading. We are in the
process of figuring out how we can resolve this, including possible
hardware upgrades (which were already planned for handling recordings
better)

I hope these two examples help show you how two similar machines can
vary drastically in performance with similar hardware. Differences in
implementation make a BIG difference.

Slainte,
Sherwood McGowan

jra at baylink.com
Guest

Posted: Fri May 16, 2008 7:37 pm Post subject: [asterisk-users] Asterisk concurrent calls count

On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote:

Quote:

It seems any constructive criticism offered, you take as an attack
against Digium. That is not a good attitude.

I dunno, Steve; I wouldn't call "Digium needs to 'man-up'" constructive
criticism, myself. I'd call it an ad-hominem.

Tilghman *does* seem to be a bit of a cheerleader, but there's nothing
wrong with that... unless you're an *employee*, and you're going out of
your way to hide it.

Cheers,
-- jra
--
Jay R. Ashworth Baylink jra at baylink.com
Designer The Things I Think RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274

Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Joseph Stalin)

jra at baylink.com
Guest

Posted: Fri May 16, 2008 7:40 pm Post subject: [asterisk-users] Asterisk concurrent calls count

On Fri, May 16, 2008 at 10:54:17AM -0400, Steve Totaro wrote:

Quote:

I am very aware of this but do you think that the SoHo box (and it's
artificial cap) is maxing out no matter what the users are doing? If
so, then why can you just upgrade concurrent use via a CC and the
website?

So, Steve.... are you saying that they're quoting conservatively, and
you're *unhappy* with them for that?

I think it's great, myself.

Cheers,
-- jra
--
Jay R. Ashworth Baylink jra at baylink.com
Designer The Things I Think RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274

Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Joseph Stalin)

tilghman at mail.jeffa...
Guest

Posted: Fri May 16, 2008 8:18 pm Post subject: [asterisk-users] Asterisk concurrent calls count

On Friday 16 May 2008 19:37:59 Jay R. Ashworth wrote:

Quote:

On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote:

Quote:

It seems any constructive criticism offered, you take as an attack
against Digium. That is not a good attitude.

I'm been a member of this community far longer than I've worked for Digium,
and even then, I form my own opinions and I call them as I see them. If I
can't say something because of insider knowledge, I know well enough to keep
my mouth shut, but this is not one of those times. And if there _is_
something wrong with the way Digium is doing something, I also am more than
happy to put up a big fuss until it's fixed.

I'm probably a bit of a loose cannon, but they knew that when they hired
me. Wink

--
Tilghman

stotaro at totarotechn...
Guest

Posted: Sat May 17, 2008 7:51 am Post subject: [asterisk-users] Asterisk concurrent calls count

On Fri, May 16, 2008 at 9:18 PM, Tilghman Lesher
<tilghman at mail.jeffandtilghman.com> wrote:

Quote:

On Friday 16 May 2008 19:37:59 Jay R. Ashworth wrote:

Quote:

On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote:

Quote:

It seems any constructive criticism offered, you take as an attack
against Digium. That is not a good attitude.

--
Tilghman

"Man up" and post some common benchmarks. It is easy to leave out
context and take one line to prove your point. Politicians do it
every day as of late.

Let me contact Sangoma, I am sure they will do it..... In fact, they
wanted me to do head to head benchmarks against Digium products. Is
Digium game because Sangoma is ready willing and able?

It is about the money, like it or not. You are going to an Avaya type
licensing scheme, everything is charged per port. The box is capable
of doing more but you turn it off until you get more money. It's like
the Definity G3s I have worked with. The box can do everything but
until you pony up, it is not activated.

Any SwitchVox sale I have tried to pitch dies quickly and this is even
involving Switchvox reps on a conference call.

How about if I don't want support and use my own hardware, then can do
I still have to pay to upgrade to SMB or whatever? Follow the logic?
Anyways, the profit margin on "appliances" is way too low. I might as
well sell 3Coms or NECs if I am selling boxes with per seat license
fees and have to hack the box to do any customization.

They are not being conservative, when all you do is put a CC and then
a button shows up to upgrade, this is the same hardware mind you....

Guess I will stick to my DL 380s and (if a GUI is required) FreePBX or
Druid (if it tests out ok).

Thanks,
Steve Totaro

stotaro at totarotechn...
Guest

Posted: Sat May 17, 2008 8:06 am Post subject: [asterisk-users] Asterisk concurrent calls count

On Sat, May 17, 2008 at 8:51 AM, Steve Totaro
<stotaro at totarotechnologies.com> wrote:

Quote:

On Fri, May 16, 2008 at 9:18 PM, Tilghman Lesher
<tilghman at mail.jeffandtilghman.com> wrote:

Quote:

On Friday 16 May 2008 19:37:59 Jay R. Ashworth wrote:

Quote:

On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote:

Quote:

It seems any constructive criticism offered, you take as an attack
against Digium. That is not a good attitude.

--
Tilghman

Anyways, isn't Asterisk 1.2.x and FC6 EOL?

Thanks,
Steve Totaro

Mike at Trest.COM
Guest

Posted: Sat May 17, 2008 12:58 pm Post subject: [asterisk-users] Asterisk concurrent calls count

Al, Randy, (and others):

What Al calls "one very weak area" for Asterisk is IMHO a difference
in market perceptions.
Asterisk is positioned for CPE - PBX - Appliance market which needs
feature-rich appeal
and mass-market focus.

Using asterisk for "large scale" does not mean that I have used it as
a large scale PBX.
Indeed, the "FARM" approach that we will be discussing on Friday 23rd
is for very-large
scale deployments with a reduced-feature-set focus.

Simply put, this is not on Digium'a program for broad market
push. Rightfully, Digium is
expecting it's distribution channels to push it CPE-PBX and mass
market solutions.

So it is up to the thousands of Asterisk consultants to be aware of
these techniques
and to serve the much smaller number of clients (mostly VoIP network
operators)
who need to deploy very large scale networks.

Indeed, I am now working on a design now that supports 100,000+
simultaneous participants
in an application specific deployment. In this scale of telephony
application, the issues of
IP bandwidth and PSTN carrier access points are much more difficult
to manage than anything
related to the Asterisk platform.

If this is your interest, then drop in
http://voipusersconference.org The context of the discussion
is NON-COMMERCIAL. I have no product or service for sale. I am just
discussing a different approach
to using Asterisk.

..mike..

At 09:42 AM 5/16/2008, randulo wrote:

Quote:

http://voipusersconference.org

On Fri, May 16, 2008 at 1:59 PM, Al Baker <bwentdg at pipeline.com> wrote:

Quote:

this is one very weak area for *. There is NO ANSWER.

Hi,

There have been a couple of threads on this subject this week, so I'd
remind everyone that next Friday's VoIP Users Conference is about
*large scale* asterisk:

After many requests, we finally have someone to talk on large scale
implementation of VoIP systems with asterisk. Using a farm of Asterisk
and Digium cards, tens Of Thousands of simultaneous calls can be made
and Mike Trest has offered to take it all apart for us to look inside.

More about Mike Trest: http://www.mike.trest.com/

_______________________________________________
-- 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

Mike at Trest.COM
Guest

Posted: Sat May 17, 2008 2:11 pm Post subject: [asterisk-users] Asterisk concurrent calls count

At 11:44 AM 5/16/2008, you wrote:

Quote:

Yes, you could probably add 2 or 3 or 10 or 15 to the number of calls
that a particular machine could handle, but from a support perspective, it
doesn't matter how many the machine could theoretically handle, it matters
how many it could handle in the particular installation in a supportable
configuration (those are all those pesky variables we've been talking about).

Absolutely! Right On! Tell it like it is! And many other
cryptic encouragements.

With very large scale deployments, I have a set of numbers available
in my head
that work well to predict how many machines will be needed for a
particular application
but I wind up being surprised by non-predictable "rate of arrival" issues.

Since most of my deployments are tied with Television and other promotional
support, a single reference by the on-screen (or on-radio) commentator, and the
phones are instantly flooded with thousands of new call setup
requests. Indeed,
one such incident in a NASCAR race with 13M viewers, produced 18,000 new calls
within two minutes. The rate of arrival of new calls was dispersed
to a farm of 60
Asterisk in three widely separated regions of the
US. However, approximately
15,000 calls were actually dropped on the PSTN / SS7 network before
ever reaching
three dispersed Asterisk farms.

Those farms were being "fed" inbound calls by a network of
250+ Nortel switches with
millions of subscribers. However, the Los Angeles area PSTN network
access facility
had only 900 spare channels available in that two minute
period. Meanwhile, every asterisk
answered every call and joint the callers into appropriate conference
groups until every
single available port was fully occupied. This illustrates that
such issues of call capacity
exist completely apart from the Asterisk or whatever machine is used
for implementation.

So everyone should not be surprised by "it depends" kinds of answers
to the question
of concurrent call counts. This application was so far off the
typical product specifications
that nothing published by Digium or anyone else could anticipate
those surprises that
come when you least expect.

..mike..

stotaro at totarotechn...
Guest

Posted: Sat May 17, 2008 3:07 pm Post subject: [asterisk-users] Asterisk concurrent calls count

On Sat, May 17, 2008 at 3:11 PM, Mike Trest - On Travel <Mike at trest.com> wrote:

Quote:

At 11:44 AM 5/16/2008, you wrote:

Quote:

I don't think anyone is expecting any rough numbers from Digium about
the telco's ingress/egress.

Thanks,
Steve Totaro

stotaro at totarotechn...
Guest

Posted: Sat May 17, 2008 3:20 pm Post subject: [asterisk-users] Asterisk concurrent calls count

On Fri, May 16, 2008 at 8:37 PM, Jay R. Ashworth <jra at baylink.com> wrote:

Quote:

On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote:

Quote:

It seems any constructive criticism offered, you take as an attack
against Digium. That is not a good attitude.

He is an employee and he does not post from a Digium account or
include that fact in his signature. Not that it is to hide the fact,
but it certainly is obfuscated.

Thanks,
Steve Totaro

michiel at vanbaak.info
Guest

Posted: Sat May 17, 2008 3:56 pm Post subject: [asterisk-users] Asterisk concurrent calls count

On 16:20, Sat 17 May 08, Steve Totaro wrote:

Quote:

On Fri, May 16, 2008 at 8:37 PM, Jay R. Ashworth <jra at baylink.com> wrote:

Quote:

On Fri, May 16, 2008 at 10:11:11AM -0400, Steve Totaro wrote:

Quote:

It seems any constructive criticism offered, you take as an attack
against Digium. That is not a good attitude.

He is an employee and he does not post from a Digium account or
include that fact in his signature. Not that it is to hide the fact,
but it certainly is obfuscated.

I think it just shows that his opinions are his, and in no way are
linked to the 'digium opinion'
--

Michiel van Baak
michiel at vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer aficionados are both called users?"

jra at baylink.com
Guest

Posted: Sat May 17, 2008 4:34 pm Post subject: [asterisk-users] Asterisk concurrent calls count

On Fri, May 16, 2008 at 08:18:46PM -0500, Tilghman Lesher wrote:

Quote:

Tilghman *does* seem to be a bit of a cheerleader, but there's nothing
wrong with that... unless you're an *employee*, and you're going out of
your way to hide it.

That's about what I thought, and -- speaking for myself -- I'm fine
with that.

Cheers,
-- jra
--
Jay R. Ashworth Baylink jra at baylink.com
Designer The Things I Think RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274

Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Joseph Stalin)

jra at baylink.com
Guest

Posted: Sat May 17, 2008 4:36 pm Post subject: [asterisk-users] Asterisk concurrent calls count

On Sat, May 17, 2008 at 08:51:04AM -0400, Steve Totaro wrote:

Quote:

It is about the money, like it or not. You are going to an Avaya type
licensing scheme, everything is charged per port. The box is capable
of doing more but you turn it off until you get more money. It's like
the Definity G3s I have worked with. The box can do everything but
until you pony up, it is not activated.

Yeah, and if you cut a jumper on a VAX11/780, it went twice as fast.

So *what*, Steve? Are they not allowed to make money?

Quote:

Any SwitchVox sale I have tried to pitch dies quickly and this is even
involving Switchvox reps on a conference call.

Ah. Then if you don't like the way they do business, vote with your
wallet; don't sell their junk.

Quote:

How about if I don't want support and use my own hardware, then can do
I still have to pay to upgrade to SMB or whatever? Follow the logic?
Anyways, the profit margin on "appliances" is way too low. I might as
well sell 3Coms or NECs if I am selling boxes with per seat license
fees and have to hack the box to do any customization.

Yup.

Quote:

They are not being conservative, when all you do is put a CC and then
a button shows up to upgrade, this is the same hardware mind you....

Guess I will stick to my DL 380s and (if a GUI is required) FreePBX or
Druid (if it tests out ok).

Sounds like the best answer to me, for you. Smile

Cheers,
-- jra
--
Jay R. Ashworth Baylink jra at baylink.com
Designer The Things I Think RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274

Those who cast the vote decide nothing.
Those who count the vote decide everything.
-- (Joseph Stalin)

Display posts from previous:

	VoIP Mailing List Archives Forum Index -> Asterisk Users	All times are GMT - 5 Hours Goto page Previous 1, 2, 3, 4 Next
Page 2 of 4

You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum