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tzafrir.cohen at xorco...
Guest
|
 Posted: Sun May 18, 2008 1:25 pm Post subject: [asterisk-users] Asterisk concurrent calls count |
 |
|
On Sun, May 18, 2008 at 02:02:18PM -0400, Steve Totaro wrote:
| Quote: | Asterisk benchmarking is one of the topics that comes up on the list
frequently and consistently for what, like the last six years (that I
have been involved with Asterisk)? I would call that a "Salt
Boulder".
On the wiki there is a long page dedicated to dimensioning. I know
beyond a shadow of a doubt that people really are very interested in
the numbers.
If Digium won't supply benchmarking, then let's have a 3rd party throw
down the gauntlet.
Digium vs Sangoma on stock kernels and stock machines with identical
call load, apps, transcoding. No optimization, just stock, and see at
what load we hit a breaking point.
|
What stock kernel?
No optimizations? (-O0? )
No optimized glibc either?
| Quote: |
Then Asterisk vs FreeSwitch on similar apps pushing them to the
breaking point as well.
|
I'd say that in both cases a competent benchmarketeer would be able to
suggest tests that will provide results proving "his" side s the clear
winner.
| Quote: |
Nothing but pure numbers.
|
Sure.
/dev/random also gives me pure numbers 
Please suggest a test environment if you think it is so easy to come up
with one.
Frankly I'd say that marketing battles are not the best place to start
working on a good benchmark. Everyone will be tainted and suspected.
--
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 |
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Mike at Trest.COM
Guest
|
| Posted: Sun May 18, 2008 2:18 pm Post subject: [asterisk-users] Asterisk concurrent calls count |
|
|
At 02:25 PM 5/18/2008, Tzafrir Cohen wrote:
| Quote: | Please suggest a test environment
|
IMHO, it is definitely NOT EASY to come up with a standardized test
without some standardized network configurations and standardized
load generation tools. It is even harder when a non-standard or
niche application use is intended.
For example: I had to run a bench mark with live traffic calls that
entered by the PSTN and VoIP gateways into a PAY-FOR-PLAY farm of asterisks.
I had to generate a minimum load of 5,000 simultaneous calls with a
specific distribution of call durations and to continuously pump that
traffic to attain specific objectives for rate-of-arrival on new calls.
This required me to tie up 10,000 phone lines. 5,000 outgoing calls
to specific numbers that would be terminated by specific carriers
plus another 5,000 inbound VoIP and TDM lines to receive those calls.
Reason eventually prevailed and we got the marketing & program
managers to understand what can be shown by a much smaller set of
lines (3,000 total, 1,500 IN and 1,500 OUT). This was a 3% sample
of the intended full scale loading rather than a 10% loading.
I would not expect any generalized benchmark to even begin to address
all of the non-Asterisk elements in this over-all system. Indeed,
how could I even base any estimates for this based on generalized
benchmarks for products optimized for mass-market PBX,IVR, or
CALL-routing applications.
I am using extreme examples to make the point that the
Integrator-Reseller has that responsibility.
For the non-extreme examples, the vendors conservative estimates for
middle-of-the-road users of pre-prepared solutions are just fine.
BTW: It required only two guys and one hour to setup and perform the
test with Asterisks. It took several days of advanced negotiation to
agree on the methodology with all concerned. This is a typical
situation when you want to make sure the client knows enough to make
a valid decision.
..mike.. |
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asterisk.org at sedwar...
Guest
|
| Posted: Sun May 18, 2008 2:32 pm Post subject: [asterisk-users] Asterisk concurrent calls count |
|
|
On Sun, 18 May 2008, Steve Totaro wrote:
| Quote: | On Sun, May 18, 2008 at 12:32 AM, Jay R. Ashworth <jra at baylink.com> wrote:
| Quote: | On Sat, May 17, 2008 at 06:43:51PM -0400, Steve Totaro wrote:
| Quote: | Maybe next they will charge $250 for "conference bridge" capabilities.
It's a joke to cripple things that can be enabled by flicking a
switch.
|
|
|
If a feature adds value to a product, the customer will pay more for it.
If a feature increases your support cost, you will charge more for it.
| Quote: | I guess I hate to see something I have viewed as such a huge paradigm
shift and disruptive force from selling boxes to selling knowledge.
|
If I buy a bare DL380 from you, will it cost the same as a fully loaded
DL380 configured, optimized, and guaranteed to handle 400 seats? I think
you sell your "knowledge" as well as a "box" as well as your time.
| Quote: | I do however, think that Digium should provide some rough concurrent
call figures and I guess that is how I got off topic on this SwitchVox
tangent. There are some common feature sets, especially when looking
at PBX functionality with or without Zap or transcoding hardware that
could be published (with a disclaimer of course). There are also
common server platforms but that is more of a moving target.
|
Someone brought up the TPC benchmark. The purpose of the benchmark is to
"standardize" a synthetic processing load to allow competing vendors to
beat their chests. Who is competing with Digium? Where is the competition?
It's not in software, its in hardware. Thus, the competitors are IBM,
Dell, HP, Zonbu, etc. Since our marketplace is so small, the
aforementioned vendors are not interested in the market so the burden
falls to the interested parties -- us.
If we can agree on a couple of benchmark scenarios, we can then test our
hardware and post our results to the wiki in table and graph form.
In the interest in starting the process, here are a couple of metrics I'd
be interested in.
) What is the maximum number of simultaneous calls (1 on 1 conversations)
that can be bridged before call quality is impaired>
) What is the maximum number of simultaneous calls that can be put into a
single meetme conference before call quality is impaired>
) What is the maximum number of 3 person (agent, customer, supervisor)
meetme conferences before call quality is impaired>
How do you objectively measure call quality? Pass a sine wave at the upper
and lower range of a human voice and compare the waveforms?
How do you construct a "standard" benchmark test bed? 2 identical Asterisk
systems, 1 beating on the other?
These metrics should be run for each technology (IAX, SIP, TDM via T1, TDM
via USB) as well as Asterisk version (1.2, 1.4, 1.6)
There are a lot of variables (OS flavor, OS version, OS tweaks, gcc
version, network interface and driver, etc.) that need to be identified
and as we collect more samples we may discover that some of these
variables are important and some are not. This implies that at least in
the "beta" stage of developing a benchmark, submitters must "own" their
samples and be willing to re-run tests as the benchmark is refined.
Thanks in advance,
------------------------------------------------------------------------
Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000 |
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jra at baylink.com
Guest
|
| Posted: Sun May 18, 2008 10:54 pm Post subject: [asterisk-users] Asterisk concurrent calls count |
|
|
On Sun, May 18, 2008 at 02:02:18PM -0400, Steve Totaro wrote:
| Quote: | If Digium won't supply benchmarking, then let's have a 3rd party throw
down the gauntlet.
Digium vs Sangoma on stock kernels and stock machines with identical
call load, apps, transcoding. No optimization, just stock, and see at
what load we hit a breaking point.
Then Asterisk vs FreeSwitch on similar apps pushing them to the
breaking point as well.
Nothing but pure numbers.
|
Generated by a tester who can define "breaking" in a fashion that suits
everyone. I've just had a problem clear up by bumping a machine from
Core2Duo to Core2Quad, that I was almost certain was a crappy Ethernet
cable, 400 ft over spec.
Defining what's "stable" and what's "not" will be the hard part, not
running the tests.
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) |
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jra at baylink.com
Guest
|
| Posted: Sun May 18, 2008 10:57 pm Post subject: [asterisk-users] Asterisk concurrent calls count |
|
|
On Sun, May 18, 2008 at 12:32:01PM -0700, Steve Edwards wrote:
| Quote: | | Quote: | I guess I hate to see something I have viewed as such a huge paradigm
shift and disruptive force from selling boxes to selling knowledge.
|
If I buy a bare DL380 from you, will it cost the same as a fully loaded
DL380 configured, optimized, and guaranteed to handle 400 seats? I think
you sell your "knowledge" as well as a "box" as well as your time.
|
This is a good point, so let me expand it:
Steve (Totaro): *this is what your clients are paying you for*. To
know those answers.
Are you turning around and bitching that someone isn't giving you those
answers for free so you can charge clients for them?
| Quote: | In the interest in starting the process, here are a couple of metrics I'd
be interested in.
) What is the maximum number of simultaneous calls (1 on 1 conversations)
that can be bridged before call quality is impaired>
) What is the maximum number of simultaneous calls that can be put into a
single meetme conference before call quality is impaired>
) What is the maximum number of 3 person (agent, customer, supervisor)
meetme conferences before call quality is impaired>
How do you objectively measure call quality? Pass a sine wave at the upper
and lower range of a human voice and compare the waveforms?
|
No, actually there are benches for that; cell providers have a standard
or two, I think.
| Quote: | How do you construct a "standard" benchmark test bed? 2 identical Asterisk
systems, 1 beating on the other?
These metrics should be run for each technology (IAX, SIP, TDM via T1, TDM
via USB) as well as Asterisk version (1.2, 1.4, 1.6)
There are a lot of variables (OS flavor, OS version, OS tweaks, gcc
version, network interface and driver, etc.) that need to be identified
and as we collect more samples we may discover that some of these
variables are important and some are not. This implies that at least in
the "beta" stage of developing a benchmark, submitters must "own" their
samples and be willing to re-run tests as the benchmark is refined.
|
Oh yeah; it's not a small job.
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) |
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|
|
stotaro at totarotechn...
Guest
|
| Posted: Sun May 18, 2008 11:19 pm Post subject: [asterisk-users] Asterisk concurrent calls count |
|
|
On Sun, May 18, 2008 at 3:32 PM, Steve Edwards
<asterisk.org at sedwards.com> wrote:
| Quote: | On Sun, 18 May 2008, Steve Totaro wrote:
| Quote: | On Sun, May 18, 2008 at 12:32 AM, Jay R. Ashworth <jra at baylink.com> wrote:
| Quote: | On Sat, May 17, 2008 at 06:43:51PM -0400, Steve Totaro wrote:
| Quote: | Maybe next they will charge $250 for "conference bridge" capabilities.
It's a joke to cripple things that can be enabled by flicking a
switch.
|
|
|
If a feature adds value to a product, the customer will pay more for it.
If a feature increases your support cost, you will charge more for it.
|
I charge per hour. If they want the additional functionality, it must
be defined in the scope of work and they will pay for it based on my
hourly rate.
| Quote: |
| Quote: | I guess I hate to see something I have viewed as such a huge paradigm
shift and disruptive force from selling boxes to selling knowledge.
|
If I buy a bare DL380 from you, will it cost the same as a fully loaded
DL380 configured, optimized, and guaranteed to handle 400 seats? I think
you sell your "knowledge" as well as a "box" as well as your time.
|
I already addressed this in another post in this thread. Your
assumptions are incorrect on how I do business.
| Quote: | | Quote: | I do however, think that Digium should provide some rough concurrent
call figures and I guess that is how I got off topic on this SwitchVox
tangent. There are some common feature sets, especially when looking
at PBX functionality with or without Zap or transcoding hardware that
could be published (with a disclaimer of course). There are also
common server platforms but that is more of a moving target.
|
Someone brought up the TPC benchmark. The purpose of the benchmark is to
"standardize" a synthetic processing load to allow competing vendors to
beat their chests. Who is competing with Digium? Where is the competition?
It's not in software, its in hardware. Thus, the competitors are IBM,
Dell, HP, Zonbu, etc. Since our marketplace is so small, the
aforementioned vendors are not interested in the market so the burden
falls to the interested parties -- us.
If we can agree on a couple of benchmark scenarios, we can then test our
hardware and post our results to the wiki in table and graph form.
In the interest in starting the process, here are a couple of metrics I'd
be interested in.
) What is the maximum number of simultaneous calls (1 on 1 conversations)
that can be bridged before call quality is impaired>
) What is the maximum number of simultaneous calls that can be put into a
single meetme conference before call quality is impaired>
) What is the maximum number of 3 person (agent, customer, supervisor)
meetme conferences before call quality is impaired>
How do you objectively measure call quality? Pass a sine wave at the upper
and lower range of a human voice and compare the waveforms?
How do you construct a "standard" benchmark test bed? 2 identical Asterisk
systems, 1 beating on the other?
These metrics should be run for each technology (IAX, SIP, TDM via T1, TDM
via USB) as well as Asterisk version (1.2, 1.4, 1.6)
There are a lot of variables (OS flavor, OS version, OS tweaks, gcc
version, network interface and driver, etc.) that need to be identified
and as we collect more samples we may discover that some of these
variables are important and some are not. This implies that at least in
the "beta" stage of developing a benchmark, submitters must "own" their
samples and be willing to re-run tests as the benchmark is refined.
Thanks in advance,
------------------------------------------------------------------------
Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
|
Steve, I would like to discuss this more with you. Please contact me
offlist if you are interested in going forward or being involved.
Thanks,
Steve Totaro |
|
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|
|
tzafrir.cohen at xorco...
Guest
|
| Posted: Mon May 19, 2008 2:03 am Post subject: [asterisk-users] Asterisk concurrent calls count |
|
|
On Mon, May 19, 2008 at 12:13:05AM -0400, Steve Totaro wrote:
| Quote: | I am complaining that they should be provided by Digium. I have an
early source of some funding for benchmarking, so it certainly will
not be free. To the vendors it will. I will do their jobs for them.
|
So far you have not come up with a description of a benchmark.
You have not even described clearly what it is that you want to test.
--
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 |
|
| Back to top |
|
|
a.olekhnovich at gmail...
Guest
|
| Posted: Mon May 19, 2008 2:52 am Post subject: [asterisk-users] Asterisk concurrent calls count |
|
|
Thanks very much for your examples
On Fri, May 16, 2008 at 8:59 PM, Sherwood McGowan <
sherwood.mcgowan at gmail.com> wrote:
| Quote: | 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
_______________________________________________
-- 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
|
--
Best Regards
Alexander Olekhnovich
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stotaro at totarotechn...
Guest
|
| Posted: Mon May 19, 2008 6:27 am Post subject: [asterisk-users] Asterisk concurrent calls count |
|
|
On Mon, May 19, 2008 at 3:03 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
| Quote: | On Mon, May 19, 2008 at 12:13:05AM -0400, Steve Totaro wrote:
| Quote: | I am complaining that they should be provided by Digium. I have an
early source of some funding for benchmarking, so it certainly will
not be free. To the vendors it will. I will do their jobs for them.
|
So far you have not come up with a description of a benchmark.
You have not even described clearly what it is that you want to test.
--
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
|
Yes, that is right "so far". Very observant, although the thread
title may be a clue......
Thanks,
Steve Totaro |
|
| Back to top |
|
|
tzafrir.cohen at xorco...
Guest
|
| Posted: Mon May 19, 2008 6:39 am Post subject: [asterisk-users] Asterisk concurrent calls count |
|
|
On Mon, May 19, 2008 at 07:27:00AM -0400, Steve Totaro wrote:
| Quote: | On Mon, May 19, 2008 at 3:03 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
| Quote: | On Mon, May 19, 2008 at 12:13:05AM -0400, Steve Totaro wrote:
| Quote: | I am complaining that they should be provided by Digium. I have an
early source of some funding for benchmarking, so it certainly will
not be free. To the vendors it will. I will do their jobs for them.
|
So far you have not come up with a description of a benchmark.
You have not even described clearly what it is that you want to test.
|
|
| Quote: | Yes, that is right "so far". Very observant, although the thread
title may be a clue......
|
As others have noted, this is mostly mmeaningless.
I think I can easily get some 1000-s of channels running on this
Asteirsk instance on my desktop.
Yeah, those would be Local channels and will push no frames at all. But
who cares: my great PBX has many concurrent calls.
--
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 |
|
| Back to top |
|
|
sherwood.mcgowan at gm...
Guest
|
| Posted: Mon May 19, 2008 7:40 am Post subject: [asterisk-users] Asterisk concurrent calls count |
|
|
Alexander Olekhnovich wrote:
| Quote: | Thanks very much for your examples
On Fri, May 16, 2008 at 8:59 PM, Sherwood McGowan
<sherwood.mcgowan at gmail.com <mailto:sherwood.mcgowan at gmail.com>> wrote:
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?
| Quote: |
--
Thanks in advance
Alexander Olekhnovich
|
------------------------------------------------------------------------
| Quote: |
_______________________________________________
-- Bandwidth and Colocation Provided by
|
http://www.api-digital.com --
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
_______________________________________________
-- 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
--
Best Regards
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
|
Alexander,
No problem |
|
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|
|
stotaro at totarotechn...
Guest
|
| Posted: Mon May 19, 2008 8:41 am Post subject: [asterisk-users] Asterisk concurrent calls count |
|
|
On Mon, May 19, 2008 at 7:39 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
| Quote: | On Mon, May 19, 2008 at 07:27:00AM -0400, Steve Totaro wrote:
| Quote: | On Mon, May 19, 2008 at 3:03 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
| Quote: | On Mon, May 19, 2008 at 12:13:05AM -0400, Steve Totaro wrote:
| Quote: | I am complaining that they should be provided by Digium. I have an
early source of some funding for benchmarking, so it certainly will
not be free. To the vendors it will. I will do their jobs for them.
|
So far you have not come up with a description of a benchmark.
You have not even described clearly what it is that you want to test.
|
|
| Quote: | Yes, that is right "so far". Very observant, although the thread
title may be a clue......
|
As others have noted, this is mostly mmeaningless.
I think I can easily get some 1000-s of channels running on this
Asteirsk instance on my desktop.
Yeah, those would be Local channels and will push no frames at all. But
who cares: my great PBX has many concurrent calls.
--
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
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Whatever, vendor.... |
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stotaro at totarotechn...
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| Posted: Mon May 19, 2008 8:54 am Post subject: [asterisk-users] Asterisk concurrent calls count |
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On Mon, May 19, 2008 at 9:41 AM, Steve Totaro
<stotaro at totarotechnologies.com> wrote:
| Quote: | On Mon, May 19, 2008 at 7:39 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
| Quote: | On Mon, May 19, 2008 at 07:27:00AM -0400, Steve Totaro wrote:
| Quote: | On Mon, May 19, 2008 at 3:03 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote:
| Quote: | On Mon, May 19, 2008 at 12:13:05AM -0400, Steve Totaro wrote:
| Quote: | I am complaining that they should be provided by Digium. I have an
early source of some funding for benchmarking, so it certainly will
not be free. To the vendors it will. I will do their jobs for them.
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So far you have not come up with a description of a benchmark.
You have not even described clearly what it is that you want to test.
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| Quote: | Yes, that is right "so far". Very observant, although the thread
title may be a clue......
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As others have noted, this is mostly mmeaningless.
I think I can easily get some 1000-s of channels running on this
Asteirsk instance on my desktop.
Yeah, those would be Local channels and will push no frames at all. But
who cares: my great PBX has many concurrent calls.
--
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 --
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Whatever, vendor....
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As you have conveniently snipped what I had previously proposed to
benchmark then your silly proposal of local channels is sneaky at the
least and at most an attempt to discredit any of the benchmarks.
Maybe one day Xorcom will be included, don't be mad that they don't
make the first cut.
The methodology is the only thing lacking, but I have a couple people
to help me in this regard so far.
Thanks,
Steve Totaro |
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jra at baylink.com
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| Posted: Mon May 19, 2008 9:42 am Post subject: [asterisk-users] Asterisk concurrent calls count |
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On Mon, May 19, 2008 at 09:54:47AM -0400, Steve Totaro wrote:
| Quote: | As you have conveniently snipped what I had previously proposed to
benchmark then your silly proposal of local channels is sneaky at the
least and at most an attempt to discredit any of the benchmarks.
Maybe one day Xorcom will be included, don't be mad that they don't
make the first cut.
The methodology is the only thing lacking, but I have a couple people
to help me in this regard so far.
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There are many decaffeinated brands that are *just* as flavorful,
Steve.
I'm pretty sure there are actually standards for this;
Bellcore/Telcordia may have something. Alex?
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) |
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