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[asterisk-biz] Spec advice


 
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mroberts1818 at gmail.com
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PostPosted: Mon Mar 17, 2008 9:15 am    Post subject: [asterisk-biz] Spec advice Reply with quote

List,

I'm putting together a conf bridge solution and the main requirement is,

Must be scalable and out of the box be able to handle 1000 callers.

Any advice on hardware and necessary bandwidth?

Obviously looking to use asterisk but if there are better suited solutions it would be appreciated.

Thx,

- mike

Sent from my Verizon Wireless BlackBerry


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abalashov at evaristes...
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PostPosted: Mon Mar 17, 2008 10:04 am    Post subject: [asterisk-biz] Spec advice Reply with quote

Well, as far as Asterisk goes, you won't get 1000+ concurrent callers or
1000 MeetMe rooms on a single box. That is just not really possible,
especially given the intensity of the audio mixing required to make
MeetMe rooms happen. You would have to build your infrastructure in a
distributed manner; set up x amount of conferences on several machines
that are each only capable of having y users, and have a proxy or
something of the sort route the calls to the machines based on that
information. Or, have the conference rooms dynamically created
according to need in a load-balanced manner.

I suppose it is also possible to bridge two conference bridges together,
technically, by stimulating an Originate command that has a kickback to
a MeetMe() cross-connect on the near end and the far end. This would
theoretically allow you to have distributed conferences in which the
members are on multiple servers. But I haven't tested how well this
works / if it is desirable in any way whatsoever.

Commercial solutions - there are lots of them, but they are all rather
expensive if you wish to install them out of the box, and on-premises,
and are difficult to work with and customise. If all you need is a
conference bridge, you might want to consider purchasing it as a service
from someone who already provides it on a very large scale, such as
RainDance, etc., who I believe does it using TDM telephony gear.

mroberts1818@gmail.com wrote:
Quote:
List,

I'm putting together a conf bridge solution and the main requirement is,

Must be scalable and out of the box be able to handle 1000 callers.

Any advice on hardware and necessary bandwidth?

Obviously looking to use asterisk but if there are better suited solutions it would be appreciated.

Thx,

- mike

Sent from my Verizon Wireless BlackBerry


_______________________________________________
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--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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stotaro at totarotechn...
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PostPosted: Mon Mar 17, 2008 12:38 pm    Post subject: [asterisk-biz] Spec advice Reply with quote

http://www.mail-archive.com/asterisk-users@lists.digium.com/msg199936.html

Depending on your configuration, such as codec translation, TDM, etc
will determine the amount of servers required. I would think you
could probably get it done with three good servers doing strictly SIP
with same codec.

Thanks,
Steve Totaro

On Mon, Mar 17, 2008 at 10:52 AM, Alex Balashov
<abalashov@evaristesys.com> wrote:
Quote:
Well, as far as Asterisk goes, you won't get 1000+ concurrent callers or
1000 MeetMe rooms on a single box. That is just not really possible,
especially given the intensity of the audio mixing required to make
MeetMe rooms happen. You would have to build your infrastructure in a
distributed manner; set up x amount of conferences on several machines
that are each only capable of having y users, and have a proxy or
something of the sort route the calls to the machines based on that
information. Or, have the conference rooms dynamically created
according to need in a load-balanced manner.

I suppose it is also possible to bridge two conference bridges together,
technically, by stimulating an Originate command that has a kickback to
a MeetMe() cross-connect on the near end and the far end. This would
theoretically allow you to have distributed conferences in which the
members are on multiple servers. But I haven't tested how well this
works / if it is desirable in any way whatsoever.

Commercial solutions - there are lots of them, but they are all rather
expensive if you wish to install them out of the box, and on-premises,
and are difficult to work with and customise. If all you need is a
conference bridge, you might want to consider purchasing it as a service
from someone who already provides it on a very large scale, such as
RainDance, etc., who I believe does it using TDM telephony gear.



mroberts1818@gmail.com wrote:
Quote:
List,

I'm putting together a conf bridge solution and the main requirement is,

Must be scalable and out of the box be able to handle 1000 callers.

Any advice on hardware and necessary bandwidth?

Obviously looking to use asterisk but if there are better suited solutions it would be appreciated.

Thx,

- mike

Sent from my Verizon Wireless BlackBerry


_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz


--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599



_______________________________________________
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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-biz


_______________________________________________
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trixter at 0xdecafbad.com
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PostPosted: Mon Mar 17, 2008 1:59 pm    Post subject: [asterisk-biz] Spec advice Reply with quote

On Mon, 2008-03-17 at 13:18 -0400, Steve Totaro wrote:
Quote:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg199936.html

Depending on your configuration, such as codec translation, TDM, etc
will determine the amount of servers required. I would think you
could probably get it done with three good servers doing strictly SIP
with same codec.


One thing that the 2 public posts seem to not ask is how many people are
actually in a given conference. I had addressed this privately, along
with a couple ideas on how to accomplish this.

1 speaker and 1000 listeners does not require the same load as 1000
speakers, at least with 2 of the 4 major asterisk conferencing modules,
two I am unsure about. Sample size for muxing also affects
performance.

Basically what was given results in guessing as to what was meant so
other than saying "1000 G.711 calls requires about 100Mbps" its
difficult to answer the other part of the question.

The features of the conference can also have an impact, for example
recording the conferences.

the way you would build out a system for 100 10 person conferences is
different than you would for a lecture style 1 speaker (or very few) and
a bunch of listeners, which is different from a (in my opinion) totally
unusable 1000 person all talking no one can hear anything conference.

I do however agree that a single system would not be able to handle 1000
conference users with asterisk, although there are other open source
solutions that could possibly do it pending the outcome of some of these
unknowns.


--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!


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stotaro at totarotechn...
Guest





PostPosted: Mon Mar 17, 2008 2:32 pm    Post subject: [asterisk-biz] Spec advice Reply with quote

On Mon, Mar 17, 2008 at 2:08 PM, Trixter aka Bret McDanel
<trixter@0xdecafbad.com> wrote:
Quote:

On Mon, 2008-03-17 at 13:18 -0400, Steve Totaro wrote:
Quote:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg199936.html

Depending on your configuration, such as codec translation, TDM, etc
will determine the amount of servers required. I would think you
could probably get it done with three good servers doing strictly SIP
with same codec.


One thing that the 2 public posts seem to not ask is how many people are
actually in a given conference. I had addressed this privately, along
with a couple ideas on how to accomplish this.

1 speaker and 1000 listeners does not require the same load as 1000
speakers, at least with 2 of the 4 major asterisk conferencing modules,
two I am unsure about. Sample size for muxing also affects
performance.

Basically what was given results in guessing as to what was meant so
other than saying "1000 G.711 calls requires about 100Mbps" its
difficult to answer the other part of the question.

The features of the conference can also have an impact, for example
recording the conferences.

the way you would build out a system for 100 10 person conferences is
different than you would for a lecture style 1 speaker (or very few) and
a bunch of listeners, which is different from a (in my opinion) totally
unusable 1000 person all talking no one can hear anything conference.

I do however agree that a single system would not be able to handle 1000
conference users with asterisk, although there are other open source
solutions that could possibly do it pending the outcome of some of these
unknowns.


--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!


Very true, I overlooked those variables. app_ices could be handy too.

Thanks,
Steve Totaro

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mroberts1818 at gmail.com
Guest





PostPosted: Mon Mar 17, 2008 9:56 pm    Post subject: [asterisk-biz] Spec advice Reply with quote

Thanks for the insight, much appreciated. I'll reach out off list when needed.

Thanks again,

-Mike

On Mon, Mar 17, 2008 at 3:22 PM, Steve Totaro <stotaro@totarotechnologies.com (stotaro@totarotechnologies.com)> wrote:
Quote:

On Mon, Mar 17, 2008 at 2:08 PM, Trixter aka Bret McDanel
<trixter@0xdecafbad.com (trixter@0xdecafbad.com)> wrote:
Quote:

On Mon, 2008-03-17 at 13:18 -0400, Steve Totaro wrote:
Quote:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg199936.html

Depending on your configuration, such as codec translation, TDM, etc
will determine the amount of servers required. I would think you
could probably get it done with three good servers doing strictly SIP
with same codec.


One thing that the 2 public posts seem to not ask is how many people are
actually in a given conference. I had addressed this privately, along
with a couple ideas on how to accomplish this.

1 speaker and 1000 listeners does not require the same load as 1000
speakers, at least with 2 of the 4 major asterisk conferencing modules,
two I am unsure about. Sample size for muxing also affects
performance.

Basically what was given results in guessing as to what was meant so
other than saying "1000 G.711 calls requires about 100Mbps" its
difficult to answer the other part of the question.

The features of the conference can also have an impact, for example
recording the conferences.

the way you would build out a system for 100 10 person conferences is
different than you would for a lecture style 1 speaker (or very few) and
a bunch of listeners, which is different from a (in my opinion) totally
unusable 1000 person all talking no one can hear anything conference.

I do however agree that a single system would not be able to handle 1000
conference users with asterisk, although there are other open source
solutions that could possibly do it pending the outcome of some of these
unknowns.


--
Trixter http://www.0xdecafbad.com Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!




Very true, I overlooked those variables. app_ices could be handy too.

Thanks,
Steve Totaro


_______________________________________________
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