Re: [asterisk-users] load-balancing AMI and load-balancing FastAGI?

[Paul Simon]

Anyone?
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Re: [asterisk-users] Load Balancing with DNS SRV without DUNDI

[Adrian Serafini]

On 05/24/2015 11:01 PM, Mehdi Shirazi wrote:

Hi
I want to load balance SIP calls between two(or more)
Asterisks with only DNS SRV. I used bidirectional sync
Unison to synchronize configuration files and internal database file
between two Asterisk boxes.
The problem is when a calls come to Asterisk1 but SIP
endpoint is registered on Asterisk2.How we can check
a SIP endpoint is registered or not and what is Contact
information in Dialplan ?

Regards
babak




If you used Opensips with a Mysql backend.  The two Opensips servers 
could query a command db with the contact URI.


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Re: [asterisk-users] Load Balancing

[Olivier]

I came accross this article (Asterisk rtp mprovements
)
mentioning DNS based load balancing.
I will give Opensips loadbalance module further reading to better
understand how it works

Thanks for the tip.


2013/4/25 

> You the couple opensips + asterisk will help you. Opensips loadbalance
> module is your friend.
>
>
>
> Sent from my iPhone
>
> On Apr 25, 2013, at 11:44 AM, Olivier  wrote:
>
> > Hello,
> >
> > I've been given the task to study what would a good way to load balance
> SIP trafic.
> >
> > The prospective setup is :
> > - call centers sending outbound SIP trafic (no inbound) from SIP devices
> (with public fixed IP address),
> > - a couple of outbound SIP trunks to which trafic from call centers is
> to be forwarded
> > - a load balancing system between call centers and SIP trunks.
> >
> > Load balancing system main task is:
> > - provide some LCR routing,
> > - improve availability.
> >
> > Can (should) it be done with Asterisk alone or should I look for other
> components ?
> >
> > Regards
> > --
> > _
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> >   http://www.asterisk.org/hello
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>
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Re: [asterisk-users] Load Balancing

[acheraime]

You the couple opensips + asterisk will help you. Opensips loadbalance module 
is your friend.



Sent from my iPhone

On Apr 25, 2013, at 11:44 AM, Olivier  wrote:

> Hello,
> 
> I've been given the task to study what would a good way to load balance SIP 
> trafic.
> 
> The prospective setup is :
> - call centers sending outbound SIP trafic (no inbound) from SIP devices 
> (with public fixed IP address),
> - a couple of outbound SIP trunks to which trafic from call centers is to be 
> forwarded
> - a load balancing system between call centers and SIP trunks.
> 
> Load balancing system main task is:
> - provide some LCR routing,
> - improve availability.
> 
> Can (should) it be done with Asterisk alone or should I look for other 
> components ?
> 
> Regards
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] Load balancing Asterisk.

[Al lists]

Foundry serverIron does support SIP and its ASIC not a linux box Load
balancer like F5,
Refer to Chapter 10 (page 677) of ServerIron manual.
It explains everything in detail.
Also you may need to play with source nat a little bit to make your specific
configuration work, but it should work, at least in theory.


On Thu, Nov 20, 2008 at 10:25 AM, Alex Balashov
wrote:

> SIP wrote:
>
> > As for the current F5 SIP load balancer, we tried it a few years back
> > and it was a dismal failure. It wanted to do cookie-based SIP load
> > balancing and only worked with certain SIP proxies.
>
> I assume that is because there is no way RFC-supported way to insert a
> cookie into a SIP session that persists throughout the entire exchange
> with a client, including all in-dialog requests, subsequent sessions, etc?
>
> The only way I know of to make a cookie stick on the UAC side is to put
> an LR parameter into the route set, but that will only last within a
> dialog.
>
> So, I'm assuming certain SIP proxies had proprietary ways of getting
> around that in order to work with F5?
>
> --
> 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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Re: [asterisk-users] Load balancing Asterisk.

[Alex Balashov]

SIP wrote:

> As for the current F5 SIP load balancer, we tried it a few years back
> and it was a dismal failure. It wanted to do cookie-based SIP load
> balancing and only worked with certain SIP proxies.

I assume that is because there is no way RFC-supported way to insert a 
cookie into a SIP session that persists throughout the entire exchange 
with a client, including all in-dialog requests, subsequent sessions, etc?

The only way I know of to make a cookie stick on the UAC side is to put 
an LR parameter into the route set, but that will only last within a dialog.

So, I'm assuming certain SIP proxies had proprietary ways of getting 
around that in order to work with F5?

-- 
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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Re: [asterisk-users] Load balancing Asterisk.

[Grey Man]

> 3. Incoming calls - I admit complete ignorance. I don't know
> how OpenSIPS handles incoming calls, but for those to arrive
> at the user reliably they must arrive from the same IP address
> the user is registered to. Otherwise their broadband router's
> NAT firewall will just block the connection. How does OpenSIPS
> handle this? (does it handle this??)

That's the big question!

My company uses a custom SIP Proxy and SIP Registrar so I can't speak
for the details of SER derivatives but the theory is most likely the
same.

Our SIP Registrar records the proxy the REGISTER request arrived on
and updates the Asterisk realtime database outboundproxy field with
that value. When Asterisk needs to send an incoming call to the user
it looks up the SIP username in the realtime database and sends the
call thorugh the correct Proxy which solves the NAT issue you mention.

One trick for young players here is that the outboundproxyport setting
is broken in Asterisk so your Proxy will have to run on port 5060.

Regards,

Greyman.

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Re: [asterisk-users] Load balancing Asterisk.

[Alex Balashov]

Nitzan Kon wrote:
> --- On Thu, 11/20/08, Grygoriy Dobrovolskyy <[EMAIL PROTECTED]> wrote:
> 
>> I am not agreed on point 2:
>> If I understood how to install opensips + heartbeat WITHOUT
>> knowing any
>> program (opensips ? heartbear ?) or programming
>> language(hell yes!) in a
>> week ( just knew what's invite and bye ;) a more aware
>> IT professional could
>> do it in 2 days
> 
> I'm actually referring mostly to the need to build, install,
> and maintain another set (2?) of Linux boxes. The software is
> the easy part.

As someone who hates dealing with hardware, I can relate and appreciate 
why this is a pain.

But it's a lot easier than setting up the alternatives!

-- 
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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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Re: [asterisk-users] Load balancing Asterisk.

[Nitzan Kon]

--- On Thu, 11/20/08, Grygoriy Dobrovolskyy <[EMAIL PROTECTED]> wrote:

> I am not agreed on point 2:
> If I understood how to install opensips + heartbeat WITHOUT
> knowing any
> program (opensips ? heartbear ?) or programming
> language(hell yes!) in a
> week ( just knew what's invite and bye ;) a more aware
> IT professional could
> do it in 2 days

I'm actually referring mostly to the need to build, install,
and maintain another set (2?) of Linux boxes. The software is
the easy part.

Granted, if that's what we need to do - that's what we'll do.

--
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Future Nine Corporation
http://www.future-nine.com

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Re: [asterisk-users] Load balancing Asterisk.

[Grygoriy Dobrovolskyy]

2. Overkill to install and maintain (if we can get a simpler
solution)

I am not agreed on point 2:
If I understood how to install opensips + heartbeat WITHOUT knowing any
program (opensips ? heartbear ?) or programming language(hell yes!) in a
week ( just knew what's invite and bye ;) a more aware IT professional could
do it in 2 days
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Re: [asterisk-users] Load balancing Asterisk.

[SIP]

Alex Balashov wrote:
> I was about to say, I'm sure F5 can do it... but...
>
>  > price was over 6 figures
>
> Why??!
>
> It's spending money on these types of things when they are unnecessary 
> that is the undoing of every struggling VoIP provider I watch, in the 
> misguided belief that only will half a million dollars get you 
> "enterprise strength."  That was the conventional wisdom about Linux ten 
> years ago too.  Who's saying that now?  Ditto.
>
>   
F5 has ALWAYS been overpriced.

Incidentally, anyone who wants to know, F5 is a unix-based box, just
like the others. Last we used the F5s, they were all running a slightly
modified BSDI. And only slightly modified in packaging.

As for the current F5 SIP load balancer, we tried it a few years back
and it was a dismal failure. It wanted to do cookie-based SIP load
balancing and only worked with certain SIP proxies.

N.

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Re: [asterisk-users] Load balancing Asterisk.

[Alex Balashov]

Nitzan Kon wrote:

> My concerns with OpenSIPS:
> 
> 1. It's a software based solution, which means higher chance
> of software-related failure, and higher chance of failure due
> to problems with the Linux box hosting it.

A little bit of proper engineering will overcome that reasonably.

> 2. Overkill to install and maintain (if we can get a simpler
> solution)

Really?

It is, admittedly, a somewhat recondite product, but you don't have to 
build everything you run into your core competency;  you can divest 
yourself of some parts of your infrastructure and streamline and all 
that and get someone else to do it, like a real Enterprise.  :-)

Secondly, as difficult as it may be, I can't imagine anything simpler to 
accomplish what you're looking for.  The logic required is quite granular.

> 3. Incoming calls - I admit complete ignorance. I don't know
> how OpenSIPS handles incoming calls, but for those to arrive
> at the user reliably they must arrive from the same IP address
> the user is registered to. Otherwise their broadband router's
> NAT firewall will just block the connection. How does OpenSIPS
> handle this? (does it handle this??)

What role are you envisioning the proxy to be in here?  If it's a 
registrar, it will have their IP information in the stored contact URI. 
  If not, the calls can be sent somewhere else for resolution. 
Something, somewhere must know how to contact the user, yes.

-- 
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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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Re: [asterisk-users] Load balancing Asterisk.

[Alex Balashov]

I was about to say, I'm sure F5 can do it... but...

 > price was over 6 figures

Why??!

It's spending money on these types of things when they are unnecessary 
that is the undoing of every struggling VoIP provider I watch, in the 
misguided belief that only will half a million dollars get you 
"enterprise strength."  That was the conventional wisdom about Linux ten 
years ago too.  Who's saying that now?  Ditto.

-- 
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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Re: [asterisk-users] Load balancing Asterisk.

[Nitzan Kon]

N,

SIP-aware LBs do exist - but way way out of my price range.

Alex, 

Remember we are an Asterisk-based provider. I'm not going
to drop enough money on a load balancer to go bankrupt. ;) That's
exactly why I'm wondering if it's possible to do this with a
DUMB load balancer. i.e. one that would cost about the same as
building another Linux box for OpenSIPS.

I don't need a million concurrent connections. I'd be perfectly
happy with a fraction of that. Not looking to replace AT&T here,
just looking for something simple that will work reliably. :)

My concerns with OpenSIPS:

1. It's a software based solution, which means higher chance
of software-related failure, and higher chance of failure due
to problems with the Linux box hosting it.
2. Overkill to install and maintain (if we can get a simpler
solution)
3. Incoming calls - I admit complete ignorance. I don't know
how OpenSIPS handles incoming calls, but for those to arrive
at the user reliably they must arrive from the same IP address
the user is registered to. Otherwise their broadband router's
NAT firewall will just block the connection. How does OpenSIPS
handle this? (does it handle this??)

Thanks!

--
Nitzan Kon, CEO
Future Nine Corporation
www.future-nine.com

--- On Thu, 11/20/08, SIP <[EMAIL PROTECTED]> wrote:

> Unless the LB is SIP-aware, and can maintain a SIP session,
> I don't see
> how it would work. As the SIP command stream sends discrete
> commands,
> without some sort of basic level of session awareness,
> there's no
> guarantee over a reasonable-length call that the INVITE and
> BYE would
> even get sent to the same Asterisk box. If there are
> on-hold messages or
> transfers occurring, you add even more possibility of error
> into the
> mix.  Now... you could do some sort of VERY long session
> timeout, but
> overall, that's a hack that's going to drop your
> concurrent connection
> count faster than using a smaller box would.
> 
> I don't know of any functioning, SIP-aware load
> balancers at the moment.
> Doesn't mean they don't exist. I just can't
> think of any off the top of
> my head.
> 
> N.
> 
> 
> 
> Nitzan Kon wrote:
> > Alex,
> >
> > I realize and agree that "hardware" load
> balancers are actually
> > software based. I'm less concerned about that and
> more about the
> > general specs:
> >
> > Foundry ServerIron XL: rated for 1,000,000 concurrent
> connections
> > Linux box where OpenSIPS is sitting: rated for ...???
> >
> > Not to mention a simple rule on a load balancer would
> be much,
> > much easier to implement. All I need is IP-based load
> balancing
> > so installing and maintaining OpenSIPS is an overkill.
> >
> > Again, I appreciate the feedback but I am not asking
> nor looking
> > for a software solution. My question is simple:
> >
> > Will a HARDWARE load balancer work? any reason why it
> WON'T work?
> >
> > Thanks!
> >
> >
> > --- On Thu, 11/20/08, Alex Balashov
> <[EMAIL PROTECTED]> wrote:
> >
> >   
> >> What do you mean by "hardware" options? 
> There are
> >> no ASIC-assisted SIP load balancers out there. 
> :-)  The
> >> embedded "hardware-based" options are
> load
> >> balancers built just like PCs - often on top of a
> UNIX
> >> kernel - that run a software application-aware
> load
> >> balancing suite.
> >>
> >> Your best bet is a proxy for the round-robin part,
> and
> >> Linux-HA for the high availability of the proxy,
> as Grygoriy
> >> suggested.
> >>
> >> Nitzan Kon wrote:
> >>
> >> 
> >>> --- On Thu, 11/20/08, Grygoriy Dobrovolskyy
> >>>   
> >> <[EMAIL PROTECTED]> wrote:
> >> 
>  2 openser servers with 3 ip adresses (1
> virtual) +
>  heartbeat to ensure the
>  failover + watchdog to ensure if
>  
> >> opensips/kamalio/openser
> >> 
>  crashes a nice
>  failover & reboot, it is working
> stable here
>  (dispatching to 10 servers +
>  owners DID dispatch to their respective
> servers)
> 
>  join #opensips on freenode if you need
> more info.
>  
> >>> Thanks for the info. :)
> >>>
> >>> I want to stay away from software solutions
> however.
> >>>   
> >> Are there
> >> 
> >>> any hardware solutions? would a plain load
> balancer
> >>>   
> >> work?
> >> 
> >>> If we can't get it working with a LB
> we'll
> >>>   
> >> look at OpenSIPS,
> >> 
> >>> but I'd like to explore hardware options
> first.
> >>>
> >>> Thanks!
> >>>
> >>> --
> >>> Nitzan Kon, CEO
> >>> Future Nine Corporation
> >>> www.future-nine.com
> >>>   
> >
> >
> >
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> >   
> http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Load balancing Asterisk.

[Grey Man]

This baby talks about being able to do hardware SIP load balancing.

http://www.f5.com/news-press-events/press/2007/20070212.html

I've never used an f5 product so I can't provide any comments from
experience. I did look at an f5 load balancer product once and the
price was over 6 figures that was a few years ago though.

Regards,

Greyman.

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Re: [asterisk-users] Load balancing Asterisk.

[Alex Balashov]

The solution to make this work and still work "statelessly" is to hash 
various unique identifying bits of the SIP headers without maintaining 
transactional, session or dialog information as such.

SIP wrote:

> Unless the LB is SIP-aware, and can maintain a SIP session, I don't see
> how it would work. As the SIP command stream sends discrete commands,
> without some sort of basic level of session awareness, there's no
> guarantee over a reasonable-length call that the INVITE and BYE would
> even get sent to the same Asterisk box. If there are on-hold messages or
> transfers occurring, you add even more possibility of error into the
> mix.  Now... you could do some sort of VERY long session timeout, but
> overall, that's a hack that's going to drop your concurrent connection
> count faster than using a smaller box would.
> 
> I don't know of any functioning, SIP-aware load balancers at the moment.
> Doesn't mean they don't exist. I just can't think of any off the top of
> my head.
> 
> N.
> 
> 
> 
> Nitzan Kon wrote:
>> Alex,
>>
>> I realize and agree that "hardware" load balancers are actually
>> software based. I'm less concerned about that and more about the
>> general specs:
>>
>> Foundry ServerIron XL: rated for 1,000,000 concurrent connections
>> Linux box where OpenSIPS is sitting: rated for ...???
>>
>> Not to mention a simple rule on a load balancer would be much,
>> much easier to implement. All I need is IP-based load balancing
>> so installing and maintaining OpenSIPS is an overkill.
>>
>> Again, I appreciate the feedback but I am not asking nor looking
>> for a software solution. My question is simple:
>>
>> Will a HARDWARE load balancer work? any reason why it WON'T work?
>>
>> Thanks!
>>
>>
>> --- On Thu, 11/20/08, Alex Balashov <[EMAIL PROTECTED]> wrote:
>>
>>   
>>> What do you mean by "hardware" options?  There are
>>> no ASIC-assisted SIP load balancers out there.  :-)  The
>>> embedded "hardware-based" options are load
>>> balancers built just like PCs - often on top of a UNIX
>>> kernel - that run a software application-aware load
>>> balancing suite.
>>>
>>> Your best bet is a proxy for the round-robin part, and
>>> Linux-HA for the high availability of the proxy, as Grygoriy
>>> suggested.
>>>
>>> Nitzan Kon wrote:
>>>
>>> 
 --- On Thu, 11/20/08, Grygoriy Dobrovolskyy
   
>>> <[EMAIL PROTECTED]> wrote:
>>> 
> 2 openser servers with 3 ip adresses (1 virtual) +
> heartbeat to ensure the
> failover + watchdog to ensure if
> 
>>> opensips/kamalio/openser
>>> 
> crashes a nice
> failover & reboot, it is working stable here
> (dispatching to 10 servers +
> owners DID dispatch to their respective servers)
>
> join #opensips on freenode if you need more info.
> 
 Thanks for the info. :)

 I want to stay away from software solutions however.
   
>>> Are there
>>> 
 any hardware solutions? would a plain load balancer
   
>>> work?
>>> 
 If we can't get it working with a LB we'll
   
>>> look at OpenSIPS,
>>> 
 but I'd like to explore hardware options first.

 Thanks!

 --
 Nitzan Kon, CEO
 Future Nine Corporation
 www.future-nine.com
   
>>
>>
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-- 
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Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Re: [asterisk-users] Load balancing Asterisk.

[SIP]

Unless the LB is SIP-aware, and can maintain a SIP session, I don't see
how it would work. As the SIP command stream sends discrete commands,
without some sort of basic level of session awareness, there's no
guarantee over a reasonable-length call that the INVITE and BYE would
even get sent to the same Asterisk box. If there are on-hold messages or
transfers occurring, you add even more possibility of error into the
mix.  Now... you could do some sort of VERY long session timeout, but
overall, that's a hack that's going to drop your concurrent connection
count faster than using a smaller box would.

I don't know of any functioning, SIP-aware load balancers at the moment.
Doesn't mean they don't exist. I just can't think of any off the top of
my head.

N.



Nitzan Kon wrote:
> Alex,
>
> I realize and agree that "hardware" load balancers are actually
> software based. I'm less concerned about that and more about the
> general specs:
>
> Foundry ServerIron XL: rated for 1,000,000 concurrent connections
> Linux box where OpenSIPS is sitting: rated for ...???
>
> Not to mention a simple rule on a load balancer would be much,
> much easier to implement. All I need is IP-based load balancing
> so installing and maintaining OpenSIPS is an overkill.
>
> Again, I appreciate the feedback but I am not asking nor looking
> for a software solution. My question is simple:
>
> Will a HARDWARE load balancer work? any reason why it WON'T work?
>
> Thanks!
>
>
> --- On Thu, 11/20/08, Alex Balashov <[EMAIL PROTECTED]> wrote:
>
>   
>> What do you mean by "hardware" options?  There are
>> no ASIC-assisted SIP load balancers out there.  :-)  The
>> embedded "hardware-based" options are load
>> balancers built just like PCs - often on top of a UNIX
>> kernel - that run a software application-aware load
>> balancing suite.
>>
>> Your best bet is a proxy for the round-robin part, and
>> Linux-HA for the high availability of the proxy, as Grygoriy
>> suggested.
>>
>> Nitzan Kon wrote:
>>
>> 
>>> --- On Thu, 11/20/08, Grygoriy Dobrovolskyy
>>>   
>> <[EMAIL PROTECTED]> wrote:
>> 
 2 openser servers with 3 ip adresses (1 virtual) +
 heartbeat to ensure the
 failover + watchdog to ensure if
 
>> opensips/kamalio/openser
>> 
 crashes a nice
 failover & reboot, it is working stable here
 (dispatching to 10 servers +
 owners DID dispatch to their respective servers)

 join #opensips on freenode if you need more info.
 
>>> Thanks for the info. :)
>>>
>>> I want to stay away from software solutions however.
>>>   
>> Are there
>> 
>>> any hardware solutions? would a plain load balancer
>>>   
>> work?
>> 
>>> If we can't get it working with a LB we'll
>>>   
>> look at OpenSIPS,
>> 
>>> but I'd like to explore hardware options first.
>>>
>>> Thanks!
>>>
>>> --
>>> Nitzan Kon, CEO
>>> Future Nine Corporation
>>> www.future-nine.com
>>>   
>
>
>
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Re: [asterisk-users] Load balancing Asterisk.

[Alex Balashov]

Nitzan Kon wrote:

> Foundry ServerIron XL: rated for 1,000,000 concurrent connections
> Linux box where OpenSIPS is sitting: rated for ...???

Because OpenSER's load balancer is hash-based and not stateful, it is 
rated for far, far more than that.

-- 
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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Re: [asterisk-users] Load balancing Asterisk.

[Nitzan Kon]

Alex,

I realize and agree that "hardware" load balancers are actually
software based. I'm less concerned about that and more about the
general specs:

Foundry ServerIron XL: rated for 1,000,000 concurrent connections
Linux box where OpenSIPS is sitting: rated for ...???

Not to mention a simple rule on a load balancer would be much,
much easier to implement. All I need is IP-based load balancing
so installing and maintaining OpenSIPS is an overkill.

Again, I appreciate the feedback but I am not asking nor looking
for a software solution. My question is simple:

Will a HARDWARE load balancer work? any reason why it WON'T work?

Thanks!


--- On Thu, 11/20/08, Alex Balashov <[EMAIL PROTECTED]> wrote:

> What do you mean by "hardware" options?  There are
> no ASIC-assisted SIP load balancers out there.  :-)  The
> embedded "hardware-based" options are load
> balancers built just like PCs - often on top of a UNIX
> kernel - that run a software application-aware load
> balancing suite.
> 
> Your best bet is a proxy for the round-robin part, and
> Linux-HA for the high availability of the proxy, as Grygoriy
> suggested.
> 
> Nitzan Kon wrote:
> 
> > --- On Thu, 11/20/08, Grygoriy Dobrovolskyy
> <[EMAIL PROTECTED]> wrote:
> > 
> >> 2 openser servers with 3 ip adresses (1 virtual) +
> >> heartbeat to ensure the
> >> failover + watchdog to ensure if
> opensips/kamalio/openser
> >> crashes a nice
> >> failover & reboot, it is working stable here
> >> (dispatching to 10 servers +
> >> owners DID dispatch to their respective servers)
> >> 
> >> join #opensips on freenode if you need more info.
> > 
> > Thanks for the info. :)
> > 
> > I want to stay away from software solutions however.
> Are there
> > any hardware solutions? would a plain load balancer
> work?
> > 
> > If we can't get it working with a LB we'll
> look at OpenSIPS,
> > but I'd like to explore hardware options first.
> > 
> > Thanks!
> > 
> > --
> > Nitzan Kon, CEO
> > Future Nine Corporation
> > www.future-nine.com



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Re: [asterisk-users] Load balancing Asterisk.

[Jeff LaCoursiere]

Hardware solutions are of course simply packaged software solutions. 
Personally I would go with something that has this wonderful support base 
and quick solutions versus dealing with a vendor.  You did mention that 
price was a consideration, right?

j

On Thu, 20 Nov 2008, Nitzan Kon wrote:

> --- On Thu, 11/20/08, Grygoriy Dobrovolskyy <[EMAIL PROTECTED]> wrote:
>
>> 2 openser servers with 3 ip adresses (1 virtual) +
>> heartbeat to ensure the
>> failover + watchdog to ensure if opensips/kamalio/openser
>> crashes a nice
>> failover & reboot, it is working stable here
>> (dispatching to 10 servers +
>> owners DID dispatch to their respective servers)
>>
>> join #opensips on freenode if you need more info.
>
> Thanks for the info. :)
>
> I want to stay away from software solutions however. Are there
> any hardware solutions? would a plain load balancer work?
>
> If we can't get it working with a LB we'll look at OpenSIPS,
> but I'd like to explore hardware options first.
>
> Thanks!
>
> --
> Nitzan Kon, CEO
> Future Nine Corporation
> www.future-nine.com
>
> ___
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Load balancing Asterisk.

[Alex Balashov]

What do you mean by "hardware" options?  There are no ASIC-assisted SIP 
load balancers out there.  :-)  The embedded "hardware-based" options 
are load balancers built just like PCs - often on top of a UNIX kernel - 
that run a software application-aware load balancing suite.

Your best bet is a proxy for the round-robin part, and Linux-HA for the 
high availability of the proxy, as Grygoriy suggested.

Nitzan Kon wrote:

> --- On Thu, 11/20/08, Grygoriy Dobrovolskyy <[EMAIL PROTECTED]> wrote:
> 
>> 2 openser servers with 3 ip adresses (1 virtual) +
>> heartbeat to ensure the
>> failover + watchdog to ensure if opensips/kamalio/openser
>> crashes a nice
>> failover & reboot, it is working stable here
>> (dispatching to 10 servers +
>> owners DID dispatch to their respective servers)
>>
>> join #opensips on freenode if you need more info.
> 
> Thanks for the info. :)
> 
> I want to stay away from software solutions however. Are there
> any hardware solutions? would a plain load balancer work?
> 
> If we can't get it working with a LB we'll look at OpenSIPS,
> but I'd like to explore hardware options first.
> 
> Thanks!
> 
> --
> Nitzan Kon, CEO
> Future Nine Corporation
> www.future-nine.com
> 
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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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Re: [asterisk-users] Load balancing Asterisk.

[Nitzan Kon]

--- On Thu, 11/20/08, Grygoriy Dobrovolskyy <[EMAIL PROTECTED]> wrote:

> 2 openser servers with 3 ip adresses (1 virtual) +
> heartbeat to ensure the
> failover + watchdog to ensure if opensips/kamalio/openser
> crashes a nice
> failover & reboot, it is working stable here
> (dispatching to 10 servers +
> owners DID dispatch to their respective servers)
> 
> join #opensips on freenode if you need more info.

Thanks for the info. :)

I want to stay away from software solutions however. Are there
any hardware solutions? would a plain load balancer work?

If we can't get it working with a LB we'll look at OpenSIPS,
but I'd like to explore hardware options first.

Thanks!

--
Nitzan Kon, CEO
Future Nine Corporation
www.future-nine.com

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Re: [asterisk-users] Load balancing Asterisk.

[Grygoriy Dobrovolskyy]

2008/11/20 Nitzan Kon <[EMAIL PROTECTED]>

> Hello!
>
> We're looking for a solution to reliably load balance our
> Asterisk boxes. So far we've been using a hodge-podge of
> directing different services to different boxes/IPs, but
> eventually I'd like to consolidate things so we can present
> a single IP address to the outside world.
>
> My question is - how do we go about doing that? I've read
> a lot of things like load-balancing via DUNDi or OpenSER,
> but it seems to me like these approaches just add to the
> list of possible failures. In other words I'd like to avoid
> software solutions.
>
> Is it possible to just put Asterisk behind a load balancer?
> I imagine most of them are optimized for web traffic rather
> than UDP voice packets. Does that matter?
>
> Would any load balancer do - or only specific models will
> work? my guess is any model will work, but some of them may
> not be able to handle the load.
>
> Any recommended models?
>
> I know there are some fancy LBs out there that can actually
> load balance based on the SIP session rather than something
> like IP, but I'm afraid to even look at the price tag. I'm
> more than fine with balancing by user IP address instead -
> if that works. :)
>
> Would appreciate any comments or ideas.
>
> Thanks!
>
> --
> Nitzan Kon, CEO
> Future Nine Corporation
> www.future-nine.com
>
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2 openser servers with 3 ip adresses (1 virtual) + heartbeat to ensure the
failover + watchdog to ensure if opensips/kamalio/openser crashes a nice
failover & reboot, it is working stable here (dispatching to 10 servers +
owners DID dispatch to their respective servers)

join #opensips on freenode if you need more info.
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Re: [asterisk-users] load balancing

[CSB]

> 
> There are a few gotchas with a SIP Proxy the main one being transfers.
> But if you can get away with not allowing transfers then you are best
> to do so as the CDR's Asterisk produces are wrong anyway.
> 
What is the transfer problem? Is it the Asterisk native type using
features.conf or the SIP type using REFER that causes problems?

Cameron


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Re: [asterisk-users] load balancing

[CSB]

> 
> For outbound trunking we go directly from Asterisk to the terminating
> gateway no SIP Proxy involved. For inbound trunking we do go through
> the SIP Proxy for the same reason you get users to. Incoming calls are
> going to be more reliable if they are not tied to a single Asterisk
> server (I guess you could use SRV records for your Asterisk servers
> for inbound trunking as well but then you're kind of duplicating the
> role of the SIP proxy).
> 
How do you decide which Asterisk server to send the inbound call to? If the
Asterisk server that the user is registered on goes down what happens to the
inbound call?

Have you considered having the SIP clients register with the SIP proxy
rather than Asterisk or is that too difficult to get working?

Regards

Cameron


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Re: [asterisk-users] load balancing

[Grey Man]

On Fri,  29 Feb 2008 6:21 +0200, Yehavi Bourvine +972-8-9489444
<[EMAIL PROTECTED]> wrote:
>
>  See the discussion a few days ago. The Asterisk server saves the value of
>  SYSNAME (defined in asterisk.conf) in the field REGSERVER inside MySQL.
>
> Regards, __Yehavi:

Ahh that's handy. That would allow a half way solution between
multiple Asterisk servers and a SIP Proxy by utilising an AGI script
or database lookup in each Asterisk server's dialplan. When the
incoming calls arrive you'll be able to know which Asterisk server to
forward them to. You still have the problems about failing over the
Asterisk servers and putting two Asterisk servers in the media path is
always best avoided if possible although probably not a huge deal.

Actually from memory there is something in sip.conf regarding
autoregexten or something where when a SIP account registers with
Asterisk it automatically adds an entry to the dialplan. If this were
employed you could forward a call to all 4 Asterisk servers and only
the one that had the registered user would forward the call.

There are  lots of ways to skin the cat but the SIP Proxy is the best
way to utilise mutliple Asterisk servers when being used for SIP
calls.

Regards,

Greyman.

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Re: [asterisk-users] load balancing

[Grey Man]

On Fri, Feb 29, 2008 at 4:03 AM, Ron <[EMAIL PROTECTED]> wrote:
> Hi Greyman,
>
>  Should it look like this now? Can i use 2 SIP Proxies just to make sure
>  i have a backup? will that cause any problem again with regards to
>  calling extension to extension? Extensions will register on the asterisk
>  still? How about outbound calls to other SIP provider or a telcobridge,
>  SIP proxy will handle that also? Basically asterisk will ask SIP proxy
>  of everything? Will RTP stream still go thru asterisk?
>
>  Also, i plan on setting these up as a Virtual PBX for multiple offices,
>  which means company A can only use Trunks for A, B is for Trunk B etc
>  etc. Does outbound to trunks have any issues? or problem is just
>  basically calling extension to extension?
>
>
>  [other voip provider][telcobridge] -- [pstn]
> ||
>  
>  [  SIP Proxy   ]
>  
>
>| | |  |
>  [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ]
>| | | |
>  
>  | mysql cluster|
>  
>
>
>  Thanks
>
>  Regards,
>  Ron

Hi Ron,

Yep it starts to get tricky :).

There will be slight difference depending exactly on what you need to
accomplish. I work for a VoIP Proivder that provides services to users
in internet land so our set up is designed for that. If you've got
VPNs or are on a LAN things will be different.

Two SIP Proxy's are definitely a good idea, you can load balance your
users across them using DNS SRV records, DNS Round Robin, IP Load
Balancer (although then you prob should have two load balancers). If
you're just starting your build network build or only have < 1000
users the extra SIP Proxy should go to the bottom of the list. SIP
Proxy's such as OpenSER are pretty stable and don't do anywhere near
as much work as the media server. It's the fault tolerance on your
Asterisk servers that is the most critical thing. They do a lot more
work and in my experience with them (4+ years) they are a lot more
likely to crash than your SIP Proxy.

With two SIP Proxy's you have an additional problem in that now you
need to set the outboundproxy field in the Asterisk realtime database
to the value of the proxy the user agent came through. Asterisk can't
do that for you (as far as I know) so you could possibly use the SIP
Proxy to do registrations or use a custom SIP Registrar. Both are a
good idea as they take registration load away from Asterisk and this
can be VERY significant as your user base grows. We use a custom SIP
Registrar.

For outbound trunking we go directly from Asterisk to the terminating
gateway no SIP Proxy involved. For inbound trunking we do go through
the SIP Proxy for the same reason you get users to. Incoming calls are
going to be more reliable if they are not tied to a single Asterisk
server (I guess you could use SRV records for your Asterisk servers
for inbound trunking as well but then you're kind of duplicating the
role of the SIP proxy).

The RTP stream will always be between the users and Asterisk the SIP
Proxy is never invovled. If you send an RTP packet to a SIP Proxy and
it will just shake its head in an irritated manner and ignore you.

Regards,

Greyman.

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Re: [asterisk-users] load balancing

[Yehavi Bourvine +972-8-9489444]

>>  If i have this kind of setup, what do i need to make it's load balance.
>>
>>  [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ]
>>| | | |
>>  -
>>  | mysql cluster |
>>  -
>>
>>  I plan on doing it via DNS SRV only, but if a user register on asterisk
>>  1 how can users at asterisk 4 reach that user. Thank You
>>
>>  Regards,
>>  Ron
>>
>
> Hi Ron,
>
> If you're using realtime each Asterisk server will know where every
> user is irrespective of which Asterisk server they registered on. The
> problem is NAT, if a client is behind NAT and registers on server 1
> then server's 2,3 & 4 are unlikely to be able to get through to it.
> Last time I lookedthe Asterisk realtime engine doesn't record which
> server an account registered on in the database so the only option I

See the discussion a few days ago. The Asterisk server saves the value of
SYSNAME (defined in asterisk.conf) in the field REGSERVER inside MySQL.

Regards, __Yehavi:

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Re: [asterisk-users] load balancing

[Ron]

Hi Greyman,

Should it look like this now? Can i use 2 SIP Proxies just to make sure 
i have a backup? will that cause any problem again with regards to 
calling extension to extension? Extensions will register on the asterisk 
still? How about outbound calls to other SIP provider or a telcobridge, 
SIP proxy will handle that also? Basically asterisk will ask SIP proxy 
of everything? Will RTP stream still go thru asterisk?

Also, i plan on setting these up as a Virtual PBX for multiple offices, 
which means company A can only use Trunks for A, B is for Trunk B etc 
etc. Does outbound to trunks have any issues? or problem is just 
basically calling extension to extension?


[other voip provider][telcobridge] -- [pstn]
||

[  SIP Proxy   ]

   | | |  |
[ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ]
   | | | |

| mysql cluster| 



Thanks

Regards,
Ron


Grey Man wrote:
> On Fri, Feb 29, 2008 at 2:01 AM, Ron <[EMAIL PROTECTED]> wrote:
>> Hi All,
>>
>>  If i have this kind of setup, what do i need to make it's load balance.
>>
>>  [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ]
>>| | | |
>>  -
>>  | mysql cluster |
>>  -
>>
>>  I plan on doing it via DNS SRV only, but if a user register on asterisk
>>  1 how can users at asterisk 4 reach that user. Thank You
>>
>>  Regards,
>>  Ron
>>
> 
> Hi Ron,
> 
> If you're using realtime each Asterisk server will know where every
> user is irrespective of which Asterisk server they registered on. The
> problem is NAT, if a client is behind NAT and registers on server 1
> then server's 2,3 & 4 are unlikely to be able to get through to it.
> Last time I lookedthe Asterisk realtime engine doesn't record which
> server an account registered on in the database so the only option I
> can think of would be to forward an incoming call for a user to all 4
> of your Asterisk servers that way the call will get through but if
> they are not behind NAT they'll get 4 incoming calls.
> 
> Bascially it's messy using the set up you've got. What you really need
> is a SIP Proxy (assuming you're using SIP, if not it's even trickier).
> The SIP Proxy could load balance requests across your Asterisk
> servers. For calls destined for your users you can use the
> outboundproxy field in the sippeers table, by setting it to the IP
> address of your SIP Proxy server you can get Asterisk to forward all
> requests for a SIP account through the proxy (there is also an
> outboundproxyport setting but avoid it as it's been broken forever).
> 
> There are a few gotchas with a SIP Proxy the main one being transfers.
> But if you can get away with not allowing transfers then you are best
> to do so as the CDR's Asterisk produces are wrong anyway.
> 
> Regards,
> 
> Greyman.
> 
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> 

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Re: [asterisk-users] load balancing

[Grey Man]

On Fri, Feb 29, 2008 at 2:01 AM, Ron <[EMAIL PROTECTED]> wrote:
> Hi All,
>
>  If i have this kind of setup, what do i need to make it's load balance.
>
>  [ asterisk 1 ] -- [ asterisk 2 ] -- [ asterisk 3 ] -- [ asterisk 4 ]
>| | | |
>  -
>  | mysql cluster |
>  -
>
>  I plan on doing it via DNS SRV only, but if a user register on asterisk
>  1 how can users at asterisk 4 reach that user. Thank You
>
>  Regards,
>  Ron
>

Hi Ron,

If you're using realtime each Asterisk server will know where every
user is irrespective of which Asterisk server they registered on. The
problem is NAT, if a client is behind NAT and registers on server 1
then server's 2,3 & 4 are unlikely to be able to get through to it.
Last time I lookedthe Asterisk realtime engine doesn't record which
server an account registered on in the database so the only option I
can think of would be to forward an incoming call for a user to all 4
of your Asterisk servers that way the call will get through but if
they are not behind NAT they'll get 4 incoming calls.

Bascially it's messy using the set up you've got. What you really need
is a SIP Proxy (assuming you're using SIP, if not it's even trickier).
The SIP Proxy could load balance requests across your Asterisk
servers. For calls destined for your users you can use the
outboundproxy field in the sippeers table, by setting it to the IP
address of your SIP Proxy server you can get Asterisk to forward all
requests for a SIP account through the proxy (there is also an
outboundproxyport setting but avoid it as it's been broken forever).

There are a few gotchas with a SIP Proxy the main one being transfers.
But if you can get away with not allowing transfers then you are best
to do so as the CDR's Asterisk produces are wrong anyway.

Regards,

Greyman.

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Re: [asterisk-users] Load balancing SIP extensions.

[Yehavi Bourvine +972-8-9489444]

Hello,

  Here is how I do this. The prerequisits are:

- MySQL to hold the extensions realtime database. MySQL is synchronized
  among all servers using the Master/slave replication model.

- The phones are spread by some external algorithm over the Asterisk servers
  (statefull load balancer, statically defined in the config file of the
  phone, etc.).

The idea is to locate on which server the destination phone is registered and
redirect the call to it. For this:
/etc/asterisk/asterisk.conf has the parameter "sysname" set to its IP address
   (you can use also a DNS name, but I want to be independent of name
   resolution). This causes the server to set the field "regserver" to be
   saved in the MySQL database to the IP address of the server.

/etc/asterisk/extensions: The logic to check whether the value of "regsever"
   is different from "sysname" and if so - redirect the call. The code
   fragments are (I am using AEL):

   To get the regserver from the database:

 MYSQL(Query resID ${connid} SELECT regserver from sip_users where
name='${EXTEN}');
 MYSQL(Fetch FetchId ${resID} RegServer);
 MYSQL(Clear ${resID});

   so now RegServer contains the server where the phone is registered. Next:

if(("${DEVSTATE(SIP/${EXTEN})}" == "UNAVAILABLE") ||
   ("${DEVSTATE(SIP/${EXTEN})}" == "INVALID")) {
if("${SYSTEMNAME}" != "${RegServer}") {
Transfer(SIP/[EMAIL PROTECTED]);
return;
 };
};

I check for the device state so in case the phone has double registration
(primary and backup server) it will be processed localy.

Regards, __Yehavi:

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Re: [asterisk-users] load balancing SIP extensions

[Vieri]

--- Anthony Francis <[EMAIL PROTECTED]> wrote:

> Have you tried placing the sip registrations in a db
> using realtime?

I'm not that sure I want to use realtime because I
would then depend on the sql service never failing (I
could use clustered active-active MySQL but that
sounds overkill, or maybe not).

I'll take a look at the pdf link of the previous post.

Thanks



  

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Re: [asterisk-users] load balancing SIP extensions

[Raj Jain]

On Fri, 22 Feb 2008 19:44 +0200, Yehavi Bourvine +972-8-9489444 <
[EMAIL PROTECTED]> wrote:

> When a call arrives I check whether the REGSERVER coloumn is the same as
> the
> local server or not. If not, then there are two options:
>
> - Pass the call via IAX to the other servers; this makes both server
> process
>  the call and the audio.
>
> - Send a "refer" message to the caller to contact the other server.
>

You may actually want to use a "redirect" message for this (e.g SIP 302
response). In any case, traversing only one server in the signaling/media
path as opposed to two would generally seem more efficient.

-- 
Raj Jain

mailto:rj2807 at gmail dot com
sip:rjain at iptel dot org
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Re: [asterisk-users] load balancing SIP extensions

[Anthony Francis]

Vieri wrote:
> What I would like to do is have two identical *
> servers which accept registrations of sip extensions
> 4000-4999. 
>
> If I define a rrDNS or LinuxHA then I should have
> load-balanced registrations. 
>
> However, say ext. 4001 is registered on *1 and 4002 is
> registered on *2, if 4001 tries to call 4002 then I
> would like to do something like:
> - lookup 4002 on *1, try to establish a call if it's
> REGISTERED here
> - if it's not registered here then try to look it up
> on *2 and establish the call there
>
> I tried to use DUNDi on my local servers but I can't
> seem to make it work. Most howtos out there explain
> the use of DUNDi when the extension ranges do not
> overlap.
> So in my case where both *1 and *2 have the same local
> extension range 4XXX, can I go the DUNDi route or
> should I stop bashing my head on that and explore
> another solution?
>
> If someone has configured a similar system then I'd
> greatly appreciate some tips.
> I read a few dundi docs like
> http://www.voip-info.org/wiki-DUNDi.
>
> Thanks
>
>
>
>   
> 
> Looking for last minute shopping deals?  
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>   
Have you tried placing the sip registrations in a db using realtime?

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Re: [asterisk-users] load balancing SIP extensions

[Jared Bellows]

>
> I tried to use DUNDi on my local servers but I can't
> seem to make it work. Most howtos out there explain
> the use of DUNDi when the extension ranges do not
> overlap.
>

The following doc describes using the same extensions across multiple *
servers. It requires using realtime, but seems to do what you describe.

http://www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson_Whitepaper.pdf
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Re: [asterisk-users] load balancing SIP extensions

[Andres Jimenez]

On Fri, Feb 22, 2008 at 5:49 PM, Vieri <[EMAIL PROTECTED]> wrote:


>  Thanks. I'll try that although I hope it won't go into
>  an infinite loop between the 2 servers.

You are right. That could happen if the phone is not registered anywhere


You can put some security in the dialplan.
 if calls comes from IAX it means that PHONE is not registered in the
other server.
Just create special extensions to take the IAX calls (instead of GoTo):

PHONE  is 101

SERVER 1

exten => 101,1, Dial SIP/101
exten => 101,1, Dial IAX-SERVER2/55101

exten => 55101,1, Dial SIP/101
exten => 55101,1, Hangup

SERVER 2

exten => 101,1, Dial SIP/101
exten => 101,1, Dial IAX-SERVER1/55101

exten => 55101,1, Dial SIP/101
exten => 55101,1, Hangup


I hope it helps,


-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]

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Re: [asterisk-users] load balancing SIP extensions

[Andres Jimenez]

On Fri, Feb 22, 2008 at 5:49 PM, Vieri <[EMAIL PROTECTED]> wrote:
>
>  --- Andres Jimenez <[EMAIL PROTECTED]> wrote:
>
>  > On Fri, Feb 22, 2008 at 11:42 AM, Vieri
>  > <[EMAIL PROTECTED]> wrote:
>  >
>  > >  However, say ext. 4001 is registered on *1 and
>  > 4002 is
>  > >  registered on *2, if 4001 tries to call 4002 then
>  > I
>  > >  would like to do something like:
>  > >  - lookup 4002 on *1, try to establish a call if
>  > it's
>  > >  REGISTERED here
>  > >  - if it's not registered here then try to look it
>  > up
>  > >  on *2 and establish the call there
>  >
>  > You can do that using the dial plan.
>  >
>  > - Create an IAX link between both servers
>  > - DIal plan in both servers:
>  > First priority Dial using SIP/EXTEN
>  > Second priority IAX/EXTEN Dial IAX/EXTEN
>
>  Thanks. I'll try that although I hope it won't go into
>  an infinite loop between the 2 servers.
>
>
>
>
>   
> 
>  Looking for last minute shopping deals?
>  Find them fast with Yahoo! Search.  
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-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]

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Re: [asterisk-users] load balancing SIP extensions

[Vieri]

--- Andres Jimenez <[EMAIL PROTECTED]> wrote:

> On Fri, Feb 22, 2008 at 11:42 AM, Vieri
> <[EMAIL PROTECTED]> wrote:
> 
> >  However, say ext. 4001 is registered on *1 and
> 4002 is
> >  registered on *2, if 4001 tries to call 4002 then
> I
> >  would like to do something like:
> >  - lookup 4002 on *1, try to establish a call if
> it's
> >  REGISTERED here
> >  - if it's not registered here then try to look it
> up
> >  on *2 and establish the call there
> 
> You can do that using the dial plan.
> 
> - Create an IAX link between both servers
> - DIal plan in both servers:
> First priority Dial using SIP/EXTEN
> Second priority IAX/EXTEN Dial IAX/EXTEN

Thanks. I'll try that although I hope it won't go into
an infinite loop between the 2 servers.



  

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Re: [asterisk-users] load balancing SIP extensions

[Yehavi Bourvine +972-8-9489444]

> What I would like to do is have two identical *
> servers which accept registrations of sip extensions
> 4000-4999.
>
> If I define a rrDNS or LinuxHA then I should have
> load-balanced registrations.
>
> However, say ext. 4001 is registered on *1 and 4002 is
> registered on *2, if 4001 tries to call 4002 then I
> would like to do something like:
> - lookup 4002 on *1, try to establish a call if it's
> REGISTERED here
> - if it's not registered here then try to look it up
> on *2 and establish the call there
>...

I've tried doing something similar and came with two options. The common to
them is that I use MySQL for realtime extensions, and set "systemname"
parameter to the IP address of the server where the phone registers.

When a call arrives I check whether the REGSERVER coloumn is the same as the
local server or not. If not, then there are two options:

- Pass the call via IAX to the other servers; this makes both server process
  the call and the audio.

- Send a "refer" message to the caller to contact the other server.

I had this working in the lab but not in production yet. If you want the
dialplan code for this then email me.

   __Yehavi:

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Re: [asterisk-users] load balancing SIP extensions

[Andres Jimenez]

On Fri, Feb 22, 2008 at 11:42 AM, Vieri <[EMAIL PROTECTED]> wrote:

>  However, say ext. 4001 is registered on *1 and 4002 is
>  registered on *2, if 4001 tries to call 4002 then I
>  would like to do something like:
>  - lookup 4002 on *1, try to establish a call if it's
>  REGISTERED here
>  - if it's not registered here then try to look it up
>  on *2 and establish the call there

You can do that using the dial plan.

- Create an IAX link between both servers
- DIal plan in both servers:
First priority Dial using SIP/EXTEN
Second priority IAX/EXTEN



Dial IAX/EXTEN

-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]

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Re: [asterisk-users] Load Balancing over 2 E1 Lines

[Dovid B]

You can use the asterisk db for this. Simply set a variable to 1 or 0 if 1 
set to 0 and use g2 if 0 set to 1 and use g1.
- Original Message - 
From: "Andres Jimenez" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Wednesday, December 12, 2007 11:28 AM
Subject: Re: [asterisk-users] Load Balancing over 2 E1 Lines


> On Dec 12, 2007 8:08 AM, Eric Delaporte <[EMAIL PROTECTED]> wrote:
>
>
>> I read something about DIAL(Zap/r1/…) for using round robin, and it seems 
>> to
>> work.
> That will give you the same number of calls routed to each line
>
>> Is there any other possible way to make sure that all lines are used in 
>> the
>> same amount of minutes?
> You are going to need an AGI app or something storing how many minutes
> have been routed through each line and, on every call, choosing the
> less used one as the line to go out.
>
>
> -- 
> Andres Jimenez
>
> GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]
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Re: [asterisk-users] Load Balancing over 2 E1 Lines

[Marco Mouta]

Why not Random application available in Asterisk ?

quite simple I believe.

asterisk1*CLI> show application Random

  -= Info about application 'Random' =-

[Synopsis]
Conditionally branches, based upon a probability

[Description]
Random([probability]:[[context|]extension|]priority)
  probability := INTEGER in the range 1 to 100


best regards,
Marco Mouta

On Dec 12, 2007 8:08 AM, Eric Delaporte <[EMAIL PROTECTED]> wrote:

>  Hi @ all,
>
>
>
> i set a server to a costumer of mine with a TE207P for use with 2 E1
> Lines.
>
> I set them together into one group in zaptel/zapata.conf
>
>
>
> The point is now, the customer has a free-volumina of 60k minutes per
> month, per line.
>
> How can i make a kind of load balancing, that both lines will be trafficed
> the same way ?
>
>
>
> I read something about DIAL(Zap/r1/…) for using round robin, and it seems
> to work.
>
>
>
> Is there any other possible way to make sure that all lines are used in
> the same amount of minutes?
>
>
>
> Thanks in regard,
>
>
>
> Eric
>
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Re: [asterisk-users] Load Balancing over 2 E1 Lines

[Andres Jimenez]

On Dec 12, 2007 8:08 AM, Eric Delaporte <[EMAIL PROTECTED]> wrote:


> I read something about DIAL(Zap/r1/…) for using round robin, and it seems to
> work.
That will give you the same number of calls routed to each line

> Is there any other possible way to make sure that all lines are used in the
> same amount of minutes?
You are going to need an AGI app or something storing how many minutes
have been routed through each line and, on every call, choosing the
less used one as the line to go out.


-- 
Andres Jimenez

GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED]
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Re: [asterisk-users] Load balancing SIP trunks?

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Nicholas Blasgen wrote:
> I have 10 SIP trunks that I'd really like to round-robin load balance.
> Currently I have a macro that switches between available lines, but there
> really must be a function in Asterisk to do this on its own.  So my question
> is just that, are there any easy ways for Asterisk to either balance between
> SIP trunks or even just a built in function to find the next available SIP
> trunk.  I think using a macro to test the state of each trunk is silly, but
> it's the only method I've found.

If the SIP trunks are other Asterisk machines you have access to you
could use DUNDi:

http://www.asterisk.org/node/48321

- --
Kind Regards,

Matt Riddell
Director
___

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Re: [asterisk-users] Load balancing SIP trunks?

[Gordon Henderson]

On Wed, 15 Aug 2007, Nicholas Blasgen wrote:

> I have 10 SIP trunks that I'd really like to round-robin load balance.
> Currently I have a macro that switches between available lines, but there
> really must be a function in Asterisk to do this on its own.  So my question
> is just that, are there any easy ways for Asterisk to either balance between
> SIP trunks or even just a built in function to find the next available SIP
> trunk.  I think using a macro to test the state of each trunk is silly, but
> it's the only method I've found.

I wonder (and sometimes question!) the wisdom of putting everything into 
asterisk when it can be implemented in the dial-plan (or as I posted 
recently putting stuff out of the dialplan into AGI when it can be done in 
the dialplan) I'm sure there are cases where both are valid, but I'm a 
great believer in the KISS principle, and if we keep the "core" of 
asterisk clean and simple, then we can develop add-ons in the dialplan, or 
elsewhere...

(And after saying that, I have to say that the dial-plan programming 
language is one of the more esoteric programming languages I've used in 
the 26 years or so I've been programming!)


So, if we have sip-outX as out 10 sip trunks (0-9), then: (untested ;-)

[globals]
sipTrunk=9


  ...


[macro-dialSipTrunk]
exten => s,1,Noop(Dialling out via round-robin SIP trunk)
exten => s,n,Set(sipTrunk=$[${sipTrunk}+1])
exten => s,n,GotoIf($[${sipTrunk}=10],skip)
exten => s,n,Set(sipTrunk=0)
exten => s,n(skip)Noop(SIP dialling on trunk: ${sipTrunk})
exten => s,n,Dial(SIP/sip-out${sipTrunk},${ARG1},${ARG2})

... maybe stuff here to deal with result of the DIAL.
... eg. on congestion you might to jump back to step 2
... but if you did that then you might want to start a 2nd counter
... which when it reached 10, you're SOL and can return congestion
... to the caller.

Maybe I've just been writing too much dialplan stuff lately!!!

Gordon



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Re: [asterisk-users] Load balancing of IAX2

[Florian Overkamp]

Hi,

Kamran Ahmad wrote:

any idea how to loadbalance IAX2 trafic to multiple
asteirsk


Use app_random:

exten = _X.,2,Random(50:6)
exten = _X.,3,Dial(IAX2/server01/${EXTEN})
exten = _X.,4,Dial(IAX2/server02/${EXTEN})
exten = _X.,5,Goto(8)
exten = _X.,6,Dial(IAX2/server02/${EXTEN})
exten = _X.,7,Dial(IAX2/server01/${EXTEN})
exten = _X.,8,Congestion
exten = i,1,Congestion

Florian
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Re: [Asterisk-Users] Load balancing for each protocol

[Yves]

I was more thinking about a linux-ha heartbeat system between the 
redundant devices. The voicemail propagation is not a problem as I don't 
use it, and the dialplan is stored on a remote database (that's another 
problem to make redundant :) ).


The point I have to check now is how to configure GnuGK,SER & Asterisk 
to do round-robin routing.


I'm looking forward to see it work ... that's a really interesting project.

Yves.

Michiel van Baak wrote:

On 00:03, Mon 13 Jun 05, Matt Riddell wrote:


Yves wrote:


Hello,

I'm trying to find a good solution for load-balancing of several 
Asterisk box, for each VoIP protocol (IAX,SIP,H323). My idea is to have 
a front VoIP router (or several) who dispatches the calls of the 
different boxes.


This routing can be done with SER for SIP (redirect server) & GnuGK  for 
H323 (gkrouted 2). Now for IAX, is it possible to configure an Asterisk 
box to act as a "Redirect server" and not as a "proxy" ?


Any help would be appreciated, even you're telling me I'm going in the 
wrong direction.



Hi,

Are the asterisk boxes on the same subnet ?
If so, you could setup an OpenBSD CARP combi.
It will also provide failover for when one of the boxes go
down. All you have to do then is to setup the load balancing
stuff in PF and you're set.
What's next is the voicemail and dialplan propagation on all
the boxes that take part in the loadbalance setup. Guess you
already have something in mind for that cause your post is
talking about connections.



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Re: [Asterisk-Users] Load balancing for each protocol

[Michiel van Baak]

On 00:03, Mon 13 Jun 05, Matt Riddell wrote:
> Yves wrote:
> >Hello,
> >
> >I'm trying to find a good solution for load-balancing of several 
> >Asterisk box, for each VoIP protocol (IAX,SIP,H323). My idea is to have 
> >a front VoIP router (or several) who dispatches the calls of the 
> >different boxes.
> >
> >This routing can be done with SER for SIP (redirect server) & GnuGK  for 
> >H323 (gkrouted 2). Now for IAX, is it possible to configure an Asterisk 
> >box to act as a "Redirect server" and not as a "proxy" ?
> >
> >Any help would be appreciated, even you're telling me I'm going in the 
> >wrong direction.

Hi,

Are the asterisk boxes on the same subnet ?
If so, you could setup an OpenBSD CARP combi.
It will also provide failover for when one of the boxes go
down. All you have to do then is to setup the load balancing
stuff in PF and you're set.
What's next is the voicemail and dialplan propagation on all
the boxes that take part in the loadbalance setup. Guess you
already have something in mind for that cause your post is
talking about connections.

-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Two of the most famous products of Berkeley are LSD and BSD. I don't think 
that this is a coincidence."

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Re: [Asterisk-Users] Load balancing for each protocol

[Matt Riddell]

Yves wrote:

Hello,

I'm trying to find a good solution for load-balancing of several 
Asterisk box, for each VoIP protocol (IAX,SIP,H323). My idea is to have 
a front VoIP router (or several) who dispatches the calls of the 
different boxes.


This routing can be done with SER for SIP (redirect server) & GnuGK  for 
H323 (gkrouted 2). Now for IAX, is it possible to configure an Asterisk 
box to act as a "Redirect server" and not as a "proxy" ?


Any help would be appreciated, even you're telling me I'm going in the 
wrong direction.


You're looking for notransfer=no (the default behaviour for IAX2)

--
Cheers,

Matt Riddell
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Re: [Asterisk-Users] Load Balancing b/w 2 asterisk servers usingSIPload balancer

[Matthew Boehm]

Jagan Mohan wrote:
>I need to do load balancing only for the following functionalities:
> 1) Registration of SIP clients to * servers.
> 2) Load balancing of the INVITEs from SIP clients to different *
> servers.
>
> I'm not interested in supporting the features, which you have
> mentioned below. I'm not aware how the below mentioned features would
> be suppported in load balancing.
>
> -Jagan
>
>
> On Fri, 11 Mar 2005 08:54:39 -0600, Matthew Boehm
> <[EMAIL PROTECTED]> wrote:
>> How do you plan on supporting call queues, parking and agents with 2
>> * servers? This is something that has blocked us from being able to
>> do our own SER-based load balancing.
>>
>> -Matthew
>>
>> Jagan Mohan wrote:
>>> Hi,
>>>
>>>   I'm trying to do load balancing between 2 asterisk servers using
>>> SIP load balancer, provided by http://www.vovida.org
>>>
>>>   I used the following options on lbproxy, but I get the below
>>> message continuously.
>>>
>>> ./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy
>>> A2
>>>
>>> "No proxies are up - can not send message to anyone"
>>>
>>> Xlite is not able to register to the asterisk server.
>>>
>>> Is there anything which needs to be tweaked on Asterisk side to get
>>> this working? Please help.
>>>
>>> Thanks,
>>> Jagan

SER can do what you want.

Google for some example SER configs. I found a nice one, but don't have the
link here at work.

-Matthew

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Re: [Asterisk-Users] Load Balancing b/w 2 asterisk servers using SIPload balancer

[Jagan Mohan]

   I need to do load balancing only for the following functionalities:
1) Registration of SIP clients to * servers.
2) Load balancing of the INVITEs from SIP clients to different * servers.

I'm not interested in supporting the features, which you have
mentioned below. I'm not aware how the below mentioned features would
be suppported in load balancing.

-Jagan


On Fri, 11 Mar 2005 08:54:39 -0600, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> How do you plan on supporting call queues, parking and agents with 2 *
> servers? This is something that has blocked us from being able to do our own
> SER-based load balancing.
> 
> -Matthew
> 
> Jagan Mohan wrote:
> > Hi,
> >
> >   I'm trying to do load balancing between 2 asterisk servers using SIP
> > load balancer, provided by http://www.vovida.org
> >
> >   I used the following options on lbproxy, but I get the below message
> > continuously.
> >
> > ./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy
> > A2
> >
> > "No proxies are up - can not send message to anyone"
> >
> > Xlite is not able to register to the asterisk server.
> >
> > Is there anything which needs to be tweaked on Asterisk side to get
> > this working? Please help.
> >
> > Thanks,
> > Jagan
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Re: [Asterisk-Users] Load Balancing b/w 2 asterisk servers using SIPload balancer

[Matthew Boehm]

How do you plan on supporting call queues, parking and agents with 2 *
servers? This is something that has blocked us from being able to do our own
SER-based load balancing.

-Matthew

Jagan Mohan wrote:
> Hi,
>
>   I'm trying to do load balancing between 2 asterisk servers using SIP
> load balancer, provided by http://www.vovida.org
>
>   I used the following options on lbproxy, but I get the below message
> continuously.
>
> ./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy
> A2
>
> "No proxies are up - can not send message to anyone"
>
> Xlite is not able to register to the asterisk server.
>
> Is there anything which needs to be tweaked on Asterisk side to get
> this working? Please help.
>
> Thanks,
> Jagan
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Re: [Asterisk-Users] load balancing 20 asterisk servers

[Roy Sigurd Karlsbakk]

For growth, all you do is add more SER and more Asterisk boxes.
Are you sure one SER box won't be sufficient?
Makes sense to me to have these TWO - you can take one of those 
off-line
without interrupting service, and that's the entire idea of this
discussion, isn't it? ;->
Yeah
Get two cisco load balancers. One of them _will_ fail.
Put them in front of two SER boxes, crossover connected.
Get a gigabit switch with a good backplane
Put your  asterisk servers behind the SER
Play :)
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Re: [Asterisk-Users] load balancing 20 asterisk servers

[Philipp von Klitzing]

Hi!

> > http://drmac.homeunix.net/images/load_balancer.jpg
> You won't need the second balancer. SER can do that.

Seconded.

> > For growth, all you do is add more SER and more Asterisk boxes.
> Are you sure one SER box won't be sufficient?

Makes sense to me to have these TWO - you can take one of those off-line 
without interrupting service, and that's the entire idea of this 
discussion, isn't it? ;->

Cheers, Philipp


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Re: [Asterisk-Users] load balancing 20 asterisk servers

[Roy Sigurd Karlsbakk]

I beleive what you're looking for is a scalable SIP proxy, like SER :)
That way, all clients registers to SER and SER redirects the caller to
one of the asterisk boxes. Search the wiki at voip-info.org for
"asterisk at large" :)
Yes, that is one of the many pages I've read. But we still have a
problem. Take a look at this image to get a better idea of my "end 
goal".

http://drmac.homeunix.net/images/load_balancer.jpg
You won't need the second balancer. SER can do that.
For growth, all you do is add more SER and more Asterisk boxes.
Are you sure one SER box won't be sufficient?
But if Asterisk won't work correctly with the load balancing due to 
packet
movement, then I need to approach this differently.
perhaps setting up a second SER box for failover will do? just failover 
with heartbeat or something...

roy
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Re: [Asterisk-Users] load balancing 20 asterisk servers

[Ariel Mónaco]

Matthew, i think it would 
be convenient that you use dns round-robin for load balancing, registering the 
clients
against Ser or Asterisk boxes. Greetings. 
Ariel.
 
- Original Message - 
From: "Matthew Boehm" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial 
Discussion" <asterisk-users@lists.digium.com>
Sent: Wednesday, February 02, 2005 5:43 
PM
Subject: Re: [Asterisk-Users] load balancing 20 
asterisk servers
> > I beleive what you're looking for is a scalable SIP proxy, like 
SER :)> > That way, all clients registers to SER and SER redirects the 
caller to> > one of the asterisk boxes. Search the wiki at 
voip-info.org for> > "asterisk at large" :)> > 
    Yes, that is one of the many pages I've read. But we still 
have a> problem. Take a look at this image to get a better idea of my 
"end goal".> >     http://drmac.homeunix.net/images/load_balancer.jpg> > For growth, all you do is add more SER and more 
Asterisk boxes.> > But if Asterisk won't work correctly with the 
load balancing due to packet> movement, then I need to approach this 
differently.> > -Matthew> > 
___> Asterisk-Users mailing 
list> Asterisk-Users@lists.digium.com> http://lists.digium.com/mailman/listinfo/asterisk-users> To UNSUBSCRIBE or update options visit:> 
   http://lists.digium.com/mailman/listinfo/asterisk-users> > > > -- > No virus found in 
this incoming message.> Checked by AVG Anti-Virus.> Version: 
7.0.300 / Virus Database: 265.8.3 - Release Date: 31/01/2005> > 

No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.8.3 - Release Date: 31/01/2005
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Re: [Asterisk-Users] load balancing 20 asterisk servers

[Matthew Boehm]

> I beleive what you're looking for is a scalable SIP proxy, like SER :)
> That way, all clients registers to SER and SER redirects the caller to
> one of the asterisk boxes. Search the wiki at voip-info.org for
> "asterisk at large" :)

Yes, that is one of the many pages I've read. But we still have a
problem. Take a look at this image to get a better idea of my "end goal".

http://drmac.homeunix.net/images/load_balancer.jpg

For growth, all you do is add more SER and more Asterisk boxes.

But if Asterisk won't work correctly with the load balancing due to packet
movement, then I need to approach this differently.

-Matthew

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Re: [Asterisk-Users] load balancing 20 asterisk servers

[Roy Sigurd Karlsbakk]

I've read several other emails and pages on the wiki but none give any
deffinate answers. if you have 20 asterisk servers each with 4 pri's, 
all
running RealTime Extensions and RealTime SIPBuddies from the same MySQL
server, what prevents you from putting all 20 servers behind a single 
load
balancer? That way all of your UA's can use the same IP to register 
to; vs
maintaining which customer is assigned to which machine.

I beleive what you're looking for is a scalable SIP proxy, like SER :)
That way, all clients registers to SER and SER redirects the caller to 
one of the asterisk boxes. Search the wiki at voip-info.org for 
"asterisk at large" :)

roy
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Re: [Asterisk-Users] load balancing 20 asterisk servers

[Rich Adamson]

> > I'd have to guess that registrations would be the tricky part of an
> > implementation simply because there are so many variations of that.
> 
> Actually, this is the easiest part. It doesn't matter how often a UA
> registers nor does it matter to which of the 20 servers handles the
> registration since all servers share the same database tables.

The actual registration interaction (those few packets) I wouldn't expect
to be an issue either. My comment was more oriented towards the
more real time interactions of call handling shortly after the registration
process, and what _might_ be impacted in terms of those calls. By that
I mean, a call (in either direction) starts out using sip to negotiate
an rtp session. If a sticky bit is applied, then all traffic from a
specific IP address is essentially assigned to a single server. If the
sticky bit is not used, then the load balancer _may_ send the initial
rtp data flow to a different server, thus breaking the sip negotiation
process (the call won't get set up).

> > Meetme (as well as other * functions) would certainly need to be
> > well thought out before considering a balancer. (Eg, where does the
> > customer's voicemail actually reside?
> 
> Voicemail is not a problem. Again, all voicemails are stored in database
> including the audio portions.
> 
> 
> The problem with MeetMe conferences still bugs me.
> 
> I was un-aware that UDP had "sessions".

I was using the term more generically. The application assumes udp
sessions exist; layer-three doesn't contain session data. In other
words, from a load balancer perspective, there is noting in an
individual packet for it to recognize a session. Therefore, the load
balancer has to keep track of these so called sessions at layer-3
only (eg, ip address). The balancer (again) in watching/balancing
incoming connections and doesn't really know about outbound data.

So, if server1 was _completing_ 90% of all outgoing calls, how would
the balancer know that it should not allocate another _incoming_ rtp
session to that server? (Maybe a poor example, but think that process
through and I'm not sure a load balancer can truly deal with the
problem.)

If the sticky bit kind of thing is applied, then a business customer
with an * box will send all calls to the same itsp server.

Analyzing the call setups from both an incoming and outgoing perspective
becomes very important. Separating the two is certainly doable, but
more thought has to be given to the different sip setup states to
ensure the process flows correctly and still load balances.


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Re: [Asterisk-Users] load balancing 20 asterisk servers

[Matthew Boehm]

> I'd have to guess that registrations would be the tricky part of an
> implementation simply because there are so many variations of that.

Actually, this is the easiest part. It doesn't matter how often a UA
registers nor does it matter to which of the 20 servers handles the
registration since all servers share the same database tables.

> Meetme (as well as other * functions) would certainly need to be
> well thought out before considering a balancer. (Eg, where does the
> customer's voicemail actually reside?

Voicemail is not a problem. Again, all voicemails are stored in database
including the audio portions.


The problem with MeetMe conferences still bugs me.

I was un-aware that UDP had "sessions".

Keep in mind that all 20 servers are basically clones of eachother. They all
share the same extensions.conf, sip.conf, voicemail.conf (all via RealTime).

-Matthew

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Re: [Asterisk-Users] load balancing 20 asterisk servers

[Rich Adamson]

Inline...

> I've read several other emails and pages on the wiki but none give any
> deffinate answers. if you have 20 asterisk servers each with 4 pri's, all
> running RealTime Extensions and RealTime SIPBuddies from the same MySQL
> server, what prevents you from putting all 20 servers behind a single load
> balancer? That way all of your UA's can use the same IP to register to; vs
> maintaining which customer is assigned to which machine.

Load balancers vary rather dramatically in exactly how they fucntion.
Some work at layer-2, others at layer-3, and some at layers above.
Some include a small app that executes on each server to monitor
processor utilitization, etc, communicating key parameters to the
balancer. I've not tried any of these with *, but I'd have to guess
that selecting a specific model that properly handles udp sessions
(including variable length registrations) might require some resarch
that is a little more extensive then what the causual observer might
guess.
 
> perhaps its just that i am not that familiar with load balancers. i was
> under the impression that a load balancer could/would send each recieved
> packet to a different server.

That assumption is basically correct, however most balancers will 
maintain some sort of session-oriented function that will try to keep
the flow directed to the server for which it first assigned the traffic,
keeping in mind that it balances 'inbound' data flows not outbound
traffic.

> this doesn't matter in the case of register requests since all asterisk
> boxes share same SIP registry database.

I'd have to guess that registrations would be the tricky part of an
implementation simply because there are so many variations of that.
(Eg, some devices/systems register every minute while others every
hour, and about everything in between. From a load balancer perspective,
does the first registration look any different from the second and
follow-on registrations, and would the balancer consider those as
the same or different end points? Might that cause a flurry of other
system activities that have not been considered?)
 
> but what about invite requests and the rtp stream? you would have a majorly
> broken conversation if each packet in the rtp stream went to a different
> asterisk box.

No, there are parameters available to cause all packets associated with
a session to stay with the initial system and not try to load balance
on a per-packet basis. Some balancers refer to the parameter as a
sticky bit. However, careful thought has to be given to how the balancer
functions when an rtp session is _first_ initiated from an internal
* system verses a remote * system as an example.
 
> or are load balancers SIP aware? or is there some sort of session control
> that the balancer is aware of and will send all packets in a "sip session"
> to the same asterisk box?

I have not heard of any balancer being sip-aware, and would suspect that
some of the nat-like issues probably apply to load balancers. It certainly
would _not_ be cool for the balancer to treat the rtp session setup as 
a new session when its tied directly to the sip negotitation process.
 
> and then what about meet me conferences? if 10 UA's all dial a conference
> DID number and all 10 get balanced to 10 different servers then they are all
> sitting in seperate rooms right?

Meetme (as well as other * functions) would certainly need to be
well thought out before considering a balancer. (Eg, where does the
customer's voicemail actually reside? How much inter-system traffic
would generated because various resources are scattered across
multiple systems such as meetme sessions, etc?)

There are obviously a lot of folks on the list that shot from the
hip with little or no practical experience. Load balancing will require
a little more well thought out engineering then that. I'm not sure
the actual realtime * implementation is at a point where these issues
can be addressed today.


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Re: [Asterisk-Users] load balancing 20 asterisk servers

[Shaun Dwyer]

Hi,
You may want to look into LVS (Linux Virtual Server). It allows load 
ballancing in a highly configurable way.
http://www.linuxvirtualserver.org/

We use it on our web and mail server to load ballance across multiple 
hosts. The way we have it configured
it will maintain a session for 15 minutes between a client and a 
specific server. So long as you have
qualify=yes in your configuration files, each client will continue to 
talk to the one server until they are turned off/
deactivated for at least 15 minutes (or whatever time period you 
configure into it). I've not tested LVS with
Asterisk, but it may be the right direction for you to take.

Cheers,
-Shaun
Matthew Boehm wrote:
I've read several other emails and pages on the wiki but none give any
deffinate answers. if you have 20 asterisk servers each with 4 pri's, all
running RealTime Extensions and RealTime SIPBuddies from the same MySQL
server, what prevents you from putting all 20 servers behind a single load
balancer? That way all of your UA's can use the same IP to register to; vs
maintaining which customer is assigned to which machine.
perhaps its just that i am not that familiar with load balancers. i was
under the impression that a load balancer could/would send each recieved
packet to a different server.
this doesn't matter in the case of register requests since all asterisk
boxes share same SIP registry database.
but what about invite requests and the rtp stream? you would have a majorly
broken conversation if each packet in the rtp stream went to a different
asterisk box.
or are load balancers SIP aware? or is there some sort of session control
that the balancer is aware of and will send all packets in a "sip session"
to the same asterisk box?
and then what about meet me conferences? if 10 UA's all dial a conference
DID number and all 10 get balanced to 10 different servers then they are all
sitting in seperate rooms right?
hints, opinions, facts...all welcome and appreciated.
-Matthew
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