Thanks Steve for suggestion, i'll check X-Auth-IP, its new for me.

Since we are talking about HA options... Is it practically doable use it:

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2

The idea is to run one FS box (Redirect-FS) in front of several FS boxes which redirect the call to active/available FS. If we make some script on redirect FS to count the active calls on media FSes and rearrange the order of redirect then loadbalacing can also be done. 

...possible?

On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <steveayre@gmail.com> wrote:
> 1. i am thinking to use kamailo in front of FS boxes, is there any
> difference between kamailo and opensips?

They're both forks of OpenSER so for the most part there's little difference.

There are some small differences though since the fork. For example,
opensips has a load_balancer module which kamalio does not (kamalio
can still do load balancing but has a different interface to do so).

> 2. if kamailo or opensips is running in front of FS, then will it send call
> to FS with original customer ip? so i can do billing etc on FS box
> -> actually i do IP based authentication and also ip based billing on FS
> box, so in case, i recieve kamailo ip on FS box then i'll loose the original
> customer overview.

It will appear coming from the proxy IP. But there is a workaround.
Configure a proxy ACL on the SIP profile and add your proxy IP to it.
Then adjust your proxy routing rules so that it adds a X-Auth-IP
header that contains the original IP.
Anything coming from anything in the proxy ACL is trusted and FS will
use the value from X-Auth-IP (if it exists).

-Steve




On 11 December 2010 14:00, Saeed Ahmed <saeedahmad1981@gmail.com> wrote:
> Hi,
>
> 1. i am thinking to use kamailo in front of FS boxes, is there any
> difference between kamailo and opensips?
>
> 2. if kamailo or opensips is running in front of FS, then will it send call
> to FS with original customer ip? so i can do billing etc on FS box
> -> actually i do IP based authentication and also ip based billing on FS
> box, so in case, i recieve kamailo ip on FS box then i'll loose the original
> customer overview.
>
> thanks
> On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <steveayre@gmail.com> wrote:
>>
>> There are a few performance tweaking tips at
>> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations.
>>
>> Yes a Sangoma card will reduce your CPU load since transcoding won't
>> be done on the CPU any longer, that will then mean there's more CPU
>> available so you'll be able to handle more calls.
>>
>> However, if you're looking to increase your number of calls then you
>> probably want a cluster of servers as Juan pointed out.
>>
>> It'll mean you can increase the capacity by adding extra servers, so
>> there'd no longer be a limit to the number of calls you could handle
>> (just add another server).
>>
>> It'll also make maintenance easier, as you'll be able to pull a server
>> from service for updates etc while traffic continues to run on the
>> other servers. Maintenance won't mean a service outage.
>>
>> If you're handling that many calls then additional servers would make
>> your service more reliable. If a server crashes you'll still have the
>> calls running on the other servers while you're fixing the problem so
>> you won't have a complete outage. If FS is behind a load balancer then
>> your customers might not even notice anything apart from a few dropped
>> calls.
>>
>> There's http://wiki.freeswitch.org/wiki/Freeswitch_HA which will
>> attempt to continue calls if FS crashes and restarts, but I think
>> that's only for SIP-SIP not SIP-ISDN.
>>
>> -Steve
>>
>>
>>
>>
>> On 7 December 2010 12:26, Stephen Wilde <wstephen80@gmail.com> wrote:
>> > Hi,
>> > I have one server running Freeswitch with some ISDN connections (via
>> > FreeTDM+Sangoma boards) and some SIP connections with service providers
>> > and
>> > customer.
>> > The usage of Freeswitch is as switching so it "bridge" each incoming
>> > call to
>> > a new outgoing call.
>> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding.
>> > Now the number of call is grow up and also the CPU load is a little high
>> > so
>> > I have the necessity to scale UP my Freeswitch to handle more calls:
>> > what is
>> > the best way to do that?
>> > My first idea is to use a Sangoma D500 board to reduce the CPU load. Can
>> > be
>> > this a solution?
>> > There are different way to scale UP?
>> > Thanks in advance,
>> > Stephen
>> >
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