[Freeswitch-users] Scale UP Freeswitch

Steven Ayre steveayre at gmail.com
Mon Dec 13 12:41:19 MSK 2010 

You can use X-Auth-IP with a FS-FS call too:

Customer --> FS1 --> FS2
FS1 = front FS
FS2 = media server

1. Create a proxy ACL on FS2
2. Add the IP of FS1 to that ACL
3. On FS1 do this in the dialplan:

<extension ...>
  <condition ...>
    <action application="set" data="sip_h_X-Auth-IP=${network_addr}" />
    <action application="bridge" data="sofia/gateway/fs2/..." />
  </condition>
</extension>

FS2 will then be able to use the customer's IP in ACLs, user directory, etc.

Remember to either set inbound_bypass_media=true on the sip profile,
or <action application="set" data="bypass_media=true" /> in dialplan
before the bridge.

-Steve



On 12 December 2010 21:32, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> hmmm... so doing that will also require X-Auth-IP, right or something more
> tricky can be done?
> On Sun, Dec 12, 2010 at 9:51 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> It is, but it relies on the caller supporting 3xx. They might not
>> handle the redirect.
>>
>> A lot won't because you could redirect them to anywhere, so lots of
>> implementations will ignore the 3xx. FreeSWITCH for instance can
>> either ignore a 3xx or will send the call back into the dialplan.
>>
>> I think you'll have more success having a FS server in front of the
>> others and bridging the call through to each server. If you set
>> inbound_bypass_media=true on the SIP profile, the RTP media will
>> bypass that server and go directly between the caller and the other FS
>> box. That means that the call won't be using any CPU since it'll only
>> wake up when a SIP packet is being sent/received. You'll still be
>> creating a session through so it'll still be allocating memory to the
>> call, a SIP proxy would use fewer resources.
>>
>> -Steve
>>
>>
>> On 12 December 2010 19:28, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
>> > 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 at 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 at 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 at 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 at 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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>> >> >> >
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>> >> >> >
>> >> >> >
>> >> >>
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