[Freeswitch-users] Scale UP Freeswitch
Stephen Wilde
wstephen80 at gmail.com
Mon Dec 13 00:09:51 MSK 2010
The FS in front that doesn't handle the media is a good solution for me!
The incoming SIP call can be bridged with a simple round robin rule to "FS
media server" and a called prefix number can be added by front FS (and
removed by FS media server) to propagate the "source IP address".
Thank you for this info: I'll do some test on this solution.
Stephen
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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