Thanks Steve for suggestion, i'll check <span class="Apple-style-span" style="font-family: arial, sans-serif; font-size: 13px; border-collapse: collapse; ">X-Auth-IP, its new for me.</span><div><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;"><br>
</span></font></div><div><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;">Since we are talking about HA options... Is it practically doable use it:<br>
<br><a href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2">http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2</a></span></font></div><div><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;"><br>
</span></font></div><div><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;"><a href="http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect#Example_2"></a>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. </span></font></div>
<div><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;"><br></span></font></div><div><font class="Apple-style-span" face="arial, sans-serif"><span class="Apple-style-span" style="border-collapse: collapse;">...possible?<br>
</span></font><div><br><div class="gmail_quote">On Sun, Dec 12, 2010 at 12:23 PM, Steven Ayre <span dir="ltr"><<a href="mailto:steveayre@gmail.com">steveayre@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div class="im">> 1. i am thinking to use kamailo in front of FS boxes, is there any<br>
> difference between kamailo and opensips?<br>
<br>
</div>They're both forks of OpenSER so for the most part there's little difference.<br>
<br>
There are some small differences though since the fork. For example,<br>
opensips has a load_balancer module which kamalio does not (kamalio<br>
can still do load balancing but has a different interface to do so).<br>
<div class="im"><br>
> 2. if kamailo or opensips is running in front of FS, then will it send call<br>
> to FS with original customer ip? so i can do billing etc on FS box<br>
> -> actually i do IP based authentication and also ip based billing on FS<br>
> box, so in case, i recieve kamailo ip on FS box then i'll loose the original<br>
> customer overview.<br>
<br>
</div>It will appear coming from the proxy IP. But there is a workaround.<br>
Configure a proxy ACL on the SIP profile and add your proxy IP to it.<br>
Then adjust your proxy routing rules so that it adds a X-Auth-IP<br>
header that contains the original IP.<br>
Anything coming from anything in the proxy ACL is trusted and FS will<br>
use the value from X-Auth-IP (if it exists).<br>
<br>
-Steve<br>
<div><div></div><div class="h5"><br>
<br>
<br>
<br>
On 11 December 2010 14:00, Saeed Ahmed <<a href="mailto:saeedahmad1981@gmail.com">saeedahmad1981@gmail.com</a>> wrote:<br>
> Hi,<br>
><br>
> 1. i am thinking to use kamailo in front of FS boxes, is there any<br>
> difference between kamailo and opensips?<br>
><br>
> 2. if kamailo or opensips is running in front of FS, then will it send call<br>
> to FS with original customer ip? so i can do billing etc on FS box<br>
> -> actually i do IP based authentication and also ip based billing on FS<br>
> box, so in case, i recieve kamailo ip on FS box then i'll loose the original<br>
> customer overview.<br>
><br>
> thanks<br>
> On Tue, Dec 7, 2010 at 2:31 PM, Steven Ayre <<a href="mailto:steveayre@gmail.com">steveayre@gmail.com</a>> wrote:<br>
>><br>
>> There are a few performance tweaking tips at<br>
>> <a href="http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations" target="_blank">http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations</a>.<br>
>><br>
>> Yes a Sangoma card will reduce your CPU load since transcoding won't<br>
>> be done on the CPU any longer, that will then mean there's more CPU<br>
>> available so you'll be able to handle more calls.<br>
>><br>
>> However, if you're looking to increase your number of calls then you<br>
>> probably want a cluster of servers as Juan pointed out.<br>
>><br>
>> It'll mean you can increase the capacity by adding extra servers, so<br>
>> there'd no longer be a limit to the number of calls you could handle<br>
>> (just add another server).<br>
>><br>
>> It'll also make maintenance easier, as you'll be able to pull a server<br>
>> from service for updates etc while traffic continues to run on the<br>
>> other servers. Maintenance won't mean a service outage.<br>
>><br>
>> If you're handling that many calls then additional servers would make<br>
>> your service more reliable. If a server crashes you'll still have the<br>
>> calls running on the other servers while you're fixing the problem so<br>
>> you won't have a complete outage. If FS is behind a load balancer then<br>
>> your customers might not even notice anything apart from a few dropped<br>
>> calls.<br>
>><br>
>> There's <a href="http://wiki.freeswitch.org/wiki/Freeswitch_HA" target="_blank">http://wiki.freeswitch.org/wiki/Freeswitch_HA</a> which will<br>
>> attempt to continue calls if FS crashes and restarts, but I think<br>
>> that's only for SIP-SIP not SIP-ISDN.<br>
>><br>
>> -Steve<br>
>><br>
>><br>
>><br>
>><br>
>> On 7 December 2010 12:26, Stephen Wilde <<a href="mailto:wstephen80@gmail.com">wstephen80@gmail.com</a>> wrote:<br>
>> > Hi,<br>
>> > I have one server running Freeswitch with some ISDN connections (via<br>
>> > FreeTDM+Sangoma boards) and some SIP connections with service providers<br>
>> > and<br>
>> > customer.<br>
>> > The usage of Freeswitch is as switching so it "bridge" each incoming<br>
>> > call to<br>
>> > a new outgoing call.<br>
>> > SIP calls use G.729 and ISDN calls use ALaw for voice encoding.<br>
>> > Now the number of call is grow up and also the CPU load is a little high<br>
>> > so<br>
>> > I have the necessity to scale UP my Freeswitch to handle more calls:<br>
>> > what is<br>
>> > the best way to do that?<br>
>> > My first idea is to use a Sangoma D500 board to reduce the CPU load. Can<br>
>> > be<br>
>> > this a solution?<br>
>> > There are different way to scale UP?<br>
>> > Thanks in advance,<br>
>> > Stephen<br>
>> ><br>
>> > _______________________________________________<br>
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>> ><br>
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