[Freeswitch-users] Scaling Freeswitch

[Colin Morelli]

Jungle + Jurijs,

Thank you, the combination of the YT video and your description is exactly
what I was looking for. So, in summation here, what you'd have is

SIP Client ----> Kamailio ----> FS (contains a dialplan to bridge
user/1234) ----> Kamailio ----> Other SIP Client

Am I understanding correctly? I assume Kamailio would remove itself from
the media flow in this case, leaving just Client --> FS --> Client for
media transfer, right?

It sounds like this still wouldn't remove the need for shared state
management across the FS cluster, though. So, you'd have a Kamailio cluster
backed by a database processing SIP registrations and location information,
backed by FS with a shared database storing call state. This should allow
the failure of either a Kamailio instance and/or FS instance and still
allow the call to be recovered. It'll also allow you to offload all SIP
registration to Kamailio and it's database, while leaving FS freed up for
media/call routing. Does this sound right?

This has been incredibly helpful, thank you so much.

Best,
Colin

On Sat, Apr 16, 2016 at 5:19 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
wrote:

> Hi,
>
>
> If Kamailio is the registrar, what role does it play in call routing? For
>> example, if a call hits FS and I instruct FS to bridge to a user, how is
>> that performed? Or does all call routing now need to be performed on
>> Kamailio, since it's the one that's aware of where users are (and their
>> associated presence)?
>>
>
> You can configure Kamailio and Freeswitch in anyway you need, Kamailio can
> be call router, freeswitch can be call router and both of them
> simultaneously can be call routers :)
>
> But if we will take following manual as starting point:
>
> http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc
>
> Then users location DB will be in Kamilio and information regarding users
> will be stored in Kamailio, so Freeswitch will not know where user located
> and all calls between extensions will go through Freeswitch and then will
> be looped back to Kamailio and Kamailio will decide where to route call.
>
> So in this case Kamailio is call router and it decides where to route
> calls, for example 44 prefix is routed to "vbox"(voicemail).
>
> Nevertheless you can add some additional routing on Freeswitch too if
> necessary, for example if you need to send calls to PSTN and you need
> transcoding.
>
> Nevertheless it is still possible to route PSTN calls directly from
> Kamailio too, without freeswitch.
>
> With kind regards,
>
> Jurijs
>
> On Sat, Apr 16, 2016 at 10:23 PM, jungle Boogie <jungleboogie0 at gmail.com>
> wrote:
>
>> On 16 April 2016 at 09:32, Colin Morelli <colin.morelli at gmail.com> wrote:
>> > To clarify, I'm just looking for pointers/references here. Although if
>> > anyone has some personal experience I'd greatly appreciate specific
>> examples
>> > and insight as well.
>>
>> This won't answer all your questions but it will give you an idea of
>> freeswitch + Kamailio:
>>
>> https://www.youtube.com/watch?v=6VsuC7-jHc4
>>
>>
>>
>> --
>> -------
>> inum: 883510009027723
>> sip: jungleboogie at sip2sip.info
>> xmpp: jungle-boogie at jit.si
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
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>> http://www.freeswitch.org
>> http://confluence.freeswitch.org
>> http://www.cluecon.com
>>
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>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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