[Freeswitch-users] Scaling Freeswitch
colin.morelli at gmail.com
Sun Apr 17 04:14:17 MSD 2016
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.
On Sat, Apr 16, 2016 at 5:19 PM Jurijs Ivolga <jurijs.ivolga at gmail.com>
> 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:
> 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
> Nevertheless it is still possible to route PSTN calls directly from
> Kamailio too, without freeswitch.
> With kind regards,
> On Sat, Apr 16, 2016 at 10:23 PM, jungle Boogie <jungleboogie0 at gmail.com>
>> 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
>> > and insight as well.
>> This won't answer all your questions but it will give you an idea of
>> freeswitch + Kamailio:
>> inum: 883510009027723
>> sip: jungleboogie at sip2sip.info
>> xmpp: jungle-boogie at jit.si
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> Official FreeSWITCH Sites
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> Official FreeSWITCH Sites
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
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