[Freeswitch-users] SIP Proxy - Kamailio/Opens* load-balancing - simple example please?

[Kamil Nigmatullin]

If you want to have a sip proxy on board then you have to learn how sip
works. You can set up opensips and the example script will work as a simple
sip proxy. But to get exactly what you want you have to learn what each
module does and how it change the packets.  I can recomend you to start
from book called Building Telephony Systems with OpenSIPS 1.6.

2015-02-03 23:27 GMT+06:00 I put the Who? in Mishehu <mishehu at freeswitch.org
>:

>  Avi,
>
> I understand your reluctance, as I've been down this path too.  Admittedly
> OpenSIPS and Kamailio both have a pretty steep learning curve even when
> compared to FS.  The trick to remember is that your config in either
> OpenSIPS or Kamailio are SIP message base, because first and foremost both
> operate on an individual SIP message basis, and it's only via the tm module
> that you get transactional SIP awareness, and limited dialog awareness with
> the dialog module.  But believe me that it's well worth the time spent to
> learn these systems, as they are highly powerful and you may find you want
> to do a lot more than just what you outlined, but you'll also have the
> power to implement that as well.
>
> As for more stateful load balancing, there is the loadbalance module which
> can track capabilities that you define on your entries, as the dispatcher
> module is basically just a fire-and-forget without any actual load
> management.
>
> --
> Yossi Neiman
>
> On 02/02/2015 02:55 AM, Avi Marcus wrote:
>
>  Hi - I know this has come up several times and I see several examples
> even on the old wiki - but I've never used a SIP proxy before, and I don't
> understand most of the configuration in the examples.
>
>  Here's what I believe is a fairly common use case:
>
>  1) I have inbound calls via various carriers. Some only support an IP
> endpoint (so DNS won't work) and some don't support 302 redirects, so I
> can't use a stateless FS endpoint either, so I need a sip proxy.
>
>  2) I need a SIP proxy that will route calls to various FreeSWITCH
> endpoints.
> (Preferably, it should route registration too (or manage registration list
> itself?))
>
>  3) I need the ability to pull freeswitch nodes out of the routing
> (waiting for them to drain is fine, I don't need failover of live calls) to
> perform maintenance, and then add them back to the routing.
>
>  4) I should be able to have this proxy on a floating IP that I can move *this
> *too, *without downtime*, for maintenance work.
>
>  I think this is mostly handled by the
> <https://wiki.freeswitch.org/wiki/OpenSIPS_configuration_for_2_or_more_FreeSWITCH_installs>
> examples <http://wiki.freeswitch.org/wiki/SBC_Setup> I
> <https://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS> found
> <http://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration>
> - but the opensips configuration files are hundreds of lines that I don't
> understand. If it's just routing calls to a backend, shouldn't that be
> possible in a small numbers of lines that are more understandable?
>
>  I imagine someone can probably just point me to a tutorial/working code
> that I can use. Several have been shared but I don't recall any one being
> particularly simple... Also, most of them tell you about compiling code. FS
> has been released in packages - has opensips? That might cut off many steps
> from an updated tutorial.
>
>  Thanks!
>
>
>  -Avi Marcus
>
>
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-- 
Kamil Nigmatullin
Tel: 77272323748
mob: 7 (707) 2517003
Skype: kamil.nigmatullin
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