The biggest issue with using a SIP proxy as a PBX is performing
authentication on outgoing calls to carriers.
I use asterisk front-end-ed by the proxy. Like this, I can provision
authentication credentials on asterisk and route the call from
asterisk to carrier through the proxy.
I don't like the idea of running the proxy on the router (if I change
the router or the firmware on the router I need to do more work) and
therefor I run the proxy and the asterisk on two small arm boxes and I
route calls between them. I register the subscribers on the proxy and
I route through asterisk only when I need to.
Regards,
Ovidiu Sas
On Tue, May 14, 2013 at 10:27 AM, u <ueberwachungsst...@googlemail.com> wrote:
> I would like to share my experience with kamailio and other home pbx servers.
>
> Kamailio on my kirkwood home router for my 6 SIP users is perhaps
> overkill: I don't really need mysql and "scalability". But at last I
> finally managed to make calling between registered users work stable.
> My voip clients only work in all NAT scenarios if I work around some
> bugs: to use csipsimple on android I had to change rtpproxy_manage()
> to rtpproxy_manage("c") in kamailio's default config, so that problems
> with conflicting c: entries in the SDP go away.
>
> I propose kamailio could ship with a special example
> kamailio-compatible.cfg that doesn't try to be RFC compliant, but
> compatible to the most common voip clients. Right now the only thing I
> would change for this is the option for rtpproxy_manage, but I'm sure
> others will know more common quirks that could safely be enabled to
> increase compatibility. I think this compatibility idea is what yate
> sticks to for their defaults. In freeswitch you also have to do it all
> manually, and it's much more work to figure things out in their
> enormous config files.
>
> The other SIP proxies I had tried before kamailio officially fit all
> my requirements, including support for multihomed dynamic IPs, but
> contrary to their claims it didn't work.
> Yate was easy to set up, but the default dialplan is more confusing
> than powerful and after having made everything work I realised yate
> was clogging my CPU and RAM and after some time always randomly
> stopped working. This is with only 2 users connected! It also wasn't
> possible to fix NAT sdp while leaving the codecs section in the SDP
> alone at the same time. I tried to debug the code, but the C++ was so
> complex that I had to give up.
> Freeswitch was much more difficult to setup, a multihomed setup with
> dynamic IP was super buggy and it also didn't help that the
> unintuitive configuration is all in complex unreadable XML
> configuration files.
>
> Kamailio and rtpproxy don't officially support dynamic IP address, but
> I can just restart both each time my DSL provider forces me to a new
> IP address. This happens automatically in the night and is no big
> hassle really. The most simple, least-featureful solution works best
> it seems.
>
> Now the last problem I have with kamailio: I don't know how to connect
> my accounts to my sip providers (i.e. Sipgate, Betamax, Dellmont).
> I would like a simple way to do this, preferably without other
> features that always seem to complicate the matters. Is there
> something more lightweight and simple than asterisk, freeswitch and
> yate, that people use successfully for this task together with
> kamailio and rtpproxy?
>
> u
>
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