[clug] Asterix at home

[Daniel Pittman]

Bob Edwards <bob at cs.anu.edu.au> writes:

> Hi all (gender/species/organism neutral enough?),

Yes, thank you.

> I'm looking into options for doing VoIP at home. I'm guessing most folk
> doing this on this list will be using a dedicated router/modem with built-in
> VoIP support, but I want to keep using my Linux home router/NAT box (with
> squid, apache, Yubikey-protected SSH etc. etc.).
>
> So, anyone doing VoIP through NAT at home using Asterix? Care to share your
> config and/or experiences?

Yes.  I don't really think three is much to share in my configuration; it is
pretty much as basic and textbook as you can get in terms of dialplan and SIP
registrations.

I use pretty standard ATA devices with an analog phone attached; given a
choice, though, I would *strongly* advise that you lay out the money for a
real VoIP phone.  The Siemens Gigaset DECT unit that is available all over the
place is passable.


That said, Asterisk is really not much fun to work with and I plan, some time
soon, to set up a trial FreeSWITCH deployment and see how much better (or
worse, though that is hard to credit as possible) it is...

> I'm thinking of running Asterix on my NAT router box, and using it to bridge
> the RTP packets from the internal to public-IP sides and back.  Is that the
> right approach?

Well, if you can pass the RTP packets through to the actual ATA you will get
marginally reduced delay, and anything that reduces delay is a net win.

One thing to watch out for: at least for me, the iptables SIP NAT helper will
mess up the SIP connection, so that not only does SIP not work, but if it is
loaded at all no RTP traffic gets through.

Regards,
        Daniel
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