[u-u] Cheap/free VOIP home phone
uu at dclg.ca
Sun May 24 23:50:38 EDT 2020
On 2020-05-24 22:01, D'Arcy Cain wrote:
> On 2020-05-24 15:24, D. Hugh Redelmeier wrote:
>> - the dream of VoIP was that it could be end-to-end but it seems to
>> normally deployed as customer to ITSP where it then gets onto the
>> PSTN. Long distance is handled differently by the ITSP,
>> transparently to the customer.
> Hugh's dream could come true. All we need to do is adopt IPV6 and give
> every phone a public IP address. We could do away with the proxy server
> (what I do) and make direct calls just like making SMTP, ssh calls and
> the like. We could also do port forwards everywhere but that would only
> allow for one phone in an office.
> That would also be asking for too much network knowledge from your
> average luser.
Actually, you can already achieve it. Even as a luser. There already
exist standards. You create SRV dns records:
[1:35:335]root at vr:~> host -t srv _sip._udp.daveg.ca
_sip._udp.daveg.ca has SRV record 10 0 5060 sip.eicat.ca.
and when you "dial" dave at daveg.ca, you should lookup _sip._udp.daveg.ca
and use the information (in this case host: sip.eicat.ca, port 5060) and
send the string "INVITE dave at daveg.ca" to that host as a sip message.
This used to work. If anyone finds otherwise, I'd be willing to debug.
... and like a mailserver, you don't need to run it yourself. I run my
own mailserver ... but even among _this_ group ... probably only half of
you do. Anyways... you don't want to talk to an endpoint here --- at
least not most endpoints or ATAs. You want to talk to a SIP call router.
In astersik, you create the extension "dave" in the default
(non-authenticated) context and route "dave" however you please.
... although... if you dial a random string at most phones or ATAs ...
they will "ring" ...
To achieve this as a luser, you simply need a service provider. They
exist. I would do it, for instance. I _think_ easydns was planning to
roll it out, too.
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