[u-u] Cheap/free VOIP home phone

[David Gilbert]

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.