[VoIP] Naive question

[Steph Kerman]

Chad Perkins wrote:
> On 11 Dec 2006 at 21:03, Steph Kerman wrote:
>> Chad,
>> Thanks for exploring some of the ramifications of my question.
>>
>> ISTR the Linksys ATA having has 2 separate RJ11s. Not having an RJ14 
>> is no real drawback as far as I can see. Anyone who sets this up with 
>> the intention of using it seriously will probably distribute it 
>> through their own inside wiring or home a residential key system or 
>> PBX on it.
> There are many models of Linksys (and the pre-Linksys branded Sipura). 
> The unit many people think of is the PAP2 (-XX) that Vonage and a 
> number of the other commercial providers sell/supply with their 
> service; it does have two RJ11s for the two lines/services it can 
> support. I have three (v3.1.3) and it is basically the same as the 
> Sipura SPA-200X of which I had one (v3.1.2), which Rick Miller is 
> trying to get online with, if anyone cares.
>
> To be fair most people aren't phone geeks that aren't going to want to 
> have to set up a PBX/key system in their house; not to mention know 
> how or want to spend the money. 
Most people in the population at large, for sure.  I was thinking in 
terms of the people who have commercial Voip and might be inclined to 
provide a second access to CNET. 

The instructions many of the Voip providers give for connecting up the 
ATA can result in back-feeding the analog side of the ATA towards the CO 
on the cable pair that originally delivered the PSTN dial tone.  Or 
limit you to using one phone plugged directly into the ATA.  Or 
stringing modular cords around.  Many ATAs, when installed as 
instructed, deliver reversed loop voltage and will not work with older 
2500 and Trimline sets.
> My irritation with the manufacturer is that they make this consumer 
> product (the two line ATA(s)) but didn't take the time (the cost of 
> place two extra traces on the circuit broad is negligible in terms of 
> the cost of the product) to make it (RJ14) so a normal person could 
> plug a two-line phone into it.
Probably not done more due to lack of competence on the part of the 
designer.  A large part of the non-phone population has no idea what an 
RJ14 is.

That said, there are ways to combine the two separate lines into an RJ14 
config. using OTS stuff from Radio Shack.
>> Can you please clarify in what sphere the Voip number (08130174) you 
>> referred to below exists? On what server or other resource would this 
>> be translated to an IP or other physical address associated with your 
>> ATA?
> Sure. 08130174 is my Stanaphone "number". Stanaphone, for those who 
> aren't familiar with them, is a VoIP company akin to FWD that offers a 
> free service (between subscribers). Like FWD they also offer an 
> optional "opt in" PSTN service for a fee ($0.016/min outbound), and 
> they Peer with many of the other providers like FWD. That means that I 
> can not only call everyone on Stanaphone free, but everyone on every 
> other network that Stanaphone peers with for free (and vice versa). 
> When I bring up my system and register with my SIP provider they 
> capture my IP address so they will know where to send incoming calls, 
> etc. However, just
> because the number-to-IP association is automatically made on their 
> server doesn't mean that the address couldn't be leveraged by a savvy 
> customer to do other things (just not automatically).
So who is your "SIP provider" from whom they capture your IP address?  
Is that another party other than Stanaphone or FWD?
>> Do the alternatives you mentioned below all result in direct 
>> connections between the two ATAs to minimize delay? I'm unaware of
>> the role of the CNET tandem you mentioned. Perhaps that is used when 
>> calling into one of Greg's inbound PSTN portals.
> No, not really. Although that is possible and I have heard of people 
> doing it, it is almost be like getting a dry copper pair between two 
> locations. It wouldn't be useful for connecting to anyone else. So, in 
> reality you need a lookup function like ENUM or a switching function 
> in the middle.
I see.  But a switching function, if I understand correctly, would 
double the delay so an ENUM would seem to be the best way.
> In our CNET we use Asterisk switches as tandems to front end the EM 
> machines and equipment. In addition to Cl.4-ish tandem functions most 
> of us also have Cl.5 services in our Asterisk tandems too; for some 
> that includes SIP extensions.
Please ID what "CI.4" and "CI.5" refer to .  Are these abbreviations for 
something?
> As I eluded to in my previous post, if the SIP device is registered to 
> a commercial providers switch as the opposed to a CNET switch 
> (Asterisk tandem) it still may be possible to have both on one (port 
> of a) SIP device, but the configuration would become totally manual.
I assumed that if it were feasible at all, it would have to be manually 
installed.  Is any of this configuration at the ATA itself or is it all 
done at the Asterisk that is sharing it?

Thanks
Steph
>> Chad Perkins wrote:
>>>> Is there any practical mechanism by which people who subscribe to
>>>> commercial Voip services could be called on a direct Voip basis by
>>>> CNET users, without going out into the PSTN and NANP? Steph
>>>>
>>> To clarify:
>>>
>>> Q: is there a way for CNET users to call Vonage-like subscribers on
>>> a strictly IP basis (without going through the PSTN per se)?
>>>
>>> A1: As Ben said, the short answer is no.
>>>
>>> A2: I know this isn't the answer you are looking for, but as others
>>> have mentioned there are some "roll your own" solutions. For
>>> example John N. could call me on my commercial VoIP line if he knew
>>> my VoIP number (08130174). Creating a CNET-to- VoIP gateway for
>>> single user would be relatively easy; making a broader, scalable
>>> solution is a little more complex.
>>>
>>> A3: The broader answer is that CNET does not "peer" with other VoIP
>>> networks. And to do this right this is what we would need to do.
>>> There are dozens if not a hundred VoIP operations that peer with one
>>> another. FWIW I'm told Vonage no longer participates in any peering
>>> arrangements (but then again chances are anyone looking to save
>>> money isn't going to be using the most expensive provider after
>>> maybe VeriZon and at&t [jab, jab]).
>>>
>>> On 11 Dec 2006 at 15:19, Steph Kerman wrote:
>>>
>>>
>>>> If the 2nd port of a 2-port ATA could be hosted by an existing CNET
>>>> *, that would be a step towards what I was asking about.
>>>>
>>> Yes, if someone has an unlocked ATA that works. Some of us are
>>> doing that now (unfortunately the ATA manufacturers don't seem to
>>> know what an RJ14 is) and a number of VoIP providers allow you to do
>>> that; so all you need is a two-line phone (if that's your pleasure).
>>>
>>>
>>>> Ideally, there
>>>> would be some way to register the same ATA port with 2 hosts or
>>>> otherwise at least provide incoming service to it.
>>>>
>>> Each host has so many configuration parameters that may not be
>>> capatable with other that this would be tricky and neither the
>>> manufacturers nor the providers are going to want to implement some
>>> thing that could really get ugly. On the other hand I think there
>>> is a slightly geeky way of accomplishing this [see* below].
>>>
>>>
>>>> Outgoing is less
>>>> important since the commercial Voip subscriber presumably has
>>>> free/cheap toll service and can call the CNET user through the
>>>> PSTN.
>>>>
>>> If outgoing were not a requirement*, it should be possible to
>>> "drive" SIP calls to a properly configured SIP device without it
>>> being "registered". This is something I have not tried and there a
>>> number of issues (without registration, I think the CNET tandem is
>>> going to need to configured for a fixed IP of the device, and manual
>>> firewall configuration will most likely be necessary to let the SIP
>>> & RTP in).
>>>
>>>
>>>> The CNET user
>>>> OTOH might not be a commercial Voip subscriber and being able to
>>>> call commercial Voip subscribers saves the CNET user toll charges.
>>>>
>>> True, but signing up with a commercial provider that will give you
>>> rates under $0.02/minute is so easy why wouldn't one... skip that.
>>> CNET users have (access to) Asterisk which comes with FWD sample
>>> configs "built in". FWD has excellent peering with other SIP
>>> providers (except at&t, VeriZon $ Vonage) and is free.
>>>
>