[VoIP] Naive question

[Chad Perkins]

On 12 Dec 2006 at 16:29, Steph Kerman wrote:

> Chad Perkins wrote:
> > On 11 Dec 2006 at 21:03, Steph Kerman wrote:
> >> Chad,
> >> Thanks for exploring some of the ramifications of my question.
[big snip]
> 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. 

CNET is a "techie" bunch for the most part, some more than others, some in different 
areas.  In an effort to get more of our Telephone Museum members involved I have 
been trying to perfect a configuration using two-line ATAs (similar to what Greg has 
done) as many of these folks are "telephone people" and/or "collectors" but not 
"computer" or "Internet" people.

> That said, there are ways to combine the two separate lines into an
> RJ14 config. using OTS stuff from Radio Shack. 

I know, probably $10.  Two extra line cords, a splitter and a coupler hitched up back 
to front.  I'm still tempted to warm up the solder gun... [snip]

> So who is your "SIP provider" from
> whom they capture your IP address?

Normally the commercial (SIP) provider is going to capture the IP address when the 
SIP application (device) starts so that they know where to send calls, status 
monitoring and "messages" (message waiting indication for example), etc.  

Here's a question if anyone knows, if a Vonage customer calls another Vonage 
customer, does it ever hit the PSTN?  I think I can safely say the other providers I 
use/am acquainted with do not.

> Is that another party other than Stanaphone or FWD?

No, one in the same.

> I see.  But a switching
> function, if I understand correctly, would double the delay so an ENUM
> would seem to be the best way. 

Yes and no.  Even with a switching function in the middle many VoIP 
implementations allow for the voice (and video) channels to be redirected to the end 
points as soon as the connection comes to avoid this very issue.  The end points do 
however maintain a control connection with the switch for things like the ability to 
transfer, conference, etc., update Call Detail Records, and more.

>> 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?

Sorry, Class 4 and Class 5.

>> 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?

This is a theoretical answer and based on some assumptions not having done this 
quite this way personally ...

1. SIP device: The ATA (or IP phone) would have to be (re)configured to accept  
unsolicited connections (if you think that sounds kind of risky you are right, read on).

2. Firewall: Based on #1 the SIP device had better be "behind" are firewall (I have no 
experience with the combo units).  Because the device is only registered with the 
commercial provider, the firewall isn't going to be expecting, or know what to do with 
incoming IP packets from the Asterisk "tandem"; therefore the firewall will need to 
have port forwarding enabled for the proper (SIP) ports ... and because the device is 
now open to the Internet on those ports you would want to add an access list 
restricting connections to just the Asterisk tandem (and hope you don't kill the 
commercial connection in the process).  This is an area that varies widely from 
vendor to vendor.

3. Asterisk tandem: Because the device isn't registering the him he isn't going to 
automatically know where on the Internet the device is in terms of IP address, so the 
'sip.conf' file in the /etc/asterisk directory will created/modified to specify the address 
to use in connection with that device.  This would probably be done one of two ways, 
either by hardcoded IP address, or by a dynamic DNS (DDNS) name.

4. Internet Address:  Because of #3 above, the end device is going to have to either 
have/share a fixed ISP IP address; or you would need to setup a DDNS entry.  
Unfortunately I don't think I've seen a device with firmware that supports DDNS.  One 
should be able to install a DDNS client on a regular PC on the same network as the 
device to get around this though.

FWIW these four basic items are roughly the same steps one needs to address in an 
ENUM setup, in fact I see no reason why under the right circumstances such a 
device couldn't participate in an ENUM based network (mind you while CNET is an 
ENUM network, it is also based primarily on the IAX2 protocol as opposed to SIP 
which can be a pain in the neck.

Moral of the story, go with an unlocked two line unit, use the standard registration 
process; it will be a lot easier with a lot less problems.

c.