[sr-dev] Interconnecting SIP servers without the PSTN

[Bill Neely]

greetings all:

I have long believed that VOIP and SIP will not reach their full 
potential until SIP servers can route calls to other SIP servers without 
having to go through the ancient telephone system, and pay their tolls.

There is nothing of substance preventing any SIP server from calling 
numbers at any other SIP server. They just need to know which numbers 
are hosted on which servers. There have been several attempts to resolve 
this issue: freenum.org, e164,org, Dundi (for asterisk). All appear to 
be dead at this time.

I think that one of the reasons for these failures was that all of these 
systems relied on the public DNS system to exchange server location 
info. Putting your SIP server address on a public system and advertising 
that this is the IP of a SIP server is simply begging for hackers to 
attempt to breach your SIP server. Its like painting a big target on 
your back.

We at Xantek have been working on an alternate approach, using AGI calls 
and responses to identify routing info. This approach allows us to limit 
server identification to registered users of the system, and registered 
users will have to provide identification (something that hackers 
probably won't do).

We also are incorporating a PIN number into the dial string, so that 
recipients are aware that the call is coming from a valid user. The PIN 
can be easily changed if fraudulent activity is suspected.

We have a working model for Asterisk set up (see voipconnect.tel for 
details), but we would like to expand into the Kamailio-verse. What we 
need is a few Kamailio experts to help with the development of the 
system on Kamailio. If you have any interest in helping, please reply to 
this post.

TIA, Bill