[sr-dev] Interconnecting SIP servers without the PSTN

Raúl Alexis Betancor Santana rbetancor at serlink.es
Sun Jun 20 10:30:15 CEST 2021

It's seems you totaly missundestood the Telco market.

There are TONS os ways to intercomunicate and route SIP traffic worldwide between SIP servers.
It's not a technical issue, never have been, Its a MONEY issue, because Telcos whant to be paid
when you use their networks to finish your traffic and on a fully open-world, who whould pay?
Thats the simple reason behind any proyect like the ones you named have failed.

Seems you came from the "wonder world" of Asterisk, where they are proud of reinventing the wheel
each couple of months/years with a new "unique and marvelous way" of doing the same that have been
standarized years ago by anyone else (IAX2 and DUNDI are clear examples of that).

The correct way of doing what you call, it's just use DNSSRV, NAPTR and that the national TELCO regulators
runs their ENUM part of the e164.arpa domain space. But again, that have not been done simply because
of a money issue, not technical one.

>From the security point of view, also solved years ago, just using TLS, DTLS or any other sec. protocol.

And using the public DNS to publish your SIP servers it's not "painting a target on you back", 
hackers will find you, that's a FACT, no matter how hard you try to hide your servers (using non-standar ports
or any other kiddy way). The solution it's to have your systems correctly setup and up-to-date and have reactive
sensors that block fraud traffic.

Raúl Alexis Betancor Santana 
Serlink Telecom S.R.L.U.

----- Mensaje original -----
De: "Bill Neely" <bill at telopar.net>
Para: sr-dev at lists.kamailio.org
Enviados: Viernes, 18 de Junio 2021 17:02:03
Asunto: [sr-dev] Interconnecting SIP servers without the PSTN

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

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