sip/voip : DID, URI, ENUM, opened networks : how to understand it better?


im coming back to ask on this forum, sorry if im at the wrong place, but i guess there are some sip experts around here (if it’s not the right place, tell me where…)

i was wondering, what about the public enum usage for sip networks?
i mean, is there any type of public networks, known for having an opened/accessible users scheme?

eg : here in france there are linphone and ippi, and others (providing software+service)
both, offers both software+service, but can be dissociated
means its possible (normally) to use another sip software with a provider, etc

the idea is, there is what i think called a DID/URI, whom looks like email principle : ; but in a way, it’s a bit similar for xmpp and sip : could email, call or message ; and normally, it’s minded to be interoperable (or federated) on an opened-network circuit.

what i try to understand is :
URI or DID might be i guess, the
however, if sip is really used today, it’s on closed networks. I mean, using data, only whatsapp and others, I guess uses xmpp or sip, but not in an opened-way, means not, what would be great to be communicate with, without having to install their app.

on ““real”” opened-community, such as callcentrix or, it’s the case : a user of a domain could reach another domain, using it’s DID/URI email.

well, I found then that there is a kind of ENUM thing I discovered few days ago. instead of defunct iNum, where it was supposed to be as an online voip accessible and free from internet (like skype) new kind of voip network, ENUM looks like to be enclosed one. As operators mainly looks for ““security”” (or security of closed-business), i found those things :

so if i understand well, mainly operators took SIP protocol to create their own voIP network, to their susbcribers ; not bad if it wasnt totally-closed?

imho, i then discovered that all operators around the world have (for a part of them) a voip/sip account for all volte or even for business lines. My mind was, sip looks like to be great working. But what surprised me in the bad way : why isnt it possible, because it’s 100% data usage, to call them directly from a common “voip/callcentric/linphone/other” sip account?

i guess it’s a question of money, but i dont understand then why people dont go directly on a sip account if their computer is internet connected h24 or even LTE/5G running permanently on their phone : it would be double unlimited plus free of charge international calls. Know people will tell me “just it’s whatsapp”, but no : whatsapp brings just simplicity + zero rating in some countries, not privacy-compliant and opened and federated usage. am I wrong?

on my own, I would have a little question for the nerds or passionate, like me, to SIP protocol for opened-networks, to ensure it’s possible to simple voip communicate with relatives, on the same scheme as email : just call a contact by it’s, and it might work. How could you convince your relatives to go on a such scheme?

I mean, in EU we have several voip initiative, whom are fully accessible from worldwide callers, where in N/america there are and callcentric, maybe others… are those users only using it for pstn-paid calls? why not skype then?

in addition : is there a directory of opened-sip URI/DID/ENUM to be in touch with, to see what operators/voip service providers permits their users to be called from anywhere around the world?

the main advantage of this is like email (or xmpp) : just with a voip/sip software, + a sip account, calls to any other sip user around the world, is generally totally free and unlimited. Why dont people adopt it, instead of whatsapp&etc?

are they some opened-voip providers directory, where some individuals or organisations could be called from over the world, through sip?

do you know some organisation, NGO or companies whom can be called from just a opened-sip network provider?

thank you for reading, and much more if you share your thoughts
(and sorry for bad eng…)

You understand it very well, hd88, and welcome!

It is very good for the operators to keep both legs on-net! :ice_hockey: :goal_net:

And it is probably not so bad for the customers because inertia is a thing and change can be hard.

Perhaps adding new things is easier than stopping old things?

And time.

There is much business to learn here, but here is some fun to enjoy on your way:

Phase 1: Steal Underpants.
Phase 2: ?
Phase 3: Profit

Making this Asterisk related… and not SIP but IAX… here’s a gem from Asterisk version 1.0 extensions.conf.sample file:

exten => 500,2,Dial(IAX2/ ; Call the Asterisk demo

Might still work :person_shrugging:

Maybe flashing QR codes for click-to-call via URL that opens an in-browser WebRTC session for SIP is where you want to go next?