Can I setup a cloud base server with Freepbx and have a few hardphones with extensions? Each one would only be calling the other phone. They will not have any VoIP service. And be put in different countries.
You can put the PBX in the cloud anywhere and connect IP phones (hardware/software) to it and make calls between them. If you have analog phones you have to get ATA devices for them.
You might not even need a cloud server, many SIP phones/ATA’s have multiple “Accounts” and can be setup to call your other SIP phones/ATA’s directly.
True but in cases where they are all connected over the public Internet and behind NAT, direct endpoint-endpoint calls are not the best solution. Too many factors. Using a B2BUA, like Asterisk/FreePBX, is the best solution to avoid many of the pitfalls of NAT’d endpoints on both sides of a call.
Many of those SIP phones/ATA’s also encourage the use of a stun server. (There I was thinking that Asterisk itself has historically alway had problems with properly traversing NAT also)
Well that depends on the deployment and configuration. You are also comparing a time when there was only Chan_SIP, which had pitfalls. PJSIP follows SIP more closely and is NAT aware for the endpoints sending traffic to it.
The solution you are suggestion with the STUN server can work, however, there are a lot of things that have to be considered with it. The use of public STUN servers can work but then that would require there are public STUN servers that can be accessed by the users. The issues of codecs have to be dealt with along with making sure each device is configured to not register and also be able to accept/make calls with not being registered. All of that is possible in the device configuration.
@Bryck would have to provide more details about the endpoints that would be involved. Are they behind NAT? What countries are they in? Do those countries have limitations on internal and external SIP traffic (in country/outside country)? Can they access public STUN servers? And really, is the expectation just to make calls between them or are there going to be features like voicemail, etc that would be wanted/needed with this at any point?
Because really, PBX or Endpoint/Endpoint w/ STUN are both subject to a couple of those conditions. If the country blocks external SIP traffic either by transport/port or both then both options have issues that would need to be addressed for it.
If’s and what-ifs apart, pragmatically, just get two yealinks and a dynamic dns for each end, choose a stun server , it will generally just work, if it doesn’t . . .
Caveates . . . (well of course there will be many)
If one is behind a provider (or country) that does DPI one might be able to change signalling port. If they block the actual internet to a major degree, then one is probably SOL. either with direct connections or Asterisk using any channel driver (except maybe iax2)
Don’t worry about codecs, all these devices will offer and accept g711 in any flavor.
OK so just so I understand. Getting the endpoints, setting up DDNS per endpoint, finding public STUN servers they can use, setting up each endpoint to not send REGISTER/SUBSCRIBE’s, to make calls when no REGISTER is detected and leave the firewalls open to the world for SIP traffic since each endpoint is an unknown IP from different parts of the world. That is the most practical and sensible way of doing this?
Just don’t use udp:5060 for your SIP signalling it was just NEVER a good idea, EVER! Everything will work fine on any port between 1025 and ~ 63k, tcp is even less attacked but I have NEVER in ten years seen any attempts outside 5000-5999 on either protocol and further you will not be using ip addresses, you will be using your dynamic dns names which are of course at any one point in time known to both ends and resolve to a unique IP address, at least within a relatively short timeframe.
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