Of course any (other) company can provide VoIP service. The crucial differences are:
1) Standard PSTN equipment (handsets & PABXs) vs VoIP equipment
2) ’Cheaper’ termination on PSTN calls (although e.g. IS can terminate on other networks, they don’t get anyone terminating on their network, so they don’t recoup anything as a percentage of calls) – reminds me of partial-settlement network peering
3) Inbound calls from the PSTN via standard dialling – basically the definition of PSTN/leased telephone-number-space
These deficiencies are not endemic to being a VoIP provider. VoIP refers just to the transport method, of course, but is traditionally used to imply an IP connection for the voice equipment too.
And the solutions to the above-listed problems aren’t too difficult (from a technological, if not policy, view):
| 1) | Use Asterisk as a VoIP (service) to PSTN (handset) translating PABX. Comparatively low hardware cost depending on implementation size |
| 2a) | Technological solution: Avoid PSTN termination. As more users migrate to VoIP, Metcalfe’s Law comes into effect – more users means more value for users to migrate, which means even more users. |
| 2b) | Technological solution 2: Terminate on the cheapest destination interface (‘least-cost-routing v2’ – this pre-supposes ability to determine that multiple interfaces connect to one destination (either a sort of reverse-NAT, or a lookup, e.g. NAPTR) vs traditional hot-potato routing) |
| 2c) | Policy solution: Regulate excessive interconnect fees (that 600%!) |
| 3) | Acquire alternative numbering. ITU-T should demand that Enum compliance allows the various Carriers to dish out extra space (currently only a maximum of 10% of the number space is usable because of the way that area codes are assigned – 2 digits instead of 3) and that it is routed to by the PSTN. Policy questions will probably block this for a while, until ICASA decides whether to promote competition. Alternatively an international provider could supply globally-visible telephone number space and use NAPTR records to direct SIP sessions. |
