I’m super happy to see the FreePBX v3 announcement.
I’m very confident in the FreePBX / Freeswitch community to give us a better VoIP futur.
In parallel to this project, i think that it is now important for VoIP telephony to define rules for providers as well as fitter / final users, specialy level two and level three providers.
Interoperability with SS7 world is important as well mainly for billing as there is today a lack of support for special numbers from VoIP providers.
But there are other important problems as well, like the absence of standard in the VoIP world for ANI (charge number in SS7 world) transmission from the caller to the final called party. This gives problems for urgency services, and law conformity. LLDP-MED is the first standard at level 2 ; ANI origination at the switch level, but this needs to be extended at level 4, using for example SIP remote party ID in a standardized way from the client IPBX to the provider. This ANI needs to be transmitted as well through SIP providers, and converted to the SS7 charge number when the destination is a PSTN number.
Another problem is codec compatibility. We absolutly need to use G711 codecs from end to end at the world scale. Too often, we have a terrible audio quality because of a double or triple conversion from / to G711 to G729 or GSM. This need to be standardised, providers should not have the possibility to use G729.
G729 should be used exceptionnaly at a final client site, when G711 cannot be used for bandwith reasons.
To often, we see peoples using G729 for no reasons, even at provider level. This is terrible.
When a party is using G729 or other lossy codec, the full call path need to switch to this codec (passthrough), to avoid multiple conversion. This is mandatory for audio quality. Most of the time this is not true today, with or even without providers in the audio path, because neither softwares neither providers do respect rules for this. Codec negotiation needs to work from end to end, or if not, stay G711 from end to end.
It is terrible to see that hardawre manufacturers begin to implement HD voice, when we still have many problems with 8 KHz VoIP.
Cluecon and similar evenements are nice and important, but it is very important as well to define simple rules at the world scale for interoperability and quality of VoIP.
We hope to see the FreePBX project give simple advices at those levels, for example in the GUI itself by small textual advices.
Last, Enum needs to be used to begin the PSTN to Full IP VoIP transtion. Enum is a transition technology, before using true URI dialing. It is strange to see that URI dialing is more common than Enum. Enum is realiable. Often more reliable than VoIP providers.
Then we’ll have a true alternative for the actual PSTN network, not before. And the change from PSTN to full IP VoIP will really begin to happen.
The market will greatly change as well with the full IP transition, with less money for PSTN and VoIP telcom providers / manufacturer, and more money for VoIP fitters, IP link providers, and VoIP software programmers.
Thanks for your listening,