On Aug 6, 2006, at 6:30 PM, Daniel Swarbrick wrote:
>> I don't want to deal with the all too common ISDN is different from
>> POTS is different from Foo is different from Bar is different from
>> Baz.
>
> Precisely why we are favouring an external SIP to PSTN gateway these
> days, or centralised PSTN breakout with SIP trunks.
SIP is not my idea of a unified call model either and it won't be for
a variety of reasons.
There is a proven standard already out there, it's called Q.931 and
that should be the yardstick.
>> There is a clear need for a project that is neither encumbered by
>> Digium's nor by Tony's licensing policy. The inability to use
>> libraries that are GPL only, such as Unicall, is a serious
>> disadvantage and OpenPBX should stay away from incorporating
>> Freeswitch on these grounds alone.
>
> The nice thing about FreeSWITCH is that their development model can
> actually encourage closed source modules
... while at the same time locking out GPL only libraries like Unicall.
Tony himself told me that Unicall was out of reach for Freeswitch, so
you better believe it.
And that's why I said there is a need for something in between
Asterisk and Freeswitch, licensing wise.
Those who need to link to closed source can use Freeswitch.
Those who need to link to GPL only source but don't want Digium's
dual-licensing, they can use OpenPBX.
And if OpenPBX isn't going to serve this space, for sure there will
be another Asterisk fork that will. Only a matter of time.
> I suspect embedded devices probably have higher long
> term reliability than PCs, and are usually cheaper.
Indeed. Running a PBX on a PC is most definitely not a mainstream
thing anyway, it's a more of a geek thing.
> Call volume becomes
> a concern though, especially with CPU-hungry codecs such as G.729a.
This is not really an issue as there are more hardware codecs for
embedded solutions available than there are codec libraries for host-
based processing, especially for patent encumbered codecs like G729.
rgds
benjk
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