The GNU Bayonne 2 Telephony Application Server
Posted Nov 3, 2005 16:52 UTC (Thu) by
cventers (subscriber, #31465)
In reply to:
The GNU Bayonne 2 Telephony Application Server by ncm
Parent article:
The GNU Bayonne 2 Telephony Application Server
Asterisk in my experience seems to be stable, but having worked with the
actual source code, it's a freaking mess in some major ways. For example,
there are a lot of various functions you can apply in the dialplan to
control a call in the middle of it, and *each one* implements its own
argument string tokenizer. (Props to the Asterisk developers on IAX,
though... it's one of the few sane protocols I've seen developed in a
while).
Of course, being an engineer in the VoIP industry, I'm not convinced that
anyone here knows anything about code quality, or even cares for that
matter. SIP is an HTTP-like protocol, though typically it uses UDP as a
transport mechanism. It seems that all the proprietary software vendors
in this industry like to charge absyminally hideous rates for
PBXs/software that never seems to be capable of more than 20 calls per
second on powerful hardware (powerful meaning that Apache, which must do
HTTP over TCP, can achieve 1500 requests/sec) [asterisk, despite its
limitations, does remarkably better].
In any case, open source's presence in VoIP is just now taking off, and
I'm expecting to see a lot of these proprietary software vendors get
absolutely flattened in the next 5-10 years. I was pretty impressed by
one thing at this year's Fall VON 2005 - Digium (makers of Asterisk) had
the largest booth with a lot of partners present, and they had constant
traffic.
I have yet to use Bayonne, but competition in open source is good.
Everyone's quality will go up.
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