Would anybody have any numbers on how large a box would be required to convert
100 or 200 SIP calls to IAX2, without transcoding, echo cancel, .. Or a setup
with individual IAX2 calls coming on one side, and trunking being used to 1
or more remote boxes on the other side, to improve bandwidth usage ?
It doesn't matter if you don't have a test done for exactly 100 or 200
calls,
I'm just looking for with configuration 'A', I was able to switch
'x'
concurrent calls before having quality problems, or system load going thru
the roof.
I'm seriously thinking about developing a trunking VPN utility that would
alow
me to add trunking outside asterisk's code, so I can keep jitter buffer.
I'm
much better coding in 'C' from ground up then changing existing code. It
would know IAX2 packet format and take packets between the local host and
each remote one and bufffer them for say 50ms (configurable) adding all
subsequent packets to the first one, flushing that macro packet, then
decoding on the other side, much like a VPN tunneling protocol. I already
have my own VPN that does almost exactly that, except I'd like it to know
much more about IAX2 packets, in order to compress that better.
Regards,
Marcelo Pacheco