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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
