[Asterisk-Users] Sip Trunking

[Eduardo Goncalves]

On Mon, 05 Jan 2004 15:42:25 -0600
Steven Critchfield <critch at basesys.com> wrote:

> > 	On my lab tests, SIP with gsm uses 26kB/s, since the link is a
> > frame-relay and cisco routers,  I've used cisco rtp header
> > compression, and got 16kB/s per channel.
> 
> Something sounds fishy here.
> 
> Asterisk sends out 50 packets a second of audio(20ms). If your numbers
> above are per channel, you achieved a 10k reduction in 50 packets, or
> 204.8 bytes average per packet. Since a GSM audio packet contains 33
> bytes of audio, this large header compression sounds fishy. If you are
> talking bits, not bytes, then it isn't that impressive. You still will
> probably find more efficiency in IAX. Try it and tell us your results
> before shooting it down.

	Sorry, the results are in bits per second, not bytes. my mistake. I'm
doing measure tests with SIP and IAX2 trunking. I'll finish today and
post the results.

	Thanks for the tips
-- 
Eduardo