[Asterisk-Users] Sip Trunking

[Steven Critchfield]

On Mon, 2004-01-05 at 12:47, Eduardo Goncalves wrote:
> On Mon, 05 Jan 2004 10:19:24 -0700
> Jared Smith <jsmith at drgutah.com> wrote:
> 
> > On Mon, 2004-01-05 at 09:24, Eduardo Goncalves wrote:
> > > 	I must use sip, cos we'll use cisco rtp header-compression to
> > > 	save
> > > bandwidth. 
> > > 
> > > 	Could you tell me the best way to send calls from asterisk1 to
> > > asterisk2, since I cannot use IAX trunking?
> > 
> > 
> > Maybe I'm way off base here, but I'm pretty sure that IAX2 trunking
> > will save you more bandwidth than rtp header compression, at least if
> > you've got multiple calls going between the two servers...
> 
> 	I don't think it's the case. I'll have only 4 channels. 
> 
> 	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.
-- 
Steven Critchfield  <critch at basesys.com>