[Asterisk-Users] Sip Trunking

[Eduardo Goncalves]

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.

 
Eduardo