[Asterisk-Users] Sip Trunking

Jared Smith
Mon, 05 Jan 2004 10:19:24 -0000

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...

(Then again, I might be a little biased, since IAX2 trunking was my
idea.)

Jared Smith