[Asterisk-Users] Sip Trunking

[Steven Critchfield]

On Mon, 2004-01-05 at 10:24, Eduardo Goncalves wrote:
> Hi list,
> 
> 	I have to connect two asterisk box, in this scenario:
> 
> [asterisk1]----sip----[asterisk2]----PSTN
> 
> 	I must use sip, cos we'll use cisco rtp header-compression to save
> bandwidth. 

Will rtp header compression necessarily be better than full removal of
IP headers? I'm betting rtp headers are already quite small and any
compression there is minimal with respect to how IAX trunking combines
what would otherwise be many packets into a single packet thus removing
about 40 bytes per packet per call over 1. 

> 	Could you tell me the best way to send calls from asterisk1 to
> asterisk2, since I cannot use IAX trunking?

Sounds like you have already made decision to go down a path. Are you
truely willing to accept better approaches?
-- 
Steven Critchfield  <critch at basesys.com>