Mailinglist Archive: opensuse (769 mails)

< Previous Next >
Re: [opensuse] Asterisk
On 2016-05-28 20:41, Per Jessen wrote:
Carlos E. R. wrote:

Notice that Asterisk doesn't behave always "peer to peer". Sometimes
it handles the call, changing to another codec in its CPU, and sends
forward to the destination. Asterisk might thus also handle NAT
without STUN if it sits on both networks.

It's not about Asterisk, it's about the SIP client. Any properly
configured SIP client can connect to Asterisk and make calls, but if
the client is on a NAT'ed network, something has to keep the NAT
entry "open" - keep-alive or STUN. I'm not really sure what the big
difference is.

Well, I'm considering a slightly different case here (based on mine).
I'm (my phone is) on a 10.*.*.* network, as would be the asterisk server
on the ISP. Both on the same network. And then thinking how would I make
a call to a SIP phone on internet. If the asterisk server sits on both
networks, it would handle it completely, I think.

Just thinking out loud :-)

Cheers / Saludos,

Carlos E. R.
(from 13.1 x86_64 "Bottle" at Telcontar)

< Previous Next >
This Thread
Follow Ups