Mailinglist Archive: opensuse (769 mails)

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Re: [opensuse] Asterisk
  • From: Per Jessen <per@xxxxxxxxxxxx>
  • Date: Sat, 28 May 2016 21:24:23 +0200
  • Message-id: <nicr97$9ek$1@saturn.local.net>
Carlos E. R. wrote:

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.

Right.

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.

A SIP-call is typically done with a URL sip://someone@domain - from
<domain>, the SIP client determines which server to contact and that's
pretty much it. With a SIP-client on 10/8, you could do the same, but
only if your telco has a NAT facility. Which I don't see any reason for
them to have. If you do it from your normal internet connection (e.g.
with a softphone), NAT'ed or otherwise, it should just work. (STUN and
keep-alive don't matter). If you want to play with it, I'll be happy
to set up a SIP account on our Asterisk.


--
Per Jessen, Zürich (19.8°C)
http://www.hostsuisse.com/ - virtual servers, made in Switzerland.

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