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. -- To unsubscribe, e-mail: opensuse+unsubscribe@opensuse.org To contact the owner, e-mail: opensuse+owner@opensuse.org